ReleaseNotes9:Firmware: Difference between revisions

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= V9 Hotfix 1 (90600.01) =
= V9 Hotfix 3 (90600.03) =
Changes included in Version 9 hotfix1
Changes included in Version 9 hotfix3
[http://mantis.innovaphone.com/view.php?id=63846 Definition]
[http://mantis.innovaphone.com/view.php?id=67520 Definition]


== New Features ==
== New Features ==
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=== SoftwarePhone: default configuration ===
=== SIP: Send OPTIONS at configurable interval ===


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The default configuration is changed to coder G.711a and license type PBX.<!---->
For keep alive pruposes on interfaces without registration.<br/>Required for Lync interoperability.<br/>(config change TSIP /options-interval 30)<!---->
''Status:''
sip.cpp/h<br/>siptrans.cpp/h
      
      
=== SoftwarePhone: license information ===
=== SIP: Fast re-routing on gateway interface w/o registration ===


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If no valid license is received from the PBX, a information is printed directly now.<!---->
OPTIONS is used to check availability of remote peer.<br/>If OPTIONS fails, not trying to send INVITE.<!---->
      
      
=== DECT: menu in IP6000/IP6010/... ===
=== SIP: New config file option /no-ms-acceptedby ===


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The DECT menu is visible in the IP6000/IP6010/IP0010/IP1060/IP3010 devices, if the required DECT multi-cell license is available.<!---->
Some IP phones get confused by this Microsoft extension.<br/>Cisco SPA 303 and Cisco SPA 962 don't stop ringing anymore<!---->
      
      
=== SoftwarePhone: de-registration at application end ===
=== ip200a/230/240: handset conversations can be monitored in a directly connected  headset ===


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The application sends a de-registration to the PBX, if the user closes it.<!---->
This feature is required for a special application and is supported only for ip200a/230/240 phones with a directly connected headset (non DHSG).<br/>It is enabled via<br/>  config add INCA_DSP /handset-spy <volume><br/>whith <volume> in the range from 1..8  <!---->
      
      
=== DECT: OEM device user configuration ===
=== PBX: Translation of Cause "Call Rejected" to Cause "User Busy" for endpoint objects only ===


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For an OEM device the user data handling must be different. This does not concern the IP1200.<!---->
If the cause is received from a gateway, it is forwarded transparently now<!---->
      
      
=== myPBX: Redial ===
=== myPBX: Show active sessions on the admin interface ===


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Add a button to the web application that writes the last dialled number into the number input field.<br/><br/>Files:<br/>client.js<br/>client.htm<br/>action_redial.png<br/>pbx.mak<!---->
The active myPBX sessions are shown on the page PBX/myPBX.<!---->
      
      
=== PBX Root Node Extern on Slave for calls from Master ===
=== Phone: Added new function keys to Phone-UI (Toggle &amp;amp; Prepare Override) ===


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If a Root Node Extern interface was configured on a slave this was currently only used in case this was a "License Only" slave.<br/><br/>Now this interface is also used for calls from the master with no destination found. This can only happen in non-replicated environments.<!---->
Description: Phone: Added new function keys to Phone-UI (Toggle &amp; Prepare Override)<br/><br/><!---->
''Status:''
pbx.cpp
      
      
=== phone: improved czech display texts ===
=== PBX: New presence activity "do not disturb" ===


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now all texts are translated to czech, previous errors were fixed (translations provided by zakharova@annexnet.cz)<!---->
If a users sets this presence activity no calls are passed.<!---->
      
      
=== SoftwarePhone: Tray Icon ===
=== PBX: Allow CFB on Gateway Type Objects ===


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Now the SoftwarePhone runs as Windows application with an tray icon.<!---->
A CFB is triggered by a User Busy. If a CFB is used for example at a Trunk, the CFB is executed when the called remote user returns busy. Because this may be unexpected the CFB was not executed at a Gateway Type Object.<br/><br/>It is now enabled again, because it is useful when connecting external systems which return busy to indicate an out of channels situation<!---->
      
      
=== PBX: Send 'Connected Name' in Connect ===
=== PBX Trunk/Gateway: Round robin within registrations to same device, different devices sequentially ===


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This is for example needed to display E-Mail Icon in myPBX for contacts in remote PBX<!---->
This way both round-robin or sequential usage of gateways can be configured<!---->
''Status:''
signal.cpp<br/>signal.h<br/>h323sig.cpp<br/>h323sig.h<br/>pbx.cpp
      
      
=== DECT: forward info message to radio ===
=== improved test for Timeslot Switch Chip of ip6010 ip3010 ip0010 ip1060 and ip6000 ===


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The info message is forwarded from the gatekeeper to the radio in an early call state now. The message is used for display updates by a OEM PBX between call setup and alerting response from DECT.<!---->
intension is better analysis of hardware problems<!---->
''Status:''
idt72_drv.cpp, idt72_drv.h
      
      
=== DECT: User import file convertion ===
=== ip22/24/28/302/305/6010/3010/1060/241/222/232: False DTMF detects ===


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Support for latin1 to UTF-8 conversion for user import added. The import feature is only used in an OEM device.<!---->
Seen on IP222 and IP6010<!---->
''Status:''
ac_dsp3.cpp
      
      
=== DECT: Internal endpoint information 'assigned-by' ===
=== phone_orchid: builtin color display test ===


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The DECT system saves the information with which method an endpoint was assigned to an user - by the administrator or with the feature code 'user login'. It is used by OEM modules.<br/><!---->
After the builtin test function has been started the display test mode is entered when the 'Esc' key is pressed. Numeric keys trigger a full screen test display, all other keys stop the display test mode.<br/>To the keys 0..9 the following patterns are assigned: <br/>DarkGray, White, Grey, Black, Red, Green, Blue, Yellow, Cyan, Magenta<!---->
      
      
=== DECT: Location update option for UTF-8 support ===
=== IP-DECT: Configuration XML data for OEM device ===


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A new option is added for the location update events to distinguish between latin1 and UTF-8 handsets. This feature is only used by an OEM device and OEM handsets for display updates to support UTF-8.<!---->
A new configuration XML attribute is added for changing a OEM GUI.<!---->
      
      
=== Truncating UTF-8 aware string copy function ===
=== Phone: "Function keys not modifiable on the phone" mask should disable creation of new function keys of masked type ===


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New string library function to correctly truncate an UTF-8 string to fit in a buffer.<br/><!---->
Description: Phone: "Function keys not modifiable on the phone" mask should disable creation of new function keys of masked type. Currently, only modification of preset function keys is disabled, but the creation of new ones enabled and possible.<!---->
      
      
=== SoftwarePhone: Setup about logo ===
=== Phone: Added command line option to hide Administration Menu and/or MAC/Serial completely ===


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The setup about innovaphone logo was changed.<!---->
Description: Phone: Added command line option to hide Administration Menu and/or MAC/Serial completely. See /hide-mask option to PHONE ADMIN-UI in wiki for more information.<br/><br/><!---->
      
      
=== SIP: Handling on INFO inside instant message session ===
=== IP-DECT: OEM registration string ===


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INFO sip:xxx@172.16.0.200:5060;transport=TCP SIP/2.0<br/>Via: SIP/2.0/tcp 172.16.1.61:5060;branch=z9hG4bK.DF25F93ADB524A283550ECBD871BC73B<br/>From: <***>;tag=902D6B1B5622453B000172A9724270D4;epid=00000003<br/>To: <***>;tag=1440372429<br/>Call-ID: 5D4FA4260D42445F4071AEA032D4321B<br/>CSeq: 3 INFO<br/>Contact: ***<br/>Content-Type: application/xml<br/>Max-Forwards: 70<br/>Supported: timer<br/>Content-Length: 82<br/><br/><?xml version="1.0"?><br/><KeyboardActivity><status status="type" /></KeyboardActivity><!---->
The Radio registration string of an OEM device is changed to check a OEM license in the DECT Master.<!---->
      
      
=== SIP: Locating remote federation server with another SRV query ===
=== SIP: Diagnostics of transport error "Remote server certificate mismatch" ===


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There are different service name decorations in use<br/>  _sips._tcp.<domain><br/>and also<br/>  _sipfederationtls._tcp.<domain><br/><!---->
If remote sverer certificate does not match the destination domain name<br/>signaling connection is refused by client.<br/><!---->
      
      
=== SIP: Treat "Privacy:off" like "Privacy:none" ===
=== myPBX: Support for WebEx meeting passwords ===


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Treat "Privacy:off" like "Privacy:none" when receiving INVITE<!---->
In WebEx a meeting can have a password that must be entered by the attendees when they join. Some WebEx accounts can only create meetings with passwords.<br/><br/>The possibility to configure a global meeting password is added to the PBX/Config/myPBX page.<!---->
      
      
=== DECT: Increase number of anonymous RAS registrations ===
=== Phone: Message function key ===


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Now the number of the allowed anonymous RAS registrations is increased to 30.<!---->
Description: Phone: Message function key. Multifunctional depending on number of unread messages. Stores one prepared message (with destination and message text) and presents the new message screen when invoked. If incoming messages pending, display the letter/message icon and jump to incoming-messages subscreen upon invocation.<!---->
      
      
=== IP241: Multi line text ctrl must support line break ===
=== phone: Finnish translations updated ===


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Multi line text ctrl must support line break when displaying received text messages or when composing new text messages.<!---->
no english placeholders anymore, Texts begin with upper case letter<!---->
      
      
=== DECT: Option not to use TLS ===
=== IP150: dimming of key LEDs and LCD backlight ===


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for easier debugging<!---->
to save power in special environments the key LEDs can be dimmed by<br/>  config add KEYS0 /light-off<br/>The lcd backlight can be configured the usual menu way on the phone. <br/><br/><!---->
''Status:''
dectradio.cpp<br/>dectradio.h
      
      
=== SIP: New config file option /no-alert-info ===
=== HTTP-Client: Allow user names longer than 16 characters ===


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Supress Alert-Info header for interoperability with Thomson ST2030 and Thomson TB30<!---->
Now user names with up to 64 characters are allowed.<!---->
      
      
=== IP241: Show active CFU destination on idle screen ===
=== Phone: Call forwarding (always, busy, no reply) destination now choosable from dial-menu ===


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Show active CFU destination in header bar of idle screen.<br/>Not only a symbol on status bar.<!---->
Phone: Call forwarding (always, busy, no reply) destination now choosable from dial-menu. Usage: enter number or search for phonebook entry, press menu-key, scroll down to choose call-forwarding (always, busy or no-reply) and acknowledge choice in CF-screen.<br/><br/><!---->
      
      
=== H.323: Display call state in "Signaling Timeout" error log ===
=== IP-DECT: Allow setting empty text for idle display ===


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To provide better indication about the nature of the problem<!---->
Some handsets will not show signal strength and battery symbols if idle display is defined, they must be set to empty string.<!---->
''Status:''
h323sig.cpp
      
      
=== SIP: add an "alias" header field parameter to the Via header field ===
=== ac_dsp3: support echo canceller trace ===


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According to RFC-5923 to allow TLS connection re-use<!---->
required to analyze echo canceller problems<!---->
''Status:''
siptrans.cpp
      
      
=== SIP: New config file option /add-cn-capability ===
=== Phone: On IP240, make OK key a headset activation key (along with Space Key) ===


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Required for mediation server (lync) interoperability.<br/>Otherwise mediation server complains:<br/>  "The Gateway peer does not support comfort noise"<!---->
Phone: On IP240, make OK key a headset activation key (along with Space key).<br/>Enabled from Phone/Preferences : Use Newline/OK Key as Headset Key.<!---->
      
      
=== new License Mechanism &lt;type&gt;@&lt;num&gt;=n ===
== Bug Fixes ==
 
 
   
=== IPxx10: error handling in sata driver ===


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For hosting provider<!---->
Old cards are producing DMA errors that were not handled properly. Try again read/write operation after error recovery.<!---->
''Status:''
inno_lic.cpp<br/>inno_lic.h
      
      
=== PBX: BC Conference instant messaging support ===
=== DECT: IP6000/IP6010/... default config Master mode off ===


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Instant messaging support with myPBX for the "Bc Conference" PBX object added.<!---->
Now the Dect Master is in mode off by default for the IP6000/IP6010/...<!---->
      
      
=== Gateway: Allow sending of Date/Time in Connect on ISDN interfaces ===
=== SoftwarePhone: DTMF to voice mail ===


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Was missing in the User Interface, so it could not be configured<!---->
Now, sending DTMF with myPBX is correctly working. This fixes sending DTMF to voice mail.<!---->
''Status:''
relay_edit_phys.xsl<br/>config.h of ip800, ip24, ip3000, ip6000, ip6010
      
      
=== Web-UI: Restyled Config Wizard ===
=== VM: Trap while processing self-forwarded call ===


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Config wizard got new look<!---->
VM: Trap while processing self-forwarded call<!---->
      
      
=== PBX CDRs: Transfered call generated 2 cdrs. This was confusing ===
=== SIP: Uninitialized data in SDP offer/answer ===


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For an incoming call which was transfered, one CDR for the incoming call and one for the outgoing transfer was generated. Now one CDR for the incoming call including the information for the outgoing transfer is generated<!---->
Applies to G.726 exclusive calls only.<!---->
''Status:''
pbx.cpp<br/>pbx_api.h<br/>pbx_wait.cpp
      
      
=== IP232: Implemented touch keyboard ===
=== Phone: Main menu scrolling below last item broken ===


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Implemented touch keyboard<!---->
Phone: Main menu scrolling below last item broken.<br/>1st item hould be activated upon down arrow press (done) and screen focus moved up (not being done - bug).<!---->
      
      
=== IP222: Alpha input using the num block ===
=== cpld update not working on ip241 ===


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Alpha input using the num block<!---->
-<!---->
''Status:''
phone_orchid.cpp phone_orchid:config.h xilinx.cpp<br/>and as consequence also ip6010:config.h
      
      
=== DECT: Option short busy tone ===
=== SIP: Interoperability with Lync and media-bypass ===


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New option 'Short busy tone' for the DECT system added. If this is enabled, the busy tone after remote hangs up is shortened.<!---->
Ack contained wrong To-Tag when calling a lync client in media-bypass scenario.<br/>Results into call drop after 30 seconds.<!---->
      
      
=== DECT: License check for OEM ===
=== Phone: Automatic key repeat did not work on IP222/IP232 ===


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The DECT Master checks the DECT Radio license for an OEM device and PBX.<!---->
Automatic key repeat did not work on phone w/o alpha keyboard<!---->
      
      
=== myPBX: Show allows that are configured using templates ===
=== Call to Voicemail did not work anymore with SRTP ===


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So that the user can see all access rights<br/><br/>files:<br/>pbx_client.cpp, client_allows.js<!---->
This was a collateral damage from fix<br/><br/>http://wiki.innovaphone.com/index.php?title=Support:DVL-Roadmap_Firmware_V9#Allow_configuration_of_SRTP_crypto_suite.2C_to_be_used_for_media_proposals<!---->
      
      
=== SoftwarePhone: Jabra BIZ 2400 support ===
=== Phone: long function key titles hide idle screen information ===


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Now the Jabra BIZ 2400 is supported by the SoftwarePhone.<!---->
Description: Phone: long function key titles hide idle screen information.<br/>Fixed: important idle screen information now shortens the amount of displayed function key name. Following information is now displayed over the function key text: a) crossed bell icon on do-not-disturb (lines 2+3) b) CFU + CFU-destination (lines 2+3) c) missed calls, unread messages and waiting callbacks (line 4)<br/><br/> <!---->
      
      
=== SIP: New config file option /no-app-rerouting ===
=== PBX Waiting: Call forwarded with DTMF mapping was shown in myPBX for each registration ===


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|Id
|[http://mantis.innovaphone.com/view.php?id=65801 65801]
|[http://mantis.innovaphone.com/view.php?id=67682 67682]
|}
|}
Interoperability with Ingate Session Border Controller.<br/>Required when handling redirect responses.<!---->
The call was not sent with the original conferenceID, so myPBX could not detect that it was in fact the same call<!---->
      
      
=== SoftwarePhone: Jabra SPEAK 410 USB support ===
=== PBX: License accounting in centralized licensing scenario wrong if master not available ===


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|[http://mantis.innovaphone.com/view.php?id=67698 67698]
|}
|}
Now the Jabra SPEAK 410 USB is supported by the SoftwarePhone.<!---->
When the master is available the slave stores the licenses from the master including the usage. This stored usage included the licenses used by the slave itself, so if after a reset the master was not available the local usage just added to this.<br/><br/>Now from the stored usage the local usage is subtracted.<!---->
      
      
=== SoftwarePhone: Jabra LINK 320 USB support ===
=== phone_orchid: microphone is not mute on a call intrusion in  silent monitoring mode / microphone cannot be muted in a conference ===


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|[http://mantis.innovaphone.com/view.php?id=67704 67704]
|}
|}
Now the Jabra LINK 320 USB is supported by the SoftwarePhone.<!---->
On a call intrusion in silent monitoring mode the microphone of the intruding party must be mute. In a conference the micro should be muted when the micro key is pressed and unmuted when the micro key is pressed again.<br/>Muting the microphone did work when only one call was active but not when two calls were active as in a intrusion/conference.<!---->
      
      
=== DECT: Device state info for OEM GUI ===
=== Phone: CLIR on text messaging did not work ===


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|}
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Device state info for OEM GUI added.<!---->
CLIR on text messaging did not work<!---->
      
      
=== phone: disable call intrusion via partner key when recording is active ===
=== SIP: Interoperability with Lync: Handling of REFER from Lync ===


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Call intrusion cannot be performed while recording is active:<br/>- recording establishes a 3party conference between local party, remote party and recorder.<br/>- call intrusion establishes a 3party conference between local party and the two remote parties<br/>- recording and call intrusion at the same time would require a 4party conference which cannot be set up because the phone has only 2 DSP coder channels.<br/><br/>Now if any kind of recording is configured call intrusion is neither offered in 'recall' menu nor performed via partner key.<!---->
REFER for blind transfer was rejected with "406 Not Acceptable",<br/>due to absence of user part in refer-to URI.<!---->
      
      
=== IP241: Do not hide all fkeys in active state ===
=== PBX Trunk: Problem with Forking to trunk if multiple GWs are registered to Trunk ===


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|[http://mantis.innovaphone.com/view.php?id=67720 67720]
|}
|}
Keep these fkeys on display that do not interfere with call controls.<!---->
If one of the gateways rejected the call (no channel, not connected, ...), the original call from which was forked was disconnected<!---->
      
      
=== SoftwarePhone: Jabra PRO 94xx support ===
=== Phone: Calls received with CLIR appear in call list with an empty entry ===


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|[http://mantis.innovaphone.com/view.php?id=67722 67722]
|}
|}
Now the Jabra PRO 9400 serie is supported by the SoftwarePhone.<!---->
Description: Phone: Calls received with CLIR appear in call list with an empty entry; now fixed to display the CLIR text: anonymous/unbekannt/...<br/><br/><!---->
      
      
=== SIP: Handling of NOTIFY(message-summary) on gateway interfaces without registration ===
=== PBX: License accounting did not work with Unknown Registrations under some special conditions ===


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|[http://mantis.innovaphone.com/view.php?id=67731 67731]
|}
|}
Required for LYNC interaction<!---->
It could happen that a registration to a user was not accounted for if the endpoint used for this already had an unknown registration at the time the user was created<!---->
      
      
=== SIP: Improved CCM interoperability ===
=== User Interface: Alignment of tables fixed ===


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|}
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Required for IP-DECT<!---->
Some strange alignments were introduced with the last hotfix<!---->
      
      
=== PBX Admin UI: When editing allowed access, show configuration inherited from templates ===
=== Gateway: Interface Name with multiple "Umlaute" did not work ===


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|[http://mantis.innovaphone.com/view.php?id=67737 67737]
|}
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Result was sometimes confusing, because it was very hard to find templates from which the current object inherited config<!---->
There was a length limitation of the URL encoded output, which was already exceeded if three "Umlaute" (or any character which is encoded in more the one byte with utf-8), were used<!---->
''Status:''
pbx.cpp<br/>pbx_api.h<br/>pbx_edit_allow.xsl
      
      
=== QSIG: Avaya expect Progress Indicator with external calls ===
=== IP-DECT: Packetization could change after handover ===


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|[http://mantis.innovaphone.com/view.php?id=67738 67738]
|}
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Avaya uses the Progress indicator 'Interworking with a public network' to identify a call as external. This Progress Indicator is now added for calls from a Number NOT with private numbering plan (which is our way to identify internal calls)<!---->
On the new radio the RTP should be sent using the same packetization as was negotiated with the original call<!---->
''Status:''
q931.cpp
      
      
=== Gateway: New interface option "No Early Media" for SIP interoperability ===
=== SIP: Fix for early media from Waitng Queue ===


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New config option "No Early Media" in section "SIP Interop Tweaks".<br/>Required for LYNC interoperability to ignore SDP answer in 183 Session Progress,<br/>since no inband progress information is provided by LYNC.<br/><!---->
PROGRESS after ALERT was not handled by SIP stack.<br/>Now 183 Session Progress with SDP is send after 180 Ringing w/o SDP.<!---->
      
      
=== PBX Mobility: Min-Alert/Max-Alert timer on configured mobility ===
=== Leak checking improved ===


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This way it can be prohibited that an announcement from the mobile provider accepts a mobility call<!---->
sometimes leaks were falsely detected. Problem if objects are about to be deleted, which were not owned by any module anymore. This happend esspecially with httpclient.<!---->
''Status:''
pbx.cpp<br/>pbx.h<br/>pbx_api.h<br/>pbx_mobility.cpp<br/>pbx_mobility.h<br/>pbx_admin.cpp<br/>pbx_edit_fork.xsl<br/>work.css
      
      
=== PBX: Name not case sensitive anymore ===
=== SIP: Generate/add SRTP key on media-relay interfaces ===


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An object Name in the PBX can now be used as user part of a SIP URI and this is not case sensitive. This could be a compatibility issue if users are configured with names which are the same except the case.<!---->
This enables even partial SRTP (SRTP on one side of media-relay).<!---->
''Status:''
pbx.cpp
      
      
=== SIP: Decode application/rlmi+xml for presence from external UC server ===
=== Phone: Enable "Activate Registration" without user/password authentication if "Protect Configuration at Phone" set ===


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|[http://mantis.innovaphone.com/view.php?id=67791 67791]
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Decode application/rlmi+xml for presence from external UC server<!---->
Description: Phone: Enable "Activate Registration" without user/password authentication if "Protect Configuration at Phone" set. Activating a registration is a state change, and not a configuration modification, so allow this option.<br/><br/> <!---->
      
      
=== ISDN: New interop flag to forward network provided or checked cli only ===
=== H.323: A name_id of length 0 resulted in invalid H.450 coding ===


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|[http://mantis.innovaphone.com/view.php?id=67796 67796]
|}
|}
Useful if the real calling number is needed and not a number provided by CLIP no screening<!---->
An empty name identification received was forwarded in H.323 as invalid H.450. Such a name is now forwarded as 'name not available'.<!---->
''Status:''
q931.cpp<br/>q931.h<br/>isdn_interop.xsl
      
      
=== DECT: Configuration option for OEM device ===
=== H.323 Malformed packet ===


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New configuration option 'Output Power' in DECT Radio for an OEM device.<!---->
The ASN.1 encoder had a bug under one special condition: For a constrained character string with a maximum length of more or equal to 16bits, with an effective length of zero, the padding for octett alignment was missing for the zero length bitfield containing the string.<br/><br/>In H.323 this only happens for the CallIdentity used for H.450 call transfer message in case of blind transfer without consultation.<br/><br/>This fix breaks compatibility with earlier versions, for this reason this fix is available for version 9,8,7 and 6.<br/><br/>If phones and PBX with versions containing and not containing this fix are mixed  the following problems will occur:<br/>- A blind transfer without consultation (initiated with the redial key) is not possible<br/>- A call which was transfered without consultation is not displayed at the transfered-to phone as transfered<br/><!---->
      
      
== Bug Fixes ==
=== SIP: Unwanted media-relay sessions when using forking/broadcast/multi-reg ===
 
 
   
=== SIP: SRTP re-keying when retrieving held call ===


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Required for Cisco interoperability<!---->
If in incoming SIP was routed to multiple destinations<br/>the final session could be media-relay although not configured.<!---->
      
      
=== SIP: Session refresh was taken as session modification ===
=== ip22/24/28/302/305: DSP affected by certain fax tones ===


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Local SRTP key was re-calculated after re-INVITE for session refreh was received.<br/>Causes SRTP decode error at remote side.<br/>CUCM scenario<!---->
Brother fax problem, the first fax is transfered, the next fax transfers fail.<br/>Switch to fax from remote is now done without reopening the channel.<br/>Closing the channel waits until t38 is switched off.<br/><!---->
''Status:''
ac_dsp3.cpp ac_dsp3.h
      
      
=== SIP: Media negotiation problem during transfer to early media source ===
=== IP30x, IP1060, IP2010, IP6010: Fax did not work if rerouted from ISDN interface to a Voip destination ===


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|}
|}
test\\9.00\\relay\\early-media failed<!---->
DSP was not configured correctly<!---->
      
      
=== HTTP-Client: Bad encoding of uri parameter in digest authentication ===
=== IP2x2: Handset microphone gain increased ===


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|}
|}
Uri parameter in digest authentication was not URL encoded<!---->
changed from 22.5db to 30db<!---->
''Status:''
ac_codec3.cpp
      
      
=== Gateway: Outgoing Call Completion did not work when outgoing call was routed through TONE interface ===
=== IP241: Remote party information truncated more than necessary ===


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|}
|}
Outgoing CC request did not went out to ISDN interface.<!---->
Line 3 of call ctrl<!---->
      
      
=== SIP: Message buffer too small for REGISTER request for re-try with authentication ===
=== IP-DECT: Changed channel option SRTP crypto suite ===


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On some installations a change-of-nonce at server side may cause volatile "Registration down error" on client side.<!---->
Now DECT system channel configuration option 'Secure RTP' is a drop down box. The DECT Master correctly transmits the changed option. This feature was changed in V9 Hotfix 2, related case #66810.<!---->
      
      
=== SIP: Not accepting calls from alternative proxy ===
=== SIP: DNS problem when SRV response provides no additional records ===


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|}
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When being registered at a proxy with 2 ip addresses the gateway does not accept calls from the alternative ip address.<!---->
If 2-step resolving is required (SRV and A) the service port<br/>of the SRV response got lost and default SI Pport 5060 was used.<!---->
      
      
=== SoftwarePhone: reloading current configuration ===
=== SIP: Trap when configuring STUN server on a SIP/TCP or SIP/TLS interface ===


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|}
|}
The device configuration is correctly loaded in setup.<!---->
STUN is for SIP/UDP only.<!---->
      
      
=== SoftwarePhone: ringing through device ===
=== myPBX: Always send an ID with CT_INITIATE ===


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Ringing through device fixed, if there are more output devices than input devices.<!---->
This is needed for interoperability with phones using v9hotfix2 or older. (See #67803)<!---->
      
      
=== Web-UI: PBX password length is limited to 15 chars ===
=== SIP: Must answer every request - even unknown/unsupported methods ===


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|}
|}
Added tooltip and fixed maxlength attribute on input elements.<!---->
Lync sends proprietary NEGOTIATE request and waits for response.<br/>Otherwise transport connection is blocked for any upcoming request.<!---->
      
      
=== SoftwarePhone: trap with busy external call ===
=== PBX Waiting: A call rejected on a primary operator was never sent to all operators ===


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|}
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If a call to external is busy and is hooked off with myPBX, a trap occurs. This is fixed now.<!---->
if again and again rejected on all primary operators. Problem happend also if rejected by "Do not disturb" on operator phone.<!---->
      
      
=== License: Character encoding problem ===
=== PBX: Master/Slave compatibility problem with version 9 and version 8 and non-ascii characters in PBX name ===


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|}
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Character encoding problem<!---->
In version 8 only latin1 characters were allowed, which means in unicode the high byte was always 0. So it could be ignored and when sending location information between master and slave sometimes the high byte contained 0xff.<br/><br/>In version 9 this non-ascii location information was not correct unicode at all.<br/><br/>The problem happened only if non-ascii characters were used when naming a PBX.<!---->
      
      
=== IP241 DSP may freeze under heavy load ===
=== SIP: Generate new SRTP key on every incoming re-negotiation ===


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A function key test script which creates/releases a big number of calls (single calls, conference calls, intrusion calls) in a very short time resulted in a frozen DSP.<br/><!---->
Required to have different encryption after transfer.<!---->
      
      
=== myPBX launcher: Remove standard URL ===
=== PBX-CDR: Mobility calls to Trunk, external number was missing in CDR ===


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Currently the launcher uses a standard URL (http://172.16.0.10/PBX0/MY/client.htm). This is obsolete and should be removed.<br/><br/>files: ctiwin.cs<!---->
only the number of the trunk itself was available<!---->
      
      
=== calling tone missing in some cases ===
=== IP150: OEM specific WEB GUI modifications did not work ===


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|}
|}
the calling tone was switched off when RTP data was received from remote party after the alerting indication even if the data did not contain in-band info.<!---->
manufacturer specific stylesheets have to be be adjusted to the new GUI style  <!---->
      
      
=== config download may trap when malformed LDAP config data  has been uploaded ===
=== PBX: End of call intrusion was not signaled to the phone ===


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|}
a buffer overrun happens on config download when a "mod cmd FLASHDIR0 add-view nnn cn=..." line with a length > 63 characters has been uploaded.  <!---->
The call intrusion tone was generated even if the intrusion was terminated<!---->
      
      
=== Trap with conference (ADSP driver) ===
=== PBX: User Interface Inconsistency. Same thing was sometimes call 'Response Timeout'/' and 'No Response Time' ===


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A trap in ADSP hardware driver occurs if a new member joins an existing conference in exclusive listen mode. All devices with ADSP hardware: IP6000/IP6010/IP0010/IP1060/IP3010.<!---->
Only Response Timeout used anymore<!---->
      
      
=== myPBX launcher: Hotkey should always bring myPBX to front ===
=== PBX-SOAP: Support UserClear for pending outgoing calls on Waiting Queues ===


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Not only if a number is marked<br/><br/>Files:<br/>ctiwin.cs<br/>client.js<!---->
UserClear was ignored in this state<!---->
      
      
=== myPBX: Script error on line 1351 ===
=== SIP: Fix for media negotiation with SRTP ===


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A script error is thrown when there is an unexpected missed-call-auth:<br/>client.js, Line 1351, this.manager is null<br/><br/>files: client.js<br/><!---->
Exchange of SRTP key may fail after hold/retrieve<!---->
      
      
=== Presence functionality is not available when registered via H323 at a non-innovaphone PBX ===
=== phone_orchid: DTMF Tones detected in voice data from microphone were propagated to remote side ===


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Presence operations via H323 are encoded in private facility elements which are unknown to a non-innovaphone PBX. Presence control calls sent to such a PBX may be misunderstood and routed back as normal voice call to the sending phone.<br/>Thus no presence control calls must be sent to such a PBX.<!---->
DTMF tones may be detected from audible feedback on pressing a dial key in connected state and also from some other source. It's better to propagate only tones requested explicitely via a dial key and not from some external source. <!---->
      
      
=== myPBX: Improvements of Finnish translation ===
=== SIP: Trap handling 491 response on reliable transport ===


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files: pbx_client_localisation.cpp<br/><br/>Soita => Puhelu<br/>Keskustele => Keskustelu<br/><br/><!---->
Trap with MAX_BUSY_TICKS<!---->
      
      
=== phone ip241 - monitor mode (handset/headset + speaker) cannot be disabled once enabled ===
=== phone_inca:  "ETH0/Isolate PC Link" checkmark could not be cleared via WEB UI once set ===


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|}
|}
if monitor mode is set by pressing the speaker key while in handset or headset mode the handsfree speaker gets the same data as the handset or headset speaker. pressing the speaker key again must mute the handsfree speaker.<!---->
Only a WEB UI problem, a "config rem ETH0 /isolate-pc" did help.<!---->
      
      
=== Trap when starting from flash_stick ===
=== SIP: Trap when outgoing SIP call is cancelled while DNS resolving is ongoing ===


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|}
and flash memory not yet programmed with bootcode<!---->
Trap when outgoing SIP is cancelled while DNS resolving is ongoing<!---->
''Status:''
ip6010.cpp
      
      
=== myPBX launcher: Map all variants of the Norwegian language to "no" ===
=== Gateway: Allow configuration of username and password for ENUM/SIP interfaces ===


{|
{|
Line 1,019: Line 1,003:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=63771 63771]
|[http://mantis.innovaphone.com/view.php?id=68147 68147]
|}
|}
There are three ISO 639-1 language codes for the variants of the Norwegian languages.<br/><br/>Norsk (Bokmål) (nb)<br/>Norsk (Nynorsk) (nn)<br/>Norsk (no)<br/><br/>myPBX only provides a translation for "no". To avoid that users with the "nb" or "nn" locale get the English translation, all language codes have to be mapped to "no" by the myPBX launcher.<br/><br/>Files: ctiwin.cs, start.js<!---->
For rare where remote destination server asks for authentication.<br/>(And all remote destination servers ask for same auth or remote destination server s always the same.)<!---->
      
      
=== no dialtone and no audible connection on ip24, ip22 and ip28 with V9 ===
=== SIP: Interoperability with LinkSys SPA3102 ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=63780 63780]
|[http://mantis.innovaphone.com/view.php?id=68174 68174]
|}
|}
even in both directions<!---->
LinkSys SPA3102 gives "g729a" as RTP payload type mapping:<br/><br/>    v=0<br/>    o=- 510843041 510843041 IN IP4 192.168.10.20<br/>    s=-<br/>    c=IN IP4 192.168.10.20<br/>    t=0 0<br/>    m=audio 16404 RTP/AVP 18 100 101<br/>    a=rtpmap:18 G729a/8000<br/>    a=fmtp:18 annexb=no<br/>    ...<br/><br/>Needs to be handled.<!---->
''Status:''
ac_dsp3.cpp
      
      
=== SIP: Allocated message size to small for INVITE redirect response (Avaya) ===
=== IP241: Fix for display rendering ===


{|
{|
Line 1,043: Line 1,025:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=63829 63829]
|[http://mantis.innovaphone.com/view.php?id=68181 68181]
|}
|}
Memory allocation is a bit to tight to fit the message due to many Via headers.<br/><br/>INVITE sip:3003@192.168.150.140:2059;transport=UDP SIP/2.0<br/>Record-Route: <sip:5793d7f@192.168.150.115;transport=udp;lr><br/>Record-Route: <sip:192.168.150.114:15060;lr;sap=315810451*1*016asm-callprocessing.sar1905633216~1304428214402~-1054885358~1><br/>Via: SIP/2.0/UDP 192.168.150.115;rport;branch=z9hG4bKC0A896726E7526620194612-AP;ft=192.168.150.115~13c4<br/>Via: SIP/2.0/UDP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526620194612<br/>Via: SIP/2.0/UDP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526621194610<br/>Via: SIP/2.0/UDP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526621194609<br/>Via: SIP/2.0/TCP 192.168.150.115;branch=z9hG4bK0e2106b7388e016424db9a29200-AP;ft=11786<br/>Via: SIP/2.0/TCP 192.168.150.118;branch=z9hG4bK0e2106b7388e016424db9a29200;avaya-cm-term-reaction=shortcut<br/>Via: SIP/2.0/TCP 192.168.150.115;branch=z9hG4bKC0A896726E7526620194608-AP;ft=12651<br/>Via: SIP/2.0/TCP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526620194608<br/>Via: SIP/2.0/TCP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526621194606<br/>Via: SIP/2.0/TCP 192.168.150.114:15070;branch=z9hG4bKC0A896726E7526621194605<br/>Via: SIP/2.0/TCP 192.168.150.115;branch=z9hG4bK0e2106b7388e018424db9a29200-AP;ft=11786<br/>Via: SIP/2.0/TCP 192.168.150.118;branch=z9hG4bK0e2106b7388e018424db9a29200<br/>Via: SIP/2.0/TCP 192.168.150.84;branch=z9hG4bK200_f1774512c29cc2e5cd78966_I2371<br/>User-Agent: Avaya one-X Deskphone AVAYA-SM-6.1.1.0.611023 Avaya CM/R016x.00.1.510.1<br/>Record-Route: <sip:5793d7f@192.168.150.115;transport=tcp;lr><br/>Record-Route: <sip:192.168.150.114:15060;transport=tcp;lr;sap=315810451*1*016asm-callprocessing.sar1905633216~1304428214355~-1054885362~1><br/>Record-Route: <sip:5793d7f@192.168.150.115;transport=tcp;lr><br/>Record-Route: <sip:192.168.150.118;transport=tcp;lr><br/>Session-Expires: 1800;refresher=uac<br/>Content-Type: application/sdp<br/>Content-Length: 215<br/>...<!---->
Display of diversion destination was corrupt.<br/>Pickup fkey labeling was wrong when party information needed truncation.<!---->
      
      
=== PBX: CDRs from Waiting Queue in case of call thru DTMF map wrong ===
=== Gerneral/Admin page was broken if too many authentication servers were configured ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=63882 63882]
|[http://mantis.innovaphone.com/view.php?id=68231 68231]
|}
|}
The CDR just ended without the transfer and following events<!---->
The number of authentication servers is now restricted to 10.<!---->
''Status:''
pbx_wait.cpp
      
      
=== GW Route Configuration: XML Content Not Well Formed When Inserting A Map ===
=== SIP: Interoperability with Lync ===


{|
{|
Line 1,067: Line 1,047:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=63884 63884]
|[http://mantis.innovaphone.com/view.php?id=68232 68232]
|}
|}
Could happen with certain types of routing maps<!---->
Media negotiation problem on calls coming from on Lync client<br/>and getting forwarded to another Lync client.<br/><!---->
      
      
=== myPBX: Handling of calls with no name and no number ===
=== IP241: Updated some display text information ===


{|
{|
Line 1,078: Line 1,058:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=63885 63885]
|[http://mantis.innovaphone.com/view.php?id=68234 68234]
|}
|}
myPBX updated calls only if a number or a name was set in the call-update or contact-call-update message.<br/><br/>Files: client.js<!---->
No need to strongly abbreviate disconnect cause text.<!---->
      
      
=== SoftwarePhone: conference no speech ===
=== Phone: Added loud note to web-ui that bool funnction key can only toggle if boolean object addressed by number ===


{|
{|
Line 1,089: Line 1,069:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=63902 63902]
|[http://mantis.innovaphone.com/view.php?id=68240 68240]
|}
|}
Sometimes the SoftwarePhone 3pty conference is not working. It is fixed now.<!---->
Description: Phone: Added loud note to web-ui that bool funnction key can only toggle if boolean object addressed by number <br/><br/><!---->
      
      
=== Preserve number of calls on SIG_CONN without connected number ===
=== phone: intrusion call started in handset mode is not terminated when going on hook when TAPI or operator run on PBX ===


{|
{|
Line 1,100: Line 1,080:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=63906 63906]
|[http://mantis.innovaphone.com/view.php?id=68249 68249]
|}
|}
On calls across PBXes the number is discarded when the SIG_CONN does not contain a connected number.<br/><br/>files: pbx.cpp (pbx_connector::receive_connected_number)<!---->
With TAPI or operator running on the PBX the the signaling of a busy condition is changed such that a disconnect instead of a release is sent. The disconnect was not handled correctly, the hookswitch state was lost and the next on-hook signal was ignored. TThus teh call could be terminated with the disc-key only.<br/><!---->
      
      
=== Calling Tone garbled when using G722 with a framesize &gt; 40 milliseconds ===
=== Phone: Switch presence fkey light on if presence activity is set ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64015 64015]
|[http://mantis.innovaphone.com/view.php?id=68256 68256]
|}
|}
The maximum G722 framesize is 40 milliseconds (320 byte packets). A configured size > 40 was not adjusted correctly and triggered the calling tones too fast.<!---->
Switch presence fkey light on if presence activity is set<!---->
      
      
=== One-way audio when using G711u ===
=== myPBX: Wrong charset was used when authenticating against reporting or changing user password ===


{|
{|
Line 1,122: Line 1,102:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64036 64036]
|[http://mantis.innovaphone.com/view.php?id=68258 68258]
|}
|}
Playback of G711u RTP audio may fail and error log "Wrong Payload Type received" is set in some cases.<!---->
Javascript uses Unicode. The PBX and the reporting use UTF8. Therefore strings have to be converted to UTF8 before encrypting them using RC4.<!---->
''Status:''
channel.cpp
      
      
=== No received cause code should be treated as 'normal clearing' ===
=== IP241: New fkey backgrounds ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64043 64043]
|[http://mantis.innovaphone.com/view.php?id=68304 68304]
|}
|}
Was sometimes treated as cause code to do re-routing. This happened esspecially with multiple registrations to v8 gateway object. A call sent successfully to the gateway on the first regsitration was sent again on the second registration after call clearing.<!---->
Less transparency to avoid interference with personalized background images<!---->
''Status:''
q931lib.cpp<br/>relay.cpp
      
      
=== missing response 'reset required' when changing PRIx-Lx config options ===
=== myPBX: Chunked response from application sharing provider did not work ===


{|
{|
Line 1,148: Line 1,124:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64055 64055]
|[http://mantis.innovaphone.com/view.php?id=68311 68311]
|}
|}
changing i.e. the ,NT-Mode' config option didn't show the 'reset required' link button after pressing 'OK'.<!---->
After receiving HTTPCLIENT_RECV_RESULT, pbx_client did not send another HTTPCLIENT_RECV.<!---->
''Status:''
falc56_drv.cpp, config.h ipac_drv.cpp V9:falc56_drv.xsl
      
      
=== ip6010 family : CFLASH overcrowds trace buffer if trace flag is turned on ===
=== phone_orchid: call pickup via partner/pickup key fails when the key is pressed immediately after audible signal ===


{|
{|
Line 1,161: Line 1,135:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64061 64061]
|[http://mantis.innovaphone.com/view.php?id=68313 68313]
|}
|}
dto<!---->
When "Audible Signal after alerting" was configured on a partner/pickup key and the key was pressed while or a short time after the audible signal was played then the picked call was mute.  <!---->
''Status:''
''Status:''
mv78x00_cfdrv.cpp
files: ac_dsp3.cpp
      
      
=== PBX: Transfer Recall timer was not started if destination was ringing after blind transfer ===
=== IP241: Activate external background image from phone menu ===


{|
{|
Line 1,174: Line 1,148:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64064 64064]
|[http://mantis.innovaphone.com/view.php?id=68351 68351]
|}
|}
After a blind transfer without consultation to a busy destination the recall timer should be started as soon as the destination is not busy anymore and the call is delivered<!---->
External background image source can be configured on web ui.<br/>Background image can be selected on phone menu.<br/>Now also external background image can be selected.<!---->
''Status:''
pbx.cpp
      
      
=== Gateway: Allow interface maps for analog interfaces as well ===
=== myPBX: Hide passwords for application sharing and reporting in config ===


{|
{|
Line 1,187: Line 1,159:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64068 64068]
|[http://mantis.innovaphone.com/view.php?id=68352 68352]
|}
|}
Was prohibited in the past, but there are uses for this.<!---->
The passwords are moved from the config line to the VARS.<br/><br/>PBX0/MY-A - Application sharing password<br/>PBX0/MY-R - Reporting password<!---->
''Status:''
ip24/config.h
      
      
=== PBX: Group not configurable for PBX object ===
=== SIP: Failed to register on dynamic PBX ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64075 64075]
|[http://mantis.innovaphone.com/view.php?id=68362 68362]
|}
|}
needed by SOAP applicatons (e.g. TAPI) to find slave PBXs<!---->
Attempt to register on a dyn PBX with SIP protocol was rejected with "301 Moved Permanently".<!---->
''Status:''
pbx.h
      
      
=== PBX: Rej.External Calls not possible for all object types ===
=== IP-DECT: Adding OEM radios to Kerberos realm did not work with passwords containing special characters ===


{|
{|
Line 1,213: Line 1,181:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64111 64111]
|[http://mantis.innovaphone.com/view.php?id=68377 68377]
|}
|}
when creating the objects it was sometimes possible to set this checkmark but later when editing the object the checkmark was not displayed anymore<!---->
The password was not URL-decoded when reading it from the UI.<!---->
''Status:''
pbx_edit_object.xsl
      
      
=== propagating comma separated option values to multiple device dependent config lines did not work ===
=== SIP: Pending control calls on gateway ===


{|
{|
Line 1,226: Line 1,192:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64114 64114]
|[http://mantis.innovaphone.com/view.php?id=68378 68378]
|}
|}
Some devices (for example ip110, ip200a, ip150, ip230, ip240) use an identical firmware which is configured at boot time according to the device it's running on. This configuration is kept on device specific config command lines. A "config change" was not handled correctly.<!---->
Incoming unsolicited NOTIFY(message-summary) may cause pending control call on Gateway.<br/>Control calls are calls (signaling connections) without media channel.<br/>These calls are now released.<!---->
      
      
=== H.323 limit registration redirects to avoid high load on misconfiguration ===
=== WEB GUI page cannot be  scrolled completely when height of left hand logo is too big ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64116 64116]
|[http://mantis.innovaphone.com/view.php?id=68382 68382]
|}
|}
In a non-replicated PBX environment it could happen that a loop of registration redirections is configured. In this case the endpoints were redirected as fast as the resources allowed between the different destinations. This could produce high load on the PBXs.<!---->
Height of FHF logo is bigger than that of the default logo, this must be considered when computing size for iframe below tab lines. <!---->
''Status:''
h323.h<br/>h323sig.cpp
      
      
=== myPBX: Save button for contacts stays disabled when using autocomplete ===
=== DTMF user configuration with invalid checkbox check for presence setting ===


{|
{|
Line 1,250: Line 1,214:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64117 64117]
|[http://mantis.innovaphone.com/view.php?id=68383 68383]
|}
|}
This can happen using Firefox. Turn autocomplete off and use the oninput event additionally.<br/><br/>files: client_contact.htm<!---->
The check of the checkmark of the presence setting was wrong.<!---->
      
      
=== Media negotiation problem H.323 to SIP call, with reverse Media and media relay ===
=== IP222: Alpha input using the num block ===


{|
{|
Line 1,261: Line 1,225:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64124 64124]
|[http://mantis.innovaphone.com/view.php?id=68398 68398]
|}
|}
a call without Media in both directions could be the result<!---->
Automatic switching back to numeric mode when leaving input ctrl.<!---->
''Status:''
h323ch.cpp<br/>sip.cpp
      
      
=== SIP: L16 codec handling was wrong ===
=== SIP: Trap when using TLS as transport ===


{|
{|
Line 1,274: Line 1,236:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64126 64126]
|[http://mantis.innovaphone.com/view.php?id=68410 68410]
|}
|}
Dynamic payload type for L16/16000 and L16/8000 got lost.<br/><br/>SDP-Offer:<br/>    m=audio 5000 RTP/AVP 107 105 0 8 9 101<br/>    a=rtcp:5001 IN IP4 192.168.10.247<br/>    a=rtpmap:107 L16/16000<br/>    a=rtpmap:105 L16/8000<br/>    a=rtpmap:0 PCMU/8000<br/>    a=rtpmap:8 PCMA/8000<br/>    a=rtpmap:9 G722/8000<br/>    a=sendrecv<br/>    a=rtpmap:101 telephone-event/8000<br/>    a=fmtp:101 0-15<br/><br/>Decoded into:<br/>    1:L16/16000(10) . xmit(20) recv(20) 192.168.10.247:5004 (0.0.0.0:0)<br/>    1:L16/8000(10) . xmit(20) recv(20) 192.168.10.247:5004 (0.0.0.0:0)<br/>    1:G711u/8000(0) . xmit(20) recv(20) 192.168.10.247:5004 (0.0.0.0:0)<br/>    1:G711A/8000(8) . xmit(20) recv(20) 192.168.10.247:5004 (0.0.0.0:0)<br/>    1:G722/16000(9) . xmit(20) recv(20) 192.168.10.247:5004 (0.0.0.0:0)<br/>    1:DTMF/8000(101) . xmit(0) recv(0) 192.168.10.247:5004 (0.0.0.0:0)<br/><!---->
Only if remote side closes transport connection while requests are pending.<!---->
      
      
=== HTTP server: Query string is lost on redirect for "Force HTTPS" ===
=== phone_orchid: begin of voice mail prompt was cut off sometimes ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64128 64128]
|[http://mantis.innovaphone.com/view.php?id=68416 68416]
|}
|}
The location header of the redirect does not include the initial query string.<br/><br/>files: http.cpp, http.h<!---->
Connection of Voip to DSP channel was delayed and thus the first packets of the RTP stream were lost.<!---->
      
      
=== Conference on IP6000 Hardware 200 and lower not working with v8hf14 and v9 ===
=== body onload attribute can be extended and tab_active method doesn't crash anymore ===


{|
{|
Line 1,296: Line 1,258:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64132 64132]
|[http://mantis.innovaphone.com/view.php?id=68430 68430]
|}
|}
The ADSP serial port has been changed from SPORT1 to SPORT0 for the IP6010.<br/>Old IP6000 hardware has the SPORT0 not connected, so now SPORT1 is again used on IP6000.<br/><!---->
Custom PBX object XSL had no method anymore to set the onload attribute of the body. This can be now extended with a XSL template parameter.<br/>Additionally the tab_active method has been called by default and the default value caused the method to crash.<!---->
      
      
=== SIP: Memory leak when receiving small datagrams on SIP port ===
=== X509: Fix for reading innovaphone info from flash ===


{|
{|
Line 1,307: Line 1,269:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64134 64134]
|[http://mantis.innovaphone.com/view.php?id=68435 68435]
|}
|}
Some SIP clients send small UDP datagrams in order to refresh mapping on NAT devices.<!---->
Parsing the innovaphone info text was incorrect<!---->
      
      
=== PBX: Potential Trap on calls to exec, map or waiting object ===
=== License: Be safe against factory reset during license invalidation ===


{|
{|
Line 1,318: Line 1,280:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64135 64135]
|[http://mantis.innovaphone.com/view.php?id=68447 68447]
|}
|}
under some rare circimstances, which are unfortunatly not known, there could be a NULL pointer access<!---->
If factory reset is done before license invalidation procedure is complete,<br/>will keep you from completing the license invalidation.<br/>Now the procedure can be completed even after factory reset.<!---->
''Status:''
pbx_exec.cpp<br/>pbx_wait.cpp<br/>pbx_map.cpp
      
      
=== phone orchid: called party microphone muted after a G722 call has been connected ===
=== IP241 - handsfree speaker volume to low ===


{|
{|
Line 1,331: Line 1,291:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64139 64139]
|[http://mantis.innovaphone.com/view.php?id=68451 68451]
|}
|}
This is an effect of the sampling rate change when switching the channel from G711 (ringing) to G722. The microphone was unmuted when the speaker volume was changed thereafter. Now the volume settings are restored automatically after the sampling rate has been changed.<!---->
The handsfree speaker volume was too low even when configured to maximum. Now the general output volume is increased by 3 dB. In case of problems the general output volume can be changed by<br/>    config add AC-DSP0 RINGER /VoiceOutputGain n<br/>with n = 1..63 -> (-32 + n)db, n = 32 -> 0dB, n = 0 -> mute<!---->
      
      
=== myPBX launcher: Hotkey does not work for same number two times in a row ===
=== wrong link to PPP Interface State Info Help ===


{|
{|
Line 1,342: Line 1,302:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64169 64169]
|[http://mantis.innovaphone.com/view.php?id=68463 68463]
|}
|}
files: TextPeeker.cs<!---->
Link out of V9 namespace<!---->
      
      
=== SDP: Wrong clock rate for G.722 codec ===
=== phone: DHSG headset not reset to idle after a hookswitch signal in idle state ===


{|
{|
Line 1,353: Line 1,313:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64182 64182]
|[http://mantis.innovaphone.com/view.php?id=68567 68567]
|}
|}
RFC-3551<br/>  Even though the actual sampling rate for G.722 audio is 16,000 Hz,<br/>   the RTP clock rate for the G722 payload format is 8,000 Hz because<br/>  that value was erroneously assigned in RFC 1890 and must remain<br/>  unchanged for backward compatibility.  The octet rate or sample-pair<br/>  rate is 8,000 Hz.<br/><!---->
most DHSG headsets generate a hookswich signal and enter voice mode when taken out of basestation. This hookswitch signal was simply ignored.<br/>Now the voice mode is cleared after one second if there is no other DHSG event before.<!---->
      
      
=== myPBX: Use different icon for logout ===
=== SIP/TCP: Transport error when connection is closed by client ===


{|
{|
Line 1,364: Line 1,324:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64184 64184]
|[http://mantis.innovaphone.com/view.php?id=68578 68578]
|}
|}
So that users can better see the difference between logging out and closing the window.<br/><br/>files: window_action_logout.png<!---->
If transaction client closes connection before final response has been sent,<br/>the server tries to open a new connection toward ephemeral port of closed connection.<!---->
      
      
=== Phone: trap upon deletion of user reg on pbx  if registration information being viewed on the phone ===
=== SIP: Fix for Dialog-Info notification ===


{|
{|
Line 1,375: Line 1,335:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64186 64186]
|[http://mantis.innovaphone.com/view.php?id=68581 68581]
|}
|}
problem: trap upon deletion of user reg on pbx if registration information being viewed on the phone<br/><br/>solution: fixed in forms code (unbind screen)<br/><br/>files: box/forms/lcd/forms_impl.cpp<br/><br/>products: phones<br/><br/>risks: none<br/><!---->
Send an empty dialig-info XML after inbound subscription.<br/>Required for interop with Grandstream GXP2010.<!---->
      
      
=== PPPOE: specific configuration not reachable from config web page ===
=== SIP: Use secondary proxy address on interfaces without registration ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64192 64192]
|[http://mantis.innovaphone.com/view.php?id=68635 68635]
|}
|}
problem: PPPOE: specific configuration not reachable from config web page, so no new PPPOE can be configured (already present ones run though), also ISDN part always visible<br/><br/>solution: fixed<br/><br/>files: ip_pppif.xsl (now check for PPPOE0, PPPOE1 and PPPOE2 types)<br/><br/>products: all (gateways effected)<br/><br/>risks: none<br/><!---->
When secondary proxy address resolved by DNS it should be used to send INVITE to.<!---->
      
      
=== myPBX: Show login screen after logout instead of closing window ===
=== SIP: Handling of 491 response on reliable transport ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64194 64194]
|[http://mantis.innovaphone.com/view.php?id=68652 68652]
|}
|}
files: client.js<!---->
Re-try of re-INVITE was missing.<!---->
      
      
=== DHCP configuration with secondary but no primary gatekeeper not possible ===
=== SIP: Treat domain part of SIP URI case-insensitive ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64232 64232]
|[http://mantis.innovaphone.com/view.php?id=68653 68653]
|}
|}
This kind of configuration is required in some redundancy scenarios. It could be configured manually on the phones but not distributed via the innovaphone DHCP server. Keep in mind that only new telephone firmware will understand a DHCP configuration providing the secondary gatekeeper only.  <!---->
According to RFC3261(19.1.4)<!---->
      
      
=== Out Of Memory Trap when running VM without prompt files ===
=== SIP: Fix for REFER handling ===


{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64243 64243]
|[http://mantis.innovaphone.com/view.php?id=68663 68663]
|}
|}
When calling into a Voice Mail object without prompt files<br/>memory objects are allocated at high rate without being freed.<br/>Memory is freed at disconnect.<br/>This may cause a OOM trap when call stays connected for a longer time.<br/><!---->
Support for attended transfer between two calls from/to different SIP endpoints.<br/>As long as both calls belong to the same signaling interface.<!---->
      
      
=== IP241: Number information interfere with status information on call ctrl ===
=== SIP: Problem decoding INFO(application/dtmf-relay) ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64250 64250]
|[http://mantis.innovaphone.com/view.php?id=68667 68667]
|}
|}
Truncate number information to match screen metrics.<br/>Show status information if space available only.<!---->
DTMF digit was not decoded from message body if whitespace between EQUAL and DIGIT.<br/>E.g. Signal= 5<!---->
      
      
=== IP241 - VLAN frames &gt; 1514 bytes were not accepted ===
=== Phone: Changing config option /sip-hold does not call for reset ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64251 64251]
|[http://mantis.innovaphone.com/view.php?id=68691 68691]
|}
|}
The default maximum receive length of 1518 does not permit for a 1516 byte frame.<br/>With the maximum receive length set to 1536 such frames are received. <!---->
Reset is required and 'reset required" must be displayed.<!---->
      
      
=== DECT: File upload for user import ===
=== SIP: Web UI for cause code mapping (fix) ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64264 64264]
|[http://mantis.innovaphone.com/view.php?id=68772 68772]
|}
|}
File upload fixed. Used for user import by an OEM device.<!---->
Did not work as expected<!---->
      
      
=== myPBX launcher: Open configuration dialog if no URL is configured ===
=== SIP: Out-Of-Memory trap if Group Indications are enabled ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64265 64265]
|[http://mantis.innovaphone.com/view.php?id=68804 68804]
|}
|}
On startup the config dialog should show up if no URL is configured, yet.<br/><br/>files: ctiwin.cs<!---->
If 'Group Indications' are enabled on a PBX object where a SIP client registers,<br/>an OOM trap may occur in case of heavy call activity on PBX.<br/><!---->
      
      
=== PBX: Trap on security denial of 'unknown' registration ===
=== Phone: "Keep Calling Party Info on Pickup Key" option from Phone-&gt;Preferences not working, fixed ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64269 64269]
|[http://mantis.innovaphone.com/view.php?id=68820 68820]
|}
|}
If there is a 'unknown' registration attempt with wrong password (Unknown Registrations - With Admin Pwd only enabled) another attempt is rejected during security hold off period. If during this period another registration attempt with correct password happens, the PBX traps.<!---->
Phone: "Keep Calling Party Info on Pickup Key" option from Phone->Preferences not working, fixed<!---->
''Status:''
pbx.cpp
      
      
=== X.509: Use random serial numbers for self-signed certificates ===
=== Phone: DnD Absence Message containing newline dumped to config-file breaks phone upon config-file upload ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64279 64279]
|[http://mantis.innovaphone.com/view.php?id=68898 68898]
|}
|}
Currently the serial numbers start at 0 after long reset and are incremented by 1 for each new certificate. This causes collisions with old certificates that were created before the long reset.<br/><br/>Use a random number in order to avoid collisions after long reset.<br/><br/>Files: x509.cpp, x509.h<!---->
Solution: All flashdir entries cotaining newline now dumped as binary.<br/><br/><!---->
      
      
=== SIP: P-Asserted-Identity in UPDATE not working ===
=== IP30x V9 hf2 media problem: Echo canceller not reliable ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64289 64289]
|[http://mantis.innovaphone.com/view.php?id=68901 68901]
|}
|}
PAI with changed remote party identification was not handled<br/>if also Session-Expires header was present in UPDATE request.<!---->
Sometimes voice from ISDN/Analog to the IP is muted after some time.<br/>Seems to be a problem in the latetest echocanceller.<br/>Change to old DSP code until fixed DSP code is available.<br/><!---->
''Status:''
ip24.mak
      
      
=== DNS/Hosts: SRV-Record, Port Default 5060 Confusing. Removed. ===
=== Denial of Service filter in ethernet library did not work ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64312 64312]
|[http://mantis.innovaphone.com/view.php?id=68907 68907]
|}
|}
Edit field is now going be empty<!---->
this filter can be useful to prevent DOS attacks on non routing devices <!---->
      
      
=== IP241: IP230X not working ===
=== SIP: Trap when handling SUBSCRIBE on federation interfaces ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64355 64355]
|[http://mantis.innovaphone.com/view.php?id=68976 68976]
|}
|}
wrong UART used<!---->
SIP client object may be deleted while DNS query is pending.<br/>But only under critical timing conditions.<br/><!---->
''Status:''
ip230x.cpp phone_orchid.cpp config.h
      
      
=== supress "Send Number" for calls triggered by a 'Dial' function key with 'Send as Control Call' checked ===
=== myPBX: Use display name for sending meeting urls and conference numbers ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64365 64365]
|[http://mantis.innovaphone.com/view.php?id=68992 68992]
|}
|}
When using a 'Dial' function key with 'Send as Control Call' checked to control a call recording device the unique original calling party number must be passed to the recorder. The 'Send Number' configured in the the PBX user object may be the same for a group of phones and does not identify a certain phone.<!---->
For normal chat messages the DN is used as the sender name. For sending the links for application sharing or audio conferences the CN was used.<!---->
      
      
=== Trunk Park/Pickup (line keys) did not work anymore ===
=== Phone: Hide calling party on Pickup key fixed ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=64373 64373]
|[http://mantis.innovaphone.com/view.php?id=68995 68995]
|}
|}
Collateral damage from fix<br/><br/>fix: #61590: PBX: Boolean Function Key was not updated when joining group<!---->
Phone: Hide calling party on Pickup key fixed<br/><!---->
''Status:''
pbx_gi.cpp<br/>pbx_gi.h<br/>pbx.cpp (v9 only)<br/>pbx.h (v9 only)<br/>pbx_api.h (v9 only)
      
      
=== IP200A IP230 IP240: manufacturing test not working ===
=== SoftwarePhone: No CT setup with remote connected calls ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=64378 64378]
|[http://mantis.innovaphone.com/view.php?id=68996 68996]
|}
|}
write fixed bootheader fails<!---->
The call transfer setup facility is removed in the call setup if the call is a by remote control connected call used in case of outgoing calls with myPBX. This fixes an empty diverting party number information element in the PBX. Now it can be used with a trunk PBX object with the enabled option 'Set Calling=Diverting No', otherwise the calling party number was removed within this object.<!---->
''Status:''
flash_firmware.cpp
      
      
=== IP241!: IP230X not working ===
=== PBX: Conference trap ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64376 64376]
|[http://mantis.innovaphone.com/view.php?id=68998 68998]
|}
|}
wrong UART used<!---->
A zero pointer trap in the broadcast conference PBX object is fixed.<!---->
''Status:''
ip230x.cpp phone_orchid.cpp config.h
      
      
=== VM, &lt;prep-callinfo&gt; doesn't apply with &lt;pbx-xfer&gt; ===
=== IP222/IP232: Hookswitch on some PCB not working ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64384 64384]
|[http://mantis.innovaphone.com/view.php?id=69001 69001]
|}
|}
VM, <prep-callinfo> doesn't apply with <pbx-xfer><!---->
Reduce LED frequency from 12kHz to 6 kHz<!---->
''Status:''
orchid_lcd.cpp
      
      
=== memory leak check missing for last parked call info ===
=== phone_orchid - builtin speaker test did not work ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=64445 64445]
|[http://mantis.innovaphone.com/view.php?id=69026 69026]
|}
|}
when a call is parked using the 'Park' function key info about the parked call is kept for later checks when the call is unparked again. the leak check for this info was missing. <!---->
works only over channel 0<!---->
      
      
=== Phone trap when leaving User-Choose screen with Menu key ===
=== myPBX: Login did not work with system names containing special characters ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=64448 64448]
|[http://mantis.innovaphone.com/view.php?id=69033 69033]
|}
|}
problem: Phone trap when leaving User-Choose screen with Menu key. Missing "SaveChages?" screen. <br/><br/>solution: fixed in code. TXT_EMPTY==SvgChg Id. Worst Case occured.<br/><br/>files: phone_user_config.cpp<br/><br/>products: All Telephones<br/><br/>risks: none<br/><!---->
Missing URL encoding when sending the connect-request message.<!---->
      
      
=== PBX CDRs: Dummy call to endpoint because of SOAP UserCall generated CDR ===
=== PBX Mobility: Mobility object hanging if call establishment with SOAP/myPBX canceled ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=64449 64449]
|[http://mantis.innovaphone.com/view.php?id=69052 69052]
|}
|}
The PBX-SOAP UserCall function initiates a call to the endpoint and transfers this call to the destination given by UserCall. This initial call to the endpoint should not generate a CDR<!---->
If a call for a mobile phone is initiated by SOAP or myPBX, a call is first sent to the mobile phone. If the mobile phone accepts the outgoing call to the destination is initiated. If the mobile phone did not accept the initial call, no other calls could be done from then on.<!---->
''Status:''
pbx_xml.cpp
      
      
=== Phone: trap when redefining function key (long press access) from destination to partner ===
=== IP-DECT: Configuration of Media preferences did not work anymore ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=64498 64498]
|[http://mantis.innovaphone.com/view.php?id=69056 69056]
|}
|}
problem: Phone traps when redefining function key (long press access) from e.g. destination number to partner<br/><br/>solution: fix in code<br/><br/>files: phone_user_func.cpp<br/><br/>products: All Telephones<br/><br/>risks: None<br/><br/><br/><!---->
Media preference were ignored. This was a collateral damage introduced with fixes from the last hotfix<!---->
   
= V9 Hotfix 4 (90600.04) =
Changes included in Version 9 hotfix4
[http://mantis.innovaphone.com/view.php?id=69495 Definition]
 
== New Features ==
 
 
      
      
=== SIP: Trap when mypbx changes presence and external uc is configured ===
=== Improved protection against Denial of Service attacks ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=64504 64504]
|[http://mantis.innovaphone.com/view.php?id=69166 69166]
|}
|}
Trap when mypbx changes presence and external uc is configured.<!---->
flooding a box with different kinds of packets may lead to out of memory conditions. The Denial of Service filter in the ethernet layer is activated where required. TCP listening sockets have a backlog limit now. The http service restricts the number of half-open sessions and limits the number of concurrent sessions according to the total memory available on a box.<!---->
      
      
=== PBX: Failover to WAN for calls to Slave PBX did not work ===
=== ISDN interop issue with SecuGATE LI 30 from Sirrix ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=69168 69168]
|}
|}
accidentally broken in v9<!---->
The SecuGATE LI30 is sending/receiving ISDN INFO messages in Call Proceeding State (State 3 and state 9), which was not supported<!---->
''Status:''
pbx.cpp<br/>pbx.h
      
      
=== IP241: Directory entries did not show number tag ===
=== PBX Mobility: Support of transfer on mobile side, not using mobility ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=64555 64555]
|[http://mantis.innovaphone.com/view.php?id=69275 69275]
|}
|}
Number tag indicates type of number (M = mobile number)<!---->
In case that the mobile phone transfers the call to another destination, this call must be removed from the mobility function, so that the mobility function is available for another call<!---->
      
      
=== IPVA, Include NAT ===
=== USB Driver merge from v10 to v9 ===


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|[http://mantis.innovaphone.com/view.php?id=69288 69288]
|}
|}
Include the NAT0 module as in other innovaphone products<!---->
First step to support USB headsets. This is not functional yet.<!---->
      
      
=== PBX: Reporting licensing for standby operation did not work ===
=== SIP: New config file option /hold-notify-as-inactive ===


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|[http://mantis.innovaphone.com/view.php?id=69293 69293]
|}
|}
Reporting did not work for endpoints registered in standby mode, because of license check failed<!---->
If set, holdNotific is interworked into "a=inactive".<br/>If not, holdNotific is interworked into "a=sendonly".<!---->
''Status:''
pbx.cpp
      
      
=== Timeout when calling Mobile endpoint which does not send alert ===
=== SIP: New config file option /prefer-pai2 ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=64563 64563]
|[http://mantis.innovaphone.com/view.php?id=69459 69459]
|}
|}
Some SIP carriers do not send correct alert but only something which can be translated to CALL-PROC. In this case the CALL-PROC was not forwarded to the caller and therefore the call timed out after 12s<!---->
Interoperability with Telepo:<br/>When receiving INVITE, get calling party id from second P-Asserted-Id header.<!---->
''Status:''
pbx_mobility.cpp
      
      
=== User Interface: OK, Cancel, Apply buttons inconsistent ===
=== new: DHCP manufacturer specific option 'boot-cfg' provides an URL to read config from after any reset except 'creset' ===


{|
{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=64567 64567]
|[http://mantis.innovaphone.com/view.php?id=69472 69472]
|}
|}
The sequence is sometimes different. Should always be OK, Cancel, Apply<!---->
This option permits to boot boxes with a fresh config provided via TFTP/HTTP without storing the config on the device.<br/>It is intended to be used as follows:<br/><br/>1. the box is started with DHCP enabled (no initial configuration)<br/>2. the box contacts the DHCP server and gets the ip-address and also the Vendor Specific Information in option 43.<br/>  Suboption 249 of the Vendor Specific Information specifies the URL of the boot config file.<br/>3. the box polls the TFTP/HTTP server for the config file.<br/>4. the box reads the config file and executes the commands provided in the file<br/><br/>The URL may contain the same meta-character strings an Update Server URL, for example #m (mac-address)<br/>The length of the URL in the DCHCP suboption is restricted to 127 characters.<br/><br/>The URL is polled in 5 second intervals.<br/>The config file is read and executed by the update process in the usual way.<br/>A 'creset' commmand as last command of the file will restart the box with the new configuration without writing any 'config' command options to the flash.<br/>After a restart by the 'creset' commmand the boot-cfg URL is ignored.<br/>After a restart by any other of the 'reset' commands or by a power cycle the boot-cfg URL is processed again.<br/><br/>On an Innovaphone DHCP-Server configuration of a boot-cfg URL and providing it to clients via suboption 249 must be explicitely enabled by<br/>    config add DHCP0 /boot-cfg<br/>    config write<br/>    config activate<br/>Once enabled the URL may be entered under "IP4/ETH0/DHCP-Server/Boot Config URL" and is provided to all clients then.<br/><br/>If an Innovaphone DHCP-Client receivess a boot-cfg URL it is displayed under "IP4/ETH0/DHCP/Boot Config URL".<br/><br/><!---->
''Status:''
pbx_edit_allow.xsl<br/>pbx_edit_cfs.xsl<br/>pbx_edit_fork.xsl<br/>pbx_edit_groups.xsl<br/>pbx_edit_object.xsl<br/>pbx_edit_presence.xsl<br/>pbx_edit_profiles.xsl<br/>pbx_edit_rights.xsl
      
      
=== HTTP-Client: Trap in minifirmware on requests using a domain name ===
=== auto complete dtmf feature codes with '#' after 2 seconds ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=64568 64568]
|[http://mantis.innovaphone.com/view.php?id=69561 69561]
|}
|}
The HTTP client does not check if the DNS module exists before using it. In the minifirmware there is no DNS module.<br/><br/>Files: httpclient_i.cpp<br/><br/>Test: 9.00/unit-tests/httpclient/basic/150-get-dns-missing<!---->
Optional feature for phones, which are not able to send a '#', e.g. the iPhone.<br/>They dial a feature like a cfu with a destination number and after two seconds, the feature code is automatically completed with a '#'.<!---->
      
      
=== A PBX user with "Full PBX Administration" Rights could not edit phone configuration ===
=== PBX: HTTP request to initiate call for mobile phone ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=69570 69570]
|}
|}
The configuration pages could be opened once but after changing an item the input was disabled<!---->
To improve GSM client functionality<!---->
      
      
=== PBX: IM sessions to external UC did not work ===
=== PBX: WSDL for SOAP API readable from box ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=64583 64583]
|[http://mantis.innovaphone.com/view.php?id=69688 69688]
|}
|}
The session was not initiated towards the external UC<!---->
Simplifies Application development in some environments<!---->
''Status:''
pbx.cpp<br/>pbx_uc.cpp
      
      
=== IP6010, IP6000: Use optimized memcpy ===
=== SIP: Get display information from Call-Info header in register refresh response ===


{|
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|[http://mantis.innovaphone.com/view.php?id=64587 64587]
|[http://mantis.innovaphone.com/view.php?id=69758 69758]
|}
|}
Use of load/store multiple and shifts for 32 bit alignment speeds up memcpy by a factor of approx 2<br/><br/>Orginal memcpy<br/><info product="IP6010" mips="800Mips"><br/><memcpy bytes="1000000" time="2ms" speed="347.826Mbyte/s"/><br/><read bytes="1000000" time="2ms" speed="347.826Mbyte/s"/><br/><write bytes="1000000" time="2ms" speed="470.588Mbyte/s"/><br/><stack_memcpy bytes="1000000" time="7ms" speed="133.333Mbyte/s"/><br/><uncached_memcpy bytes="1000000" time="41ms" speed="24.169Mbyte/s"/><br/><aes bytes="1000000" time="135ms" speed="7.373Mbyte/s"/><br/><sha bytes="1000000" time="70ms" speed="14.260Mbyte/s"/><br/></info><br/> <br/>Optimized memcpy:<br/><info product="IP6010" mips="800Mips"><br/><memcpy bytes="1000000" time="1ms" speed="888.888Mbyte/s"/><br/><read bytes="1000000" time="2ms" speed="347.826Mbyte/s"/><br/><write bytes="1000000" time="2ms" speed="421.052Mbyte/s"/><br/><stack_memcpy bytes="1000000" time="7ms" speed="142.857Mbyte/s"/><br/><uncached_memcpy bytes="1000000" time="15ms" speed="64.000Mbyte/s"/><br/><aes bytes="1000000" time="138ms" speed="7.200Mbyte/s"/><br/><sha bytes="1000000" time="70ms" speed="14.285Mbyte/s"/><br/></info><br/><br/>CPU load with the test test/9.00/box/dsp/ip6010 shows approx 1% lower CPU load.<br/>Enet test test/9.00/box/enet/ip6010 shows 10638Kbyte/s transfer rate, compared to 9708Kbyte/s with the old memcpy.<br/><br/>With ECC enabled the CPU load was 19% / 21% without SRTP and 31% / 33% with SRTP<br/>With ECC Enet test test/9.00/box/enet/ip6010 shows 10638Kbyte/s transfer rate10309<!---->
Get display information from Call-Info header in 200/OK<!---->
''Status:''
ip6010.mak ip6000.mak arm.mak box/arm/memcpy.S<br/><br/>v8: ip6010.mak, box/box.mak, box/memcpy.S
      
      
=== PBX: Publish to external UC, From header wrong ===
=== IP-DECT: Display update ===


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|[http://mantis.innovaphone.com/view.php?id=69770 69770]
|}
|}
A From header of anonymous was sent<!---->
Now, the endpoint's display is updated if the registration forwards a display update.<!---->
''Status:''
pbx.cpp
      
      
=== myPBX: Detect standby cases in a different way ===
=== Voicemail &lt;pbx-query-obj&gt; queries new properties ===


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|}
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Needed because the behaviour of api_other_location has changed (#64540)<br/><br/>files: pbx_client.cpp<!---->
"user_type": returns "1"(endpoint) or "2"(other object like gateway)<br/><br/>"pseudo_type": returns types alike "vm", "bc_conf", "trunk"<br/><br/>"pseudo_text": returns friendly texts alike "Voicemail", "Bc Conference", "Trunk Line"<br/><br/><!---->
   
== Bug Fixes ==
 
 
      
      
=== A Bootcode Update could disrupt the Media stream for some seconds ===
=== Kerberos: Protect against ping pong attacks ===


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This was observed on phone devices with relatively slow flash memory when a bootcode update took place while a call was active.   <!---->
Do not answer with an error message to unexpected or malformed messages.<br/><br/>This protects against the "Kerberos Server Spoofed Packet Amplification DoS" attack. The attack causes two Kerberos servers to send each other error messages in a ping pong style.<!---->
      
      
=== Bootcode: Support for RGB LED added ===
=== RTP-DTMF: Start handling of RTP-DTMF on reception of END event ===


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|}
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...<!---->
Workaround for Bug in MS Lync / Mediation server.<br/>Mediation server changes destination port while sending RTP-DTMF redundancy retransmissions.<!---->
''Status:''
box.h boot.c boot.h flash.c flash_firmware.c soc_init.c platform_comcerto.c boot_comcerto.mak
      
      
=== phone: picking up a call failed sometimes ===
=== SIP: Presence interoperability with ESTOS UC server ===


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|}
|}
Sometimes pressing the partner, pickup or park key to pick up an alerting or parked call had no effect.<!---->
'Do Not Disturb' signaling without presence/tuple/contact element.<!---->
      
      
=== phone: prevent the pc port of the ethernet switch from receiving frames directly from the phone firmware ===
=== SIP: Support for Mediation Server Cluster ===


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In some cases is not desired that frames sent by the phone firmware via the cpu port are recieved by the pc port. This may be prevented now by<br/>    config add ETH0 /isolate-pc  <!---->
Load balancing and fail-over acc. to spec.<!---->
      
      
=== PBX User Interface did not work with Groups containing XML reserved characters (&amp;amp;,&lt;,&gt;,...) or non-ascii ===
=== Memory leak with each new telnet session ===


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|}
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XML or URI encoding was missing in some paces. The browser could not display the page.<br/>This happend when using the left PBX/Group tree for nvigation<!---->
Memory leak with each new telnet session (last packet upon exit not cleaned up)<br/><br/><!---->
''Status:''
pbx_admin.cpp<br/>pbx_objs_left.xsl<br/>pbx_objs_right.xsl
      
      
=== IP222/IP232 Bootcode: Support for hardware 300 ===
=== SIP: Media negotiation problem ===


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|}
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USB-Clock is generated from PRE-USB Clock, needs a 25 pre-divider<!---->
On media-relay with exclusive codec we can answer an incoming SDP offer right away without passing to app.<!---->
''Status:''
startup_orchid.s
      
      
=== TLS: Error on processing huge handshake messages ===
=== PPP connection fails after LCP renegotiation with different authentication methods (PAP -&gt; CHAP) ===


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|}
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The current implementation does not work with handshake messages that are bigger than 8 kilobytes. Especially the CertificateRequest message that is used for MTLS can be bigger.<br/><br/>files: tls.cpp<!---->
the active authentication entity was not stopped when another authentication was started and and signalled layer down after some timeout <!---->
      
      
=== PBX v5 SoftwarePhones licenes did not work on v9 or v8 PBX ===
=== PBX Twinning: Blind transfer to other phone in a twin phone configuration was not possible ===


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|}
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An old v5 SoftwarePhone license installed on a v9 PBX did not work for v5 SoftwarePhones<!---->
Only normal call to other phone or consulation call was supported<!---->
''Status:''
inno_lic.cpp<br/>inno_lic.h
      
      
=== DECT: Registration string for test registration ===
=== Web-UI: Fixed layout of DynPBX configuration ===


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The registration string for the test registration is fixed to work with V9.<!---->
Use min-width for fieldset.left<!---->
      
      
=== PBX:OEM Voicemail license did not work ===
=== IPVA didn't run on vSphere5 ===


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|}
|}
collateral damage from supporting licenses from different versions on a PBX<!---->
IPVA didn't run on vSphere5<!---->
''Status:''
inno_lic.cpp
      
      
=== SIP: Interworking of "Q.931 CALL PROCEEDING" into "183 Session Progress" ===
=== SIP: Interoperability of INFO(application/dtmf-relay) with Polycom ===


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Required if only CALL PROCEEDING and no ALERTING is received.<!---->
Fail to decode DTMF signal, since "application/dtmf-relay" body does not contain any CRLF.<br/>While CRLF is required according to "SIP INFO Package for DTMF".<br/><!---->
      
      
=== IP241: Diverting/Transfering party was not shown ===
=== IP222/IP232: residual echo  in handset mode ===


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If name (not number) was called and call was diverted/transferred<br/>the diverting/transferring party (dialed name) was invisible.<!---->
Change Codec config <!---->
''Status:''
ac_codec3.cpp
      
      
=== phone: Secretary/Executive function of Partner key does not work when "Subscribe for Dialog Info" is checked. ===
=== ip22/24/28/302/305: DSP affected by certain fax tones causing a trap ===


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When "Partner Type" is 'Secretary' or 'Executive' and "Subscribe for Dialog Info" is checked the key does not work correctly because this functionalities require group indications. To prevent configuration errors "Subscribe for Dialog Info" must be supressed in this case. <!---->
Assert removed for testing<br/><!---->
''Status:''
ac_dsp3.cpp
      
      
=== Linux: Reset required links ===
=== PBX: Phone config was not sent to phone, if phone was power cycled shorty after registration ===


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|}
|}
Reset required links in the Linux XML pages fixed.<br/><!---->
The new registration after the power cycle was not detected as new registration but as re-transmission of the previous registration, so it was not reported to the PBX and no phone config was sent<!---->
      
      
=== ip800 trace telling wrong information about power source ===
=== CX0-Wave-Encoding sometimes produced white noise ===


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PCBs since V300 cannot detect POE power and trace therefore told 'not powered'.<!---->
CX0-Wave-Encoding sometimes produced white noise.<br/>Also a memory leak was eliminated.<!---->
''Status:''
ip800.cpp
      
      
=== Phone: When activating a callback through function-key, the stored number behind the function-key is also called ===
=== IP-DECT: Hanging call after call transfer ===


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problem: phone: When activating a callback through function-key, the stored number behind the function-key is also called. press function key invokes the callback, release function key in idle mode invokes idle-function-key action<br/><br/>solution: fixed in code: now ignoring function key invocations through function-key-release if popup active<br/><br/>files: app_fkey.cpp forms.* forms_impl.*<br/><br/>products: all telephones<br/><br/>risks: none<br/><!---->
The hanging call in dectradio is fixed which occurred if a user does an unattended call transfer to an unassigned number and the transferred call is not disconnected.<!---->
      
      
=== Linux: Fix address from ETH1 network ===
=== SRTP: Bad key exchange on H.323/SIP interworking ===


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|}
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Linux network configuration is not working if Linux gets a fix ETH1 network address. The default gateway address is wrong from ETH0. Now the network mask and the default gateway can be configured.<!---->
MKI was added in SDP due to uninitialized data object.<!---->
      
      
=== Incorrect rpcap timestamp after TRACE LOST messages ===
=== Pass DNS-TTL to SIP ===


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The RPCAP timestamp (Wireshark) after a TRACE LOST message was incorrect, as the TRACE LOST message contained an incorrect timestamp.<!---->
In order to do load-balancing SIP needs to ge hold on the TTL of DNS resource records.<!---->
      
      
=== PBX CDRs: Wrong Display Name sometimes with Pickup ===
=== SIP: Support of maddr parameter in redirect response for REGISTER ===


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when calls routed between locations were involved<!---->
Set when redirecting REGISTER.<br/>Read when processing redirect response.<!---->
''Status:''
pbx.cpp
      
      
=== PBX CDRs: Transfer not indicated if transfer initiated by myPBX ===
=== G726 codec obsolete ===


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|}
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in case of transfer with consultation<!---->
The G726 codec was rarely used (if ever) in real life. In addition there are signaling problems specially with DECT peers when G726 is selected. Thus G726 is removed from the list of supported coders in all products.<!---->
''Status:''
pbx_client.cpp
      
      
=== SIP: Interworking of calls with Q.931 Bearer Capability "Unrestricted digital information" rejected ===
=== IP222: Input ip addresses does not work ===


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|}
|}
Calls with Q.931 Bearer Capability "Unrestricted digital information" were rejected.<!---->
Entering an ip address on the phone's menu does not work.<br/>All digits are doubled.<!---->
      
      
=== PBX: Conference trap ===
=== IP222: Call reroute did not work ===


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|}
|}
A zero pointer trap in the broadcast conference PBX object is fixed.<!---->
Using redial key on in incoming not-connected call did not work.<!---->
      
      
=== PBX CDRs: On Transfer to Waiting, the "connected" message did not contain peer ===
=== PBX: Call Diversion/Forward to '-' did not always show expected result ===


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Happend with transfer with consultation to a waiting queue<!---->
A call diversion to a destination'-' can be used to explicitly no execute a diversion of this type. So if a user has an CFU to '-' and this diversion is valid for a given call (Filter, Boolean), the phone should ring.<br/><br/>In fact the call was rejected.<br/><br/>There was also a problem with CFB in case of "busy on ... calls"<!---->
''Status:''
pbx_wait.cpp
      
      
=== IPVA, RAM-Dependent Flash Limitation ===
=== IP-DECT: New radio BMC firmware PCS05Ak ===


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Flash Utilization must be limited with RAM <= 64MB<br/>To prevent loss of flash data when uploading IPVA v9hf1:<br/>1)Perform Maintenance/Config/Download<br/>2)Upload the new ipva.bin<br/>3)Perform Maintenance/Config/Upload with the step 1) config.<br/><br/>Note: <br/>-The flash directory space is 13MB with 64MB Ram<br/>-The max. flash directory space is available from on 192MB Ram<!---->
The new radio BMC firmware PCS05Ak for the IP1200 fixes a trap by the DECT system if more than 255 DECT users without an endpoint subscription are sent to it.<!---->
      
      
=== IP222: changes for new hardware ===
=== PBX: No CLIR on internal calls did not work for SOAP ===


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* Reduce average current through IR-LED <br/>* Re-init LCD orientation (fight against resets of the display controller)<!---->
If the features "No CLIR on internal Calls" is activated on a PBX a CLI is sent to the called phone even if the call was sent with "CLI presentation restricted". The same should be case on SOAP/TAPI when monitoring this user.<br/><br/>Now when "No CLIR on internal Calls" is enabled all number information available is provided on SOAP.<!---->
''Status:''
orchid_lcd.cpp
      
      
=== Setup wizard used wrong default password on some products ===
=== PBX: Reject calls without media, if no known facility ===


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If no password was entered on<br/>- ip6010<br/>- ip2010<br/>- ip1060<br/>- ip302<br/>- ip305<br/>- ip6000<br/>- ip2000<br/>A wrong password was configured<!---->
Fixes compatibility issues between versions. For example presence subscription sessions from v8 phones being forwarded to voicemail<!---->
''Status:''
ip6000/config_wizard.txt<br/>ip24/config_wizard.txt<br/>ip6010/config_wizard.txt
      
      
=== myPBX: Usability improvements for call transfers ===
=== PBX Waiting: Not possible to send DTMF to Waiting Queue from myPBX Numeric Keyboard ===


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|}
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Blind transfer of incoming alerting calls.<br/>Transfer of outgoing alerting calls.<br/>Autoselect new outgoing calls.<br/>Hide transfer buttons on the selected call.<br/><br/>files: client.js<!---->
The Waiting Queue prohibited the facilities used for this to be sent to calling phone<!---->
      
      
=== PBX Mobility: Use of RTP-DTMF was prohibited even if media terminated in PBX itself ===
=== PBX: Filter for internal or external calls at CFs did not work CFB or CFNR if call already diverted ===


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For DTMF within a call (e.g. ** for R-Key) RTP-DTMF cannot be used end-to-end, because the Mobility object would not see these DTMFs.<br/><br/>If media is terminated in the PBX (because of media-relay or whenever tones are played by the PBX itself) RTP-DTMF can and should be used<!---->
Problem:<br/><br/>User A has CFU to User B<br/>User B has CFNR for ext. Calls only to User C<br/><br/>An internal call to A was diverted to B (ok) and after no response diverted to C (nok)<!---->
''Status:''
channel.cpp
      
      
=== PBX CDRs: Wrong information if forward on other PBX ===
=== PBX Waiting: In case of "Announcemen w/o Connect" together with Alert-Timeout 0, DTMF dialing was not possible ===


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|}
The leg1 info from the other slave was not evaluated for CDRs<!---->
In this case as only response to the incoming SETUP a PROGRESS was sent. This meant, that the caller was still in overlap dialing state, so a phone does not send DTMF, but translates input keys to INFO dialing messages.<br/><br/>A CALL-PROC is now sent before PROGRESS, which terminates the dialing.<!---->
''Status:''
pbx.cpp<br/>ep_lib.cpp<br/>ep_lib.h<br/>fty.cpp<br/>fty.h<br/>h450.cpp<br/><br/>
      
      
=== Sockets: misleading trace: connect &amp;amp; bind events carry uninitialized IP6 addr ===
=== SIP: Send BYE with Reason header with "Q.850 Recovery on timer expiry" ===


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problem: sockets: misleading trace: connect &amp; bind events carry uninitialized IP6 addr<br/><br/>solution: now set to 0, fixed in code<br/><br/>files: types.h socket.h<br/><br/>products: all products<br/><br/>risks: none<br/><!---->
If session refresh is outstanding the call is released with BYE with "Reason: Q.850;cause=16".<br/>Better send BYE with "Reason: Q.850;cause=102"<!---->
      
      
=== ip241 - G722 3party conference did not work ===
=== PBX Waiting: No ringback when doing two-stage dialing to a Gateway/Trunk object ===


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|}
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When a 3party conference was setup with at least one G722 connection involved the microphone data of one party was terminated at the conference point  <!---->
A local ringback is now switched on, when receiving ALERT from called party<!---->
      
      
=== DECT: Wrong packetsize sometimes used after renegotiation ===
=== SIP: Bug when decoding SRTP keys from SDP ===


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|}
|}
This could cause interop problems with some PBX, which was not able to receive media packets above a certian size.<!---->
Bug in decode base64.<!---->
''Status:''
h323ch.cpp
      
      
=== VM, Session Guard Timer Not Stopped For &lt;pbx-record&gt; ===
=== Web-UI: Input field to small for SIP-URI ===


{|
{|
Line 2,177: Line 2,103:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65097 65097]
|[http://mantis.innovaphone.com/view.php?id=69578 69578]
|}
|}
The following script didn't work correctly with v9: http://wiki.innovaphone.com/index.php?title=Howto:A_Simple_Recording_Solution_On-Top_of_The_innovaphone_Voicemail<br/><!---->
Input control on "UC" tab of External UC properties dialog was too small.<br/>Only part of SIP-URI was visible.<!---->
      
      
=== myPBX: Improvements of French translation ===
=== phone: assume an outbound call to be an external call if connected number info is missing in connect event ===


{|
{|
Line 2,188: Line 2,114:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65115 65115]
|[http://mantis.innovaphone.com/view.php?id=69581 69581]
|}
|}
Transfer direct -> Transférer<br/>Transférer -> Mettre en communication<br/><br/>files: pbx_client_localisation.cpp<!---->
In certain ISDN configurations the PBX can not provide the connected number info in the connect event for an outbound call. In this case the the call was assumed to be an internal call and consequently was not recorded when transparent recording of external calls was configured.<br/>Now an external call is assumed in this case.<br/> <br/><!---->
      
      
=== SIP: Use DNS query without recursion on closed federation interfaces ===
=== IP-DECT: Reset link ===


{|
{|
Line 2,199: Line 2,125:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65126 65126]
|[http://mantis.innovaphone.com/view.php?id=69584 69584]
|}
|}
Keep DNS server from doing automatic recursion when resolving domain names on closed federation interfaces.<!---->
The reset link of DECT System Config GUI page is fixed.<!---->
      
      
=== H.323: Don't forward G.729B capability if silience compreession not enabled ===
=== SIP: Do not check remote certificate name when calling PBX client with TLS ===


{|
{|
Line 2,210: Line 2,136:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65133 65133]
|[http://mantis.innovaphone.com/view.php?id=69598 69598]
|}
|}
This solves quality issues some SIP provider have with G.729B.<!---->
Check is reqired only when calling an unregistered client or when giving INVITE to registrar.<!---->
''Status:''
h323ch.cpp
      
      
=== DECT: Master Call GUI close/clear for OEM ===
=== phone: VLAN signaling priority could not be configured via phone menu ===


{|
{|
Line 2,223: Line 2,147:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65137 65137]
|[http://mantis.innovaphone.com/view.php?id=69633 69633]
|}
|}
The link 'close' and 'clear' of the DECT Master call page did not work for an OEM device and is fixed now.<!---->
Under "Menu/Administration/IP Settings/VLAN" there was only a "VLAN Priority" menu item. This menu item did override the 'Priority RTP Data' value but not the 'Priority Signaling' value as entered via WEB configuration.<br/>Now the items "Prio. RTP Data" and "Prio. Signaling" replace the "VLAN Priority" item.<!---->
      
      
=== Phone UI: Truncate h323id on header of idle screen to reserve space for e164 number ===
=== IPxx10-sata: trap after config /trace /track activation ===


{|
{|
Line 2,234: Line 2,158:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65140 65140]
|[http://mantis.innovaphone.com/view.php?id=69642 69642]
|}
|}
Long h323id's may take all available space and keep e164 number from being displayed.<!---->
Instruccion was accessing uninitialized pointer.<!---->
''Status:''
forms_color.cpp
      
      
=== Network Link GUI ===
=== IP222: Alpha input using the num block ===


{|
{|
Line 2,247: Line 2,169:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65142 65142]
|[http://mantis.innovaphone.com/view.php?id=69646 69646]
|}
|}
The network (ETH0/ETH1) Link page was not reloaded after pressing 'ok' or ' clear' in IP800 and IP1200 and is fixed now.<!---->
Input mode changes back to numeric in screen "Wahlvorbereitung".<br/>Should stay in alpha mode.<!---->
      
      
=== myPBX: Sessions sometimes get stuck ===
=== IP200: Manufacturing test fails is hwbuild is reprogrammed ===


{|
{|
Line 2,258: Line 2,180:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65143 65143]
|[http://mantis.innovaphone.com/view.php?id=69662 69662]
|}
|}
Session IDs used the full dword scope but btree_compare can only handle values in the positive integer scope.<br/><br/>files: pbx_client.cpp<!---->
Old hwbuild was not cleared, because on INCA devices the hwbuild is located on a higher address was not cleared.<!---->
''Status:''
flash_firmware.cpp
      
      
=== firmware of a certain image size did not boot from scattered flash segments ===
=== H.323: Media Negotiation problem with Lync interop and SRTP ===


{|
{|
Line 2,269: Line 2,193:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65154 65154]
|[http://mantis.innovaphone.com/view.php?id=69687 69687]
|}
|}
when booting from scattered flash segments the firware is decompressed in smaller chunks. A false alarm was raised after successful decompress for a last chunk smaller than 5 byte.<!---->
With a retrieve from the lync after hold (which is signaled as a reinvite with sendrevc) new media parameters were sent, containing new SRTP keys. These new media parameters cannot be used, on the PBX which is initiating new end to end media negotiation at the same time. These media parameter were not ignored properly.<!---->
      
      
=== PBX CDRs: Missing transfer-to on operator connect, when calling waiting from other PBX ===
=== phone_orchid: wrong volume setting when monitor mode is entered ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65178 65178]
|[http://mantis.innovaphone.com/view.php?id=69734 69734]
|}
|}
This was a general problem with transfer on another PBX. Happens with normal transfer on other PBX as well<!---->
when monitor mode ise entered by pressing the speaker key in a handset conversation the handsfree speaker is enabled in addition to the handset speaker.<br/>the volume was reconfigured with the wrong value.<!---->
''Status:''
pbx.cpp<br/>pbx_wait.cpp
      
      
=== Phone: Broken "Display Name on Pickup/Partner Key" Option ===
=== phone_orchid: Calls received with CLIR appear in call list with an empty entry ===


{|
{|
Line 2,293: Line 2,215:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65179 65179]
|[http://mantis.innovaphone.com/view.php?id=69737 69737]
|}
|}
problem: Broken "Display Name on Pickup/Partner Key" : pickup function key always displaying numbers, never H.323 alias (name)<br/><br/>solution: fixed in code<br/><br/>files: box/forms/lcd/phone_forms_impl.cpp phone/app/app_disp.cpp app_ctl.h<br/><br/>products: all inca telephones<br/><br/>risks: none<br/><br/><!---->
Calls received with CLIR or without a number/name appeared in call list with an empty entry; now either "anonymous" (CLIR) or "unknown" is displayed instead of a name<br/><br/><!---->
      
      
=== SIP-H323 calls with SRTP: No media after multiple Hold/Retrieve ===
=== SIP: Take SDP id and version as 64bit integer ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65185 65185]
|[http://mantis.innovaphone.com/view.php?id=69738 69738]
|}
|}
After first Hold/Retrieve there was no SRTP, after the next Hold/Retrieve very often no media<!---->
o-line of SDP offer/answer is defined as 64bit integer<br/>Interoperability with MX-ONE<!---->
''Status:''
h323ch.cpp
      
      
=== Phone UI: Truncate h323id on header of idle screen to reserve space for e164 number ===
=== Relay: Forward facilities to local destinations ===


{|
{|
Line 2,317: Line 2,237:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65258 65258]
|[http://mantis.innovaphone.com/view.php?id=69764 69764]
|}
|}
problem: Long h323id's may take all available space and keep e164 number from being displayed.<br/><br/>solution: new user pretty print algorithm now tries to display the whole e164 number and fills the remaining space left aligned with h323 id. ellipsis is only one char on the phone lcd and notes that the text ahead of it has been shortened. following cases are possible: <br/>1) |123...| number too long <br/>2) |123456| number just right<br/>3a)| 12345| no name found or less than 2chars left for name<br/>3b)|  1234|<br/>4) |n... 1| name shortened with ellipsis, number complete<br/>5) |nam 23| enough room for name and number<br/>6) |name  | no number - enough space for name<br/>7) |name01| no number - enough space for name (just right)<br/>8) |nam...| no number - name shortend with ellipsis<br/><br/> <br/>files: forms_impl.cpp<br/><br/>products: all inca telephones<br/><br/>risks: none<!---->
This fix is related to the previous fix #66629 for V9 hotfix2. Now, facilities are only forwarded, if the destination is a physical interface, not e.g. a SIP provider.<!---->
      
      
=== PBX: Busy on ... Calls should be executed on calls from on object to itself ===
=== SIP: Trap when handling NOTIFY(application/qsig) ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65260 65260]
|[http://mantis.innovaphone.com/view.php?id=69771 69771]
|}
|}
was disabled in 2008, no idea what this should be good for. Can be very confusing<!---->
Traps if no progress indicator present in tunneled DISCONNECT message.<!---->
''Status:''
pbx.cpp
      
      
=== PBX Broadcast: CFB configured at broadcast was always executed if "Execute member diversions" ===
=== WEB-UI Config Upload screen blank after upload of a file with a lot of failing lines ===


{|
{|
Line 2,341: Line 2,259:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65261 65261]
|[http://mantis.innovaphone.com/view.php?id=69812 69812]
|}
|}
If "Execute Member Diversions" was checked a call to Broadcast was also sent to CFB destination<!---->
When uploading a file with about 50 failing lines the screen was left blank without any info about the failing operation.<!---->
''Status:''
pbx_bc.cpp
      
      
=== IP241: Call list does not show display name if call was diverted ===
=== phone: "Keep Calling Party Info on Pickup Key" option from Phone-&gt;Preferences did not work with very long numbers/names ===


{|
{|
Line 2,354: Line 2,270:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65269 65269]
|[http://mantis.innovaphone.com/view.php?id=69816 69816]
|}
|}
Display name of the diverted-to party was not shown in call list.<br/>Also truncate names and numbers to avoid interference with date and time.<!---->
speciall when using this option in conjunction with the "Display Name on Pickup/Partner Key" option the calling party info was not correctly displayed<!---->
      
      
=== Gateway: Not possible to enter wildcards ('.') in interface maps ===
=== IP6010: SRTP using AES-192 and AES-256 did not work ===


{|
{|
Line 2,365: Line 2,281:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65280 65280]
|[http://mantis.innovaphone.com/view.php?id=69828 69828]
|}
|}
wrong check for correct value<!---->
Due to a bug in the encryption driver of the IP6010, only AES-128 worked on this platform.<!---->
''Status:''
gk.cpp
      
      
=== Gateway: Configured signaling port got lost, when ediiting interface maps ===
=== IP-DECT: Location update (OEM) ===


{|
{|
Line 2,378: Line 2,292:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65303 65303]
|[http://mantis.innovaphone.com/view.php?id=69863 69863]
|}
|}
The signaling port was reset to the standard port when saving interface mappings<!---->
Location update with message waiting information is fixed if the endpoint roams. This is only used by OEM devices.<!---->
''Status:''
gk.cpp
      
      
=== PBX: Name Identification was missing when initaiting IM Session ===
=== Logging: Threshold for error event "SRTP authentication failed" ===


{|
{|
Line 2,391: Line 2,303:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65304 65304]
|[http://mantis.innovaphone.com/view.php?id=69920 69920]
|}
|}
No display name of initiating party was sent<!---->
Error event was triggered at the very first decrypt failure.<br/>Some decrypt failure are expected during media re-negotiation.<br/>Trigger this error event after 10 decrypt failures in line.<br/><!---->
''Status:''
pbx.cpp
      
      
=== Trap in rarely used OS function bufman::remove ===
=== PBX Calls Page/SOAP wrong number ===


{|
{|
Line 2,404: Line 2,314:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65338 65338]
|[http://mantis.innovaphone.com/view.php?id=69921 69921]
|}
|}
could result in negative length of buffer<!---->
In a configuration with escapes for calls from a slave and a node not the root node and the call forwarded to the master, because the number could not be resolved locally, wrong escapes were added to the called number<!---->
''Status:''
os.cpp
      
      
=== phone: blinking of central MWI-LED could not be disabled for a MWI function key without an own LED ===
=== phone ip222: MWI LED not working ===


{|
{|
Line 2,417: Line 2,325:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65354 65354]
|[http://mantis.innovaphone.com/view.php?id=69924 69924]
|}
|}
When the MWI function key was assigned to a key without an own LED the central MWI-LED did blink although "Active State/LED" was set to "off".<br/> <!---->
The MWI LED was not switched on for pending voice mails and/or pickups<!---->
      
      
=== Associating a call from a Gateway object with Gateway object Node did not work if CGPN represented an incomplete Number ===
=== Trap during fax transfer ===


{|
{|
Line 2,428: Line 2,336:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65359 65359]
|[http://mantis.innovaphone.com/view.php?id=69950 69950]
|}
|}
This can result in not finding the correct Node Extern interface<!---->
Seen on Ip28, can happen on AC_DSP3 ( IP22/24/28/302/305/1060/301/6010 )<!---->
''Status:''
''Status:''
pbx.cpp
ac_dsp3.h
      
      
=== HTTP /home did not work for URLs containing more then one '/' ===
=== phone_orchid:  checkmark "Phone/User-x/General/Options/No DTMF Detection" has no effect ===


{|
{|
Line 2,441: Line 2,349:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65364 65364]
|[http://mantis.innovaphone.com/view.php?id=70114 70114]
|}
|}
/home could be used for PBX0/MY/start.htm, but this did nor work because of multiple '/'<!---->
If this checkmark is set DTMF digits entered via keyboard in a connected call shall be sent in-band as voice data, not encoded in RTP-DTMF packets as usual.<br/> <!---->
''Status:''
http.cpp
      
      
=== PBX UI: Registration with Admin Pwd misleading, PBX Pwd is better ===
= V9 Hotfix 5 (90600.05) =
Changes included in Version 9 hotfix5
[http://mantis.innovaphone.com/view.php?id=70208 Definition]
 
== New Features ==
 


{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=65370 65370]
|}
The Admin Pwd could be understood as the General/Admin password. PBX Pwd should be pretty clear what this is.<!---->
''Status:''
pbx_edit_object.xsl<br/>pbx_general.xsl
      
      
=== phone_orchid: handsfree mode does not work with  DSP Fimware 494E003ce4.640.11 ===
=== IP-DECT: Default frame size to 30ms ===


{|
{|
Line 2,467: Line 2,368:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65379 65379]
|[http://mantis.innovaphone.com/view.php?id=70140 70140]
|}
|}
the calling party was fully echoed when the called party was in handsfree mode.   <!---->
Now the default frame size is 30ms.<!---->
      
      
=== IP241: Time and date not refreshed while menu screen is active ===
=== SIP: Support for transparent message headers on transfer ===


{|
{|
Line 2,478: Line 2,379:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65394 65394]
|[http://mantis.innovaphone.com/view.php?id=70163 70163]
|}
|}
Refresh staus bar in menu screen<!---->
Interoperability with Telepo.<br/><!---->
      
      
=== PBX-SOAP: Trap if initiating multiple outgoing calls from a Waiting object at the same time ===
=== added support for USB tracing with Wireshark ===


{|
{|
Line 2,489: Line 2,390:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65418 65418]
|[http://mantis.innovaphone.com/view.php?id=70373 70373]
|}
|}
Some applications do this to deliver voice messages<!---->
USB traces are now supported by the debug class and can be opened with Wireshark and a new innovaphone.dll.<!---->
''Status:''
pbx_wait.cpp<br/>pbx_wait.h
      
      
=== SIP: Gateway interfaces without registration need another config option ===
=== SIP: Prefer P-Asserted-Identity with tel-URI ===


{|
{|
Line 2,502: Line 2,401:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65436 65436]
|[http://mantis.innovaphone.com/view.php?id=70417 70417]
|}
|}
New config option "Local Domain" added.<br/>Required for URI processing on incoming calls.<br/>If domain part of destination URI matches the local domain,<br/>the domain part is stripped from the destination information.<!---->
Prefer P-Asserted-Identity with tel-URI over that with sip-URI.<br/>In case there are two P-Asserted-Identity headers in INVITE.<!---->
      
      
=== Gateway: Record URL at SIP interface was lost when Internal registration was configured ===
=== Gateway: Forward Display Info received from ISDN Setup to H.323 ===


{|
{|
Line 2,513: Line 2,412:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65443 65443]
|[http://mantis.innovaphone.com/view.php?id=70562 70562]
|}
|}
UI problem<!---->
needed for compatibility with SecuGATE LI30<!---->
''Status:''
gk.cpp
      
      
=== VM, Project script didn't run for endpoints having "Send Number" configured ===
=== &lt;pbx-getcallinfo&gt; returning diversion reason ===


{|
{|
Line 2,526: Line 2,423:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65456 65456]
|[http://mantis.innovaphone.com/view.php?id=70697 70697]
|}
|}
VM, Project script didn't run for endpoints having "Send Number" configured<!---->
as<br/><pbx-getcallinfo out-leg2-reason="..."/><!---->
      
      
=== VM, email attachments weren't sent for https URLs ===
=== myPBX: Interface for IM provider ===


{|
{|
Line 2,537: Line 2,434:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65458 65458]
|[http://mantis.innovaphone.com/view.php?id=70867 70867]
|}
|}
VM, email attachments weren't sent for https URLs<!---->
Interface between the JavaScript client and the myPBX launcher.<!---->
''Status:''
IM to JS:<br/>prepare_dial_name(value)<br/>prepare_im(value)<br/>ep_request(name,number)<br/><br/>JS to IM:<br/>innovaphone_updateXml(xml)<br/>innovaphone_sessionInfo(domain,name,number,cn,dn)
      
      
=== PBX: Call-Intrusion could result in wrong name display ===
=== SIP: Diagnostics of transport error "SIP Overload" ===


{|
{|
Line 2,548: Line 2,447:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65462 65462]
|[http://mantis.innovaphone.com/view.php?id=70883 70883]
|}
|}
esspecially for silent intrusion<!---->
Limitation of buffer allocation<!---->
''Status:''
''Status:''
signal.cpp<br/>h450asn1.h<br/>pbx.cpp<br/>pbx.h
siptrans.cpp/h<br/>sipmsg.h
      
      
=== AD Replication, Alphanumeric symbols were rejected in maps ===
=== IP-DECT: License restriction for OEM device ===


{|
{|
Line 2,561: Line 2,460:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65499 65499]
|[http://mantis.innovaphone.com/view.php?id=70887 70887]
|}
|}
A bug introduced by code-adaption from v8 to v9. I.e. this problem is new with v9 and does not occur under v8.<br/><!---->
A license restriction for an OEM device is added. It is not used in the IP1200.<!---->
      
      
=== Kerberos: Do not allow registration of multiple databases for one realm name ===
=== SIP: Diagnostics of error "Registration expired" ===


{|
{|
Line 2,572: Line 2,471:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65589 65589]
|[http://mantis.innovaphone.com/view.php?id=70938 70938]
|}
|}
This happened when a box hosted multiple PBXes with the same system name.<br/><br/>files: <br/>kerberos_if.cpp<br/>kerberos_kdc.h (v9 only)<br/>kerberos_kdc.cpp<br/>kerberos_db.cpp<!---->
Missing registration refresh<!---->
''Status:''
sip.cpp/h<br/>
      
      
=== myPBX: Default values for configuration of external LDAP search ===
=== Possibility to clear the call list in mypbx ===


{|
{|
Line 2,583: Line 2,484:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65593 65593]
|[http://mantis.innovaphone.com/view.php?id=71202 71202]
|}
|}
Use the same default values as on the phone.<br/><br/>Name Attributes: "sn,givenName,company"<br/>Number Attributes: "telephoneNumber:D,homePhone:P,mobile:M,:@"<br/><br/>files: pbx_client.cpp<br/><!---->
A new button is now available in mypbx to clear the call list.<!---->
      
      
=== Phone: Bei "Nachricht antworten" wird von der gerade aktiven Reg gesendet ===
=== phone: LED mode of Join Group function key can be set both for idle and for active state ===


{|
{|
Line 2,594: Line 2,495:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65630 65630]
|[http://mantis.innovaphone.com/view.php?id=71247 71247]
|}
|}
Es sollte aber von der Reg gesendet werden, auf der die Msgs empfangen wurde.<br/>Wird ja bei Voice-Calls auch so gemacht.<!---->
sometimes the "not in group" state must be signaled as the exception<!---->
      
      
=== IP241: Connected call was displayed as transferred call ===
=== myPBX: Show implicit visibility in groups ===


{|
{|
Line 2,605: Line 2,506:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65641 65641]
|[http://mantis.innovaphone.com/view.php?id=71250 71250]
|}
|}
When name was dialed instead of number, the connected number (received with connect message) caused display to show transfer condition.<!---->
Active members of groups can see the presence and the calls of other group members. In order to make that clear to the user, now the visibility settings of myPBX show in what groups the user is visible.<!---->
''Status:''
phonesig.cpp
      
      
=== myPBX: Outgoing IM session did not reach state "up" on instant connect ===
=== IP-DECT: CSS changes for new OEM device ===


{|
{|
Line 2,618: Line 2,517:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65651 65651]
|[http://mantis.innovaphone.com/view.php?id=71282 71282]
|}
|}
The following message was missing when the remote entpoint connected the call right away: <update-im id="0" state="up"/><br/><br/>files: pbx_client.cpp, pbx_client.h<br/><br/><!---->
For a new OEM device with a changed style some CSS classes are added.<!---->
      
      
=== myPBX: Performance improvements ===
=== PBX: New User property 'Do not Disturb' ===


{|
{|
Line 2,629: Line 2,528:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65654 65654]
|[http://mantis.innovaphone.com/view.php?id=71439 71439]
|}
|}
Searching in lists was suboptimal<br/>Removed unused method<br/><br/>files: pbx_client.cpp, pbx_client.h<!---->
No calls are sent to the user if set.<br/><br/>Can be set by 'External UC' applications (e.g. Estos Procall)<!---->
      
      
=== Phone: Croatian translation completed (some missing entries) ===
=== PBX: Support up to six devices for a user ===


{|
{|
Line 2,640: Line 2,539:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65661 65661]
|[http://mantis.innovaphone.com/view.php?id=71506 71506]
|}
|}
problem: Croatian translation completed (some missing entries)<br/><br/>solution: translated<br/><br/>files: phone/text/*<br/><br/>products: all phones<br/><br/>risks: none<br/><!---->
Sometimes 4 devices are to little.<br/>A little bit of cleanup on the user interface was done as well with this change<!---->
      
      
=== IP241: Fix for display rendering ===
=== phone: Mic Off/On controllable via Soap:UserRc(&lt;call&gt;,14/15) ===


{|
{|
Line 2,651: Line 2,550:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65678 65678]
|[http://mantis.innovaphone.com/view.php?id=71721 71721]
|}
|}
Zeroize png object when image loading fails.<!---->
To allow Soap app's control of the mute key<!---->
      
      
=== IP241: Fix for multi line editor ===
=== SoftwarePhone: New Tray Icon ===


{|
{|
Line 2,662: Line 2,561:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65686 65686]
|[http://mantis.innovaphone.com/view.php?id=71969 71969]
|}
|}
Fix for multi line editor<!---->
The SoftwarePhone has got a new tray icon differs from the myPbx one, now.<!---->
   
== Bug Fixes ==
 
 
      
      
=== DECT: Trap during registration up handling ===
=== SIP: Message decoding error ===


{|
{|
Line 2,673: Line 2,576:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65698 65698]
|[http://mantis.innovaphone.com/view.php?id=70116 70116]
|}
|}
Trap in DECT Master fixed. It occurs if the master endpoint is in delete state and a RAS registration up event is received.<!---->
Only concerns Message headers whose value starts and with quotes, but are not quoted.<br/>E.g.<br/>  Referred-By: "Huvudnummer"<sip:400@abcdef.ghi;fnrid=1759>;from-tag=5decdf1a;to-tag=2515833546;org-cid="6afa95ede909d311906f00013e11cdb3@192.168.2.115"<br/><!---->
      
      
=== myPBX: Forward instant messages only once in order to prevent loops ===
=== Linux: VLAN GUI removed ===


{|
{|
Line 2,684: Line 2,587:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65699 65699]
|[http://mantis.innovaphone.com/view.php?id=70131 70131]
|}
|}
Forwarded messages are detected by comparing the remote endpoint and the sender name.<br/><br/>files: pbx_client.cpp<br/>test: unit-tests\\client\\iteration07\\402-multisession-loop2<!---->
It is not necessary to configure a VLAN to Linux. Now the GUI page Linux VLAN is removed.<!---->
      
      
=== myPBX: Use displayname as sender name of instant messages ===
=== ip24/ ip6010/phone_orchid: in-band ring back tone generation into voice channel did not work ===


{|
{|
Line 2,695: Line 2,598:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65704 65704]
|[http://mantis.innovaphone.com/view.php?id=70146 70146]
|}
|}
The cn was used before. Now one of the following attributes is used: dn, cn, h323<br/><br/>files: pbx_client.h, pbx_client.cpp<br/>test: unit-tests\\client\\iteration07\\250-singlecall-dn<!---->
sometimes a ring back tone must be sent in-band to an ISDN channel. the tone was generated but passed to the wrong destination. <!---->
      
      
=== IP241: Trap on display rendering ===
=== SIP: Problems with CLEARMODE ===


{|
{|
Line 2,706: Line 2,609:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65726 65726]
|[http://mantis.innovaphone.com/view.php?id=70149 70149]
|}
|}
When entering too many digits while preparing block dialing.<!---->
CLEARMODE was not offered as prefered codec outgoing INVITE.<br/>Destination accepted G711a instead of CLEARMODE:<br/><br/>Offer:<br/>  v=0<br/>  o=- 18 1 IN IP4 172.16.66.77<br/>  s=-<br/>  c=IN IP4 172.16.66.77<br/>  t=0 0<br/>  m=audio 16422 RTP/AVP 4 18 8 0 96 97 101 13<br/>  a=rtpmap:96 G726-32/8000<br/>  a=rtpmap:97 CLEARMODE/8000<br/>  a=rtpmap:101 telephone-event/8000<br/>  a=fmtp:18 annexa=yes<br/>  a=fmtp:18 annexb=no<br/>  a=fmtp:101 0-15<br/>  a=ptime:20<br/><br/>Answer:<br/>  v=0<br/>  o=cp10 131833822144 131833822144 IN IP4 172.16.66.175<br/>  s=SIP Call<br/>  c=IN IP4 172.16.67.132<br/>  t=0 0<br/>  m=audio 33526 RTP/AVP 8 0 97<br/>  b=AS:75<br/>  a=rtpmap:8 PCMA/8000/1<br/>  a=rtpmap:0 PCMU/8000/1<br/>  a=rtpmap:97 CLEARMODE/8000/1<br/>  a=ptime:30<br/><!---->
      
      
=== Phone: Message reply uses current registration instead of registration which received the incoming message ===
=== IP-DECT: Master radio list sync highlighting ===


{|
{|
Line 2,717: Line 2,620:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65729 65729]
|[http://mantis.innovaphone.com/view.php?id=70150 70150]
|}
|}
problem: Message reply uses current registration instead of registration which received the incoming message. Now message sending functions also take message source as parameter.<br/><br/>solution: fixed in code. <br/><br/>files: phone/app/* phone/list/* phone/dir/*<br/><br/>products: All Telephones<br/><br/>risks: none<!---->
In the IP-DECT Master radio list the active sync highlighting is fixed, now.<!---->
      
      
=== IP241: Fix for XML screen dump ===
=== SIP: No update of name presentation after transfer on QSIG interworking ===


{|
{|
Line 2,728: Line 2,631:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65737 65737]
|[http://mantis.innovaphone.com/view.php?id=70155 70155]
|}
|}
XML screen dump is only used for automated testing.<!---->
Fix for QSIG/SIP interworking.<br/>Display name of callTransferComplete was not passed to SIP.<!---->
      
      
=== MWI does not work in various Node/Pbx combination ===
=== IP222/IP232: Handset  volume with 90772 andHF4 too low ===


{|
{|
Line 2,739: Line 2,642:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65750 65750]
|[http://mantis.innovaphone.com/view.php?id=70226 70226]
|}
|}
MWI does not work in various Node/Pbx combination<!---->
This is a side effect of the echo canceller setting, volume in IP222/232 increased. IP241 uses old settings.<!---->
      
      
=== Appended called number to CF at gateway object needs to be adjusted acording to destination node ===
=== SIP: Cannot use SIP phone with myPBX ===


{|
{|
Line 2,750: Line 2,653:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65754 65754]
|[http://mantis.innovaphone.com/view.php?id=70252 70252]
|}
|}
Reroute from master to slave in an e164 scenario for IP down did not work<!---->
Blind transfer (sending REFER) did not work since v9hotfix3.<!---->
''Status:''
pbx.cpp<br/>pbx.h<br/>pbx_api.h
      
      
=== Trap: When Dectmaster registers user at PBX using SIP protocol ===
=== IP241: Use menu-key to enter tftp mode ===


{|
{|
Line 2,763: Line 2,664:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65798 65798]
|[http://mantis.innovaphone.com/view.php?id=70263 70263]
|}
|}
Occurred on IPBL[4.1.22]<!---->
tftp was activated with alt-key, on on IP240 menu-key is used<!---->
''Status:''
platform_orchid.c<br/>boot241.y
      
      
=== PBX: Routing did not work correct for calls between Nodes/PBXs for some Facility Calls ===
=== IP222/IP232: Sporadic Problem with the LCD Display after soft-reset ===


{|
{|
Line 2,774: Line 2,677:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65825 65825]
|[http://mantis.innovaphone.com/view.php?id=70269 70269]
|}
|}
Some Facility calls are routed without checking for call-forwards (e.g. MWI interrogation). When these calls where routed to a different PBX, routing to an object not within root node did not work, if caller was not in root node<!---->
It seem to be problematic to reset all orchid modules, e.g. the DMA module during software reset.<br/>Now only USB and ENET modules are reset, the display gets also reset.<br/>The display reset is released in the firmware.<!---->
''Status:''
''Status:''
pbx.cpp
start_orchid.S<br/>platform_orchid.c<br/>phone_orchid.cpp<br/>boot222.y<br/>boot232.y<br/>boot241.y
      
      
=== PBX User Interface: Text refers to pages PBX/General or PBX/Password which do not exist anymore ===
=== IP241: Missing fkey icons ===


{|
{|
Line 2,787: Line 2,690:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65834 65834]
|[http://mantis.innovaphone.com/view.php?id=70353 70353]
|}
|}
Should be pages PBX/Config/General and PBX/Config/Security<!---->
Icons types list, face, mask-white and mask-black are available now.<!---->
''Status:''
pbx_global.xsl<br/>pbx_objs_left.xsl<br/>pbx_regs.xsl<br/>pbx_soap.xsl<br/>pbx_import.xsl<br/>pbx_export.xsl<br/>pbx_dyn.xsl<br/>pbx_calls_view.xsl<br/>pbx_calls_classic.xsl
      
      
=== SIP: Fix for SDP answer to SDP offer with "a:inactive" ===
=== PBX: Conference trap ===


{|
{|
Line 2,800: Line 2,701:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65863 65863]
|[http://mantis.innovaphone.com/view.php?id=70362 70362]
|}
|}
Interop with CUCM.<br/>Should return RTP/AVP(inactive) if offer was RTP/AVP(inactive).<br/>Not not RTP/SAVP(inactive).<!---->
Some reworks of the PBX conference object. Fixes traps with call transfers of conference calls and conference calls to other PBX objects or mobility. Object update is also possible without call and chat clearing, now. Set maximum call number takes effect for maximum incoming calls, now.<!---->
      
      
=== SIP: Allow message bodies larger than 20kB ===
=== IP241: Status icon for conditional call diversion ===


{|
{|
Line 2,811: Line 2,712:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65880 65880]
|[http://mantis.innovaphone.com/view.php?id=70363 70363]
|}
|}
Problems handling Event Notification Extension for Resource Lists.<br/>Bodies can grow very large.<!---->
Now having 2 different status icons for diversion.<br/>One for unconditional and one for conditional forwarding.<!---->
      
      
=== ip6010 - ethernet statistics counter values missing ===
=== IP241: Moving focus on screen "Call Diversion" activates diversion ===


{|
{|
Line 2,822: Line 2,723:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65905 65905]
|[http://mantis.innovaphone.com/view.php?id=70377 70377]
|}
|}
only the counter names but no values were displayed on the ethernet statistics page  <!---->
Using UP/DOWN/LEFT/RIGHT keys on the "Call Diversion" screen makes ON/OFF changing it state.<!---->
      
      
=== Message Waiting Interrogation: Result message coding wrong ===
=== Webmedia: Recording of G.722 did not work ===


{|
{|
Line 2,833: Line 2,734:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65912 65912]
|[http://mantis.innovaphone.com/view.php?id=70382 70382]
|}
|}
a malformed message was displayed in wireshark<!---->
Empty file was created.<!---->
''Status:''
h450.cpp<br/>h450asn1.h
      
      
=== SIP: Set CLIR if display string of From-URI contains "Anonymous" ===
=== myPBX: Closing a chat window sometimes causes a Java Script error ===


{|
{|
Line 2,846: Line 2,745:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65925 65925]
|[http://mantis.innovaphone.com/view.php?id=70439 70439]
|}
|}
Not only if userpart of From-URI contains "anonymous".<!---->
client.js, line 2033<br/><br/>The Problem is a collision of closing the window and incoming messages for that window.<!---->
      
      
=== ip6010 - same MAC address was assigned to ETH0 and ETH1 ===
=== SIP: DNS resolving _stun._udp.xten.com did not work ===


{|
{|
Line 2,857: Line 2,756:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65939 65939]
|[http://mantis.innovaphone.com/view.php?id=70449 70449]
|}
|}
this results in problems when both interfaces are connected to the same LAN segment <!---->
Because of unusual DNS response.<br/>Target attribute of SRV answer records did not contain host domain name, but ip address.<br/><br/><result val="0" title="SUCCESS"><br/><br/><answer_rrs title="Answer Records"><br/><rr rr-type="33" rr-name="SRV" name="_stun._udp.xten.com" ttl="3407" priority="10" weight="0" port="3478" target="216.93.246.16"/><br/><rr rr-type="33" rr-name="SRV" name="_stun._udp.xten.com" ttl="3407" priority="10" weight="0" port="3478" target="216.93.246.14"/><br/></answer_rrs><br/><br/><ns_rrs title="Authoritative Nameserver Records"><br/></ns_rrs><br/><br/><ar_rrs title="Additional Records"><br/></ar_rrs><br/><br/></result><br/><!---->
      
      
=== PBX-SOAP: Don't provide caller number if CLIR was used on call to monitored endpoint ===
=== H.323: Renegotiation to Fax did not work under some conditions ===


{|
{|
Line 2,868: Line 2,767:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65944 65944]
|[http://mantis.innovaphone.com/view.php?id=70462 70462]
|}
|}
If this was an internal call, the PBX knows the calling number anyway, but it should not be sent on SOAP<!---->
Problem happened<br/>- If switch to fax was done right after connect. This is typically done by IP Fax Servers<br/>- If multiple signaling hops (e.g. multiple PBXs) were used<br/>- If connect to a tone interface happened during dialing<!---->
''Status:''
pbx_xml.cpp
      
      
=== PBX-SOAP: UserDTMF did not send DTMF to Voicemail or Waiting Objects ===
=== myPBX: Norwegian translation updated ===


{|
{|
Line 2,881: Line 2,778:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65958 65958]
|[http://mantis.innovaphone.com/view.php?id=70477 70477]
|}
|}
It only sent DTMFs to a VOIP connection<!---->
Sent => Send<br/>Logginn => Pålogging <br/>Permanent logginn => Forbli pålogget<!---->
''Status:''
pbx_xml.cpp
      
      
=== Gateway SIP Interfaces: Could not configure internal registration for a disabled interface ===
=== ethernet broadcasts bearing unicast IP packets with an arbitrary destination address were passed to the local IP-stack ===


{|
{|
Line 2,894: Line 2,789:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65975 65975]
|[http://mantis.innovaphone.com/view.php?id=70503 70503]
|}
|}
and if a interface was disabled afterwards, the config for the internal registration was lost<!---->
some load-balancing implementations send unicast IP packets (specially TCP-SYN) as ethernet broadcast packets.<br/>Such packets must be silently discarded if the IP destination address is not the address of one of the local interfaces.<br/><!---->
''Status:''
gk.cpp
      
      
=== SIP: Trap when receicing provisional response with RSeq header ===
=== IP222: Cannot leave menu screen with ESC when IP address ctrl was active ===


{|
{|
Line 2,907: Line 2,800:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65986 65986]
|[http://mantis.innovaphone.com/view.php?id=70569 70569]
|}
|}
Trap when trying to send PRACK<!---->
ESC was taken as BS (backspace).<br/>ESC must be handled as ESC when BS is no longer possible (empty input).<!---->
      
      
=== SIP: Media negotiation problem during hold/retrieve ===
=== PBX: Presence update for PBX objects ===


{|
{|
Line 2,918: Line 2,811:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=65995 65995]
|[http://mantis.innovaphone.com/view.php?id=70577 70577]
|}
|}
Media negotiation problem during hold/retrieve on local media<!---->
Now the presence is updated in myPBX for other PBX objects if the access is changed to allowed.<!---->
      
      
=== debug.xml, Tracing Modified ===
=== PBX: Conference presence info ===


{|
{|
Line 2,929: Line 2,822:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66004 66004]
|[http://mantis.innovaphone.com/view.php?id=70581 70581]
|}
|}
A few trace options were altered<br/>-SIP<br/>-LDAP<br/><!---->
Now the broadcast conference object updates the presence info and is shown as callable in myPBX.<!---->
      
      
=== Web-UI: Removed config option "external style sheet" ===
=== myPBX launcher: Hotkey does not work with some applications ===


{|
{|
Line 2,940: Line 2,833:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66006 66006]
|[http://mantis.innovaphone.com/view.php?id=70623 70623]
|}
|}
Does not work in version 9. <!---->
Try to get the selected phone number using WM_COPY before sending CTRL-C. This should help in some of the cases where the hotkey didn't work before.<!---->
      
      
=== Phone: Changing current presence activity does not remove presence note ===
=== SIP: Do not try to map local listen port with STUN when SIP/TCP is used ===


{|
{|
Line 2,951: Line 2,844:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66017 66017]
|[http://mantis.innovaphone.com/view.php?id=70654 70654]
|}
|}
Remove presence note when changing presence activity.<!---->
Mapping of local listen port is only required fur SIP/UDP<!---->
      
      
=== Phone: Pre-load more directory entries ===
=== IP222: Use R-key as BACKSPACE on edit fields ===


{|
{|
Line 2,962: Line 2,855:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66026 66026]
|[http://mantis.innovaphone.com/view.php?id=70664 70664]
|}
|}
Until to now the number of pre-loaded directory entries was defined by number of lines on screen.<br/>More entries have been loaded on demand when scrolling up or down.<!---->
Not ESC-key.<br/>ESC-key is for leaving screen only.<!---->
      
      
=== ip6010 - frame loss on ethernet ports running in a VLAN ===
=== Trap: When Dectmaster registers user at PBX using SIP protocol ===


{|
{|
Line 2,973: Line 2,866:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66028 66028]
|[http://mantis.innovaphone.com/view.php?id=70675 70675]
|}
|}
receiving of VLAN tagged frames did not work stable, when running ping -t over a longer time a frame loss from 5 to 10 percent was reported <!---->
After closing regstration Dectmaster starts another call.<br/>Call is rejected, but signaling enity is deleted before call object.<!---->
      
      
=== IPVA, More Watchdog Info ===
=== IP-DECT: Hanging call after failed call transfer ===


{|
{|
Line 2,984: Line 2,877:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66029 66029]
|[http://mantis.innovaphone.com/view.php?id=70756 70756]
|}
|}
a bit more diagnostics info is written in case of a watchdog<!---->
The hanging call in dectradio is fixed which occurred if a user does an unattended call transfer and the call transfer fails.<!---->
      
      
=== PBX Broadcast: CFNR was executed only after No Response Timeout even if no member ===
=== SIP: Allow STUN to be used to map local media ports on SIP/TCP interfaces ===


{|
{|
Line 2,995: Line 2,888:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66032 66032]
|[http://mantis.innovaphone.com/view.php?id=70809 70809]
|}
|}
If there is no member in the broadcast group, a CFNR configured at the Broadcast object should be executet immediatelly.<br/><br/>This was a collateral damage from hotfix<br/><br/>65261: PBX Broadcast: CFB configured at broadcast was always executed if "Execute member diversions" <!---->
STUN cannot map signaling TCP port, but UDP media ports.<!---->
''Status:''
''Status:''
pbx_bc.cpp
medialib.h<br/>media.cpp<br/>h323ch.h/cpp<br/>sip.h/cpp<br/>siptrans.h/cpp<br/>
      
      
=== H.323: Media Negotiation problem, with reverse media calls thru media-relay/exclusive media interface ===
=== IPVA, Keyboard Console, Credentials With Special Chars Couldn't Be Entered ===


{|
{|
Line 3,008: Line 2,901:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66037 66037]
|[http://mantis.innovaphone.com/view.php?id=70873 70873]
|}
|}
This could result in no media with mobility calls to sip provider<!---->
The Scancode table wasn't finished<!---->
''Status:''
h323ch.cpp
      
      
=== DECT: Gateway menu missed ===
=== SIP: Reduce memory footprint of SIP stack ===


{|
{|
Line 3,021: Line 2,912:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66077 66077]
|[http://mantis.innovaphone.com/view.php?id=70886 70886]
|}
|}
Now, the Gateway menu is added again in V9 for the IP1200.<!---->
Free INVITE request buffer when receiving ACK. No need to keep any longer.<br/>Delete INVITE client transaction when cancelling.<!---->
      
      
=== Web-UI: Fixed links into wiki reference ===
=== SIP: No route processing if neither Record-Route header nor Contact header is present ===


{|
{|
Line 3,032: Line 2,923:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66079 66079]
|[http://mantis.innovaphone.com/view.php?id=70971 70971]
|}
|}
Fixed dead links into wiki reference<!---->
Misleading trace message:<br/>  sip_call::process_routing(0xA8) Unsupported transport protocol: sip:user@domain.com;user=phone<!---->
      
      
=== PBX/Access: Visibility defined for a domain, should just add to the other definitions ===
=== IP6010: Wrong timer under high load ===


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|}
|}
Other then a definition based on a name, which should overwrite everything else<!---->
-Clear IRQ in handle-interrupt after os_interrupt is too late, since IRQ´s a enabled again and e.g. the timer irq is called again if a lower level IRQ like the enet occurs.<br/>-The IRQ needs to be cleared in the serial-irq handler, in all case. After the serial-irq other interrupts are enabled.<br/><!---->
''Status:''
''Status:''
pbx.cpp
ip6010.cpp<br/>ip6010.h
      
      
=== IP3010/6010: fax problems ===
=== SIP: Media negotiation problem ===


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|}
|}
* CED is not transfered <br/>* Wrong T38 encoding in V8 <!---->
Internal re-negotiation during early media on incoming SIP call.<br/>Provide received offer to app again.<!---->
''Status:''
ac_dsp3.cpp ( AC491 doesnt want the V21/V22... relay bits set )<br/>config.h ( config.h, X missing, on V9 this parameter is not needed )
      
      
=== Kerberos: Wrong RC4 key usage type for authenticators ===
=== SIP: Offer CLEARMODE only if bearer capabilities are "Unrestricted Digital Information" ===


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|[http://mantis.innovaphone.com/view.php?id=71162 71162]
|}
|}
The AP-REQ message uses key usage type 7 if it is encapsulated in an TGS-REQ.<br/><br/>files:<br/>kerberos_client.cpp<br/>kerberos_prot.cpp<br/>kerberos_prot.h<!---->
On "Unrestricted Digital Information" only CLEARMODE is offered (no audio codecs).<br/>On other bearer capabilities no CLEARMOE is offered (only audio codecs).<!---->
      
      
=== Config upload error message changed to show all skipped lines ===
=== phone: dialog and presence subscriptions sometimes got lost after PBX restart when phone config was stored on PBX ===


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|}
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Request from Ascom<!---->
This happened specially when both "Store Phone Config" and "Discard Config on Phone" was checked in the user object because of a unsubscribe/subscribe race condition.\t<br/><!---->
''Status:''
servlet_post.cpp servlet_post.h
      
      
=== DECT: Call waiting without Feature Codes ===
=== when editing a phone config template the dialing location inherited from a predecessor template was stored in the edited templat ===


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|}
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Call waiting is not working if the feature codes are disabled and the first call is in alerting state. Now the following call is not rejected.<!---->
after a template has been edited unchanged information units inherited from predecessor templates must be removed from the edited template. this did not work for the dialing location and thus a later change in a predecessor template had no effect. <!---->
      
      
=== HTTP-Server: Configuration page was broken ===
=== SIP: Endpoints behind NAT could not register at public PBX ===


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|}
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The processing of the configuration form accidentially produced a too long command line.<br/><br/>files: http.cpp, http.h<!---->
PBX must send response to the (public) IP address where request was received from.<br/>Not to the (private) IP address in Via header or Contact-URI.<!---->
      
      
=== HTTP-Server: Restrict number of filters and paths on configuration page ===
=== ipva, update to _platform_tracing.xsl ===


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|}
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The GUI looked like the user could define more than ten entries. Also there was a trap when the user defined one CF path too much.<br/><br/>files: http.cpp, http.xsl<!---->
ip6 missing<!---->
      
      
=== PBX-CDR: Forwarded call representation improved ===
=== SIP: No media after accepting a waiting call ===


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|}
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CDR for incoming call, which shows the forwarding as well and another CDR for the outgoing call to be used for billing only<!---->
Call waiting on a phone.<br/>Going onhock while another call is waiting starts ringer.<br/>After going offhook again the waiting call is accepted, but no media in both directions.<!---->
''Status:''
pbx.cpp
      
      
=== DECT: Delete duplicate LDAP 'pbx' &lt;gw&gt; items ===
=== CX0 Wave-Encoding Not Working If Fact-Chunk Present In Header ===


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|}
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Now duplicate LDAP 'pbx' <gw> items are deleted by the DECT users module.<!---->
CX0 Wave-Encoding Not Working If Fact-Chunk Present In Header<!---->
      
      
=== SoftwarePhone: GN 8120 USB volume keys ===
=== DHCP Server Identifier could not be cleared via WEB interface ===


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|[http://mantis.innovaphone.com/view.php?id=71305 71305]
|}
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The volume up and down key mapping for the GN 8120 USB is fixed.<!---->
When the field 'Server Identifier' was cleared and OK was pressed the just cleared value reappeared but after a reset 'Server Identifier' was clear. <!---->
      
      
=== PBX-SOAP: FindUser should not show hidden objects ===
=== IP241: Show ISDN display information in full length ===


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|[http://mantis.innovaphone.com/view.php?id=71330 71330]
|}
|}
Could be confusing<!---->
Use scrolling if required<!---->
''Status:''
pbx_xml.cpp
      
      
=== PBX External UC: Update presence of users with correct External UC set only ===
=== NAT: Mapping to different internal UDP port did not work ===


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|[http://mantis.innovaphone.com/view.php?id=71339 71339]
|}
|}
If presence info was received on an External UC interface for users not configured for External UC, the presence was still updated. This was wrong<!---->
Configured destination port got lost after configuration<!---->
''Status:''
pbx_uc.cpp
      
      
=== Phone: Changing or resetting the presence activity leaves presence note as is - should delete it ===
=== phone: send config to PBX only when the config was edited on phone ===


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|}
|}
problem: phone: Changing or resetting the presence activity leaves presence note as is. Which usually does not reflect what the user wants; remove the (absence) note. <br/><br/>solution: fixed in code, remove note upon activity chnage from function key or in user-menu<br/><br/>files: phone/app/app_disp.cpp phone/app/app_gui.cpp<br/><br/>products: all telephones<br/><br/>risks: deletes the old user activity note on the phone. <br/><br/> <br/><!---->
A config from an older PBX may contain duplicate elements which are stripped by the phone. I such a stripped config is sent back to the PBX the PBX will return the old config again.<!---->
      
      
=== Phone: Function key "Park" not displaying parked calls ===
=== SIP: Send first NOTIFY(dialog) after sending 200/OK for SUBSCRIBE(dialog) ===


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|}
|}
problem: phone; Function key "Park" not displaying parked calls <br/><br/>solution: fixed in code<br/><br/>files: forms/lcd/forms_phone_impl.cpp, forms/lcd_lib/forms_color.cpp<br/><br/>products: all telephones<br/><br/>risks: none<br/><!---->
NOTIFY(dialog) was sent before 200/OK for SUBSCRIBE(dialog)<br/><br/>\tRequest: SUBSCRIBE sip:150@192.168.200.14<br/>\tStatus: 401 Unauthorized<br/>\tRequest: SUBSCRIBE sip:150@192.168.200.14<br/>\tStatus: 100 Trying<br/>\tRequest: NOTIFY sip:158@192.168.200.214:5060<br/>\tStatus: 200 OK<br/>\tStatus: 481 Subscription Does Not Exist<br/><br/><!---->
      
      
=== phone: DHCP provided configuration was not displayed under "Phone/User x/General/Options" ===
=== SIP: Interop with Nortel CS1000 SIPLine GW ===


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|}
|}
DHCP provided configuration values always override locally configured values. A locally changed value is stored and displayed in the edit fields but has no effect as long as as DHCP provides another value. To make this clear the currently active values are displayed below or behind the corresponding edit field.<br/><br/>But instead of the active values the local values were displayed.<br/>Thus when ticking for example "Enable Secure RTP", 'enabled' was displayed behind the checkmark even when Secure RTP was not enabled in the DHCP configuration.<!---->
Nortel sends 183/Progress with 'sendrecv' answer<br/>followed by UPDATE with 'inactive' offer<br/>followed by UPDATE with 'sendrecv' offer.<br/><br/>Innovaphone SIP stack remains in 'inactive' state.<!---->
      
      
=== IP6010-CF: Kingston compact flash was not recognized ===
=== H.323 Signaling Timeout event was generated for 'normal' RAS re-transmissions ===


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|}
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the card was not recognized because a register was wrongly initialized.<!---->
A single retransmission is normal under heavy load, so this is no reason for an event. Signaling Timeout events are now generated only if they cause a state change.<!---->
      
      
=== phone: Hexadecimal values instead of descriptive texts were displayed for some rare disconnect causes ===
=== H.323 re-negotiation: Don't reuse media proposals if a select was already sent ===


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|}
|}
"0x57 - unknow cause" was displayed instead of "user not a CUG member". Mainly german descriptive texts were missing.<!---->
This fixes compatibility issues with SIP, especially when SRTP is used.<!---->
      
      
=== Kerberos: Disable RC4 encryption by default ===
=== IP241: Show both parties of each call on screen "Pickup List" ===


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|}
|}
RC4 encryption is now disabled by default. It can be turned on on the General/Admin page.<br/><!---->
Currently only calling party infomation is displayed.<!---->
''Status:''
kerberos_if.h<br/>kerberos_provider.h<br/>kerberos_provider.cpp<br/>command.h<br/>command.cpp<br/>command.xsl
      
      
=== IPVA, Double-Fault Handling Should Skip Trace History ===
=== phone: when a menu item was selected by the adjacent key sometimes the function assigned to this key was also executed ===


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|}
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Otherwise a triple fault occurs in case of a corrupted trace buffer<!---->
The function assigned to the menu item (for example pickup) was executed first and then the key-function (for example dial).  <!---->
      
      
=== PBX administration: Slave password is lost when configuring myPBX ===
=== PBX: Wrong web page when submitting an object and an error happens ===


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|}
|}
When the configuration of myPBX was submitted on the PBX/Config/myPBX page the slave password was reset.<!---->
If for example a dupicate number is detected, the same web page should be displayed including the error message for the duplicate number. But not the same page was displayed but a page which could contain information not related to the object.<!---->
''Status:''
pbx.cpp<br/>pbx_client.xsl
      
      
=== SOAP, Send leg2Info.originalCalled Info ===
=== v8 to v9 upgrade problem with gateway registration names containing non-ASCII characters ===


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|}
|}
As CallInfo.No with type="leg2orig"<!---->
In general this was a problem with config line arguments seperated by ':'. This happened with the <number>:<name> argument within gateway definitions. The ':' was url-encoded and <name> interpreted as <number><!---->
''Status:''
pbx_xml.cpp
      
      
= V9 Hotfix 2 (90600.02) =
=== PBX: CFU was executed on PRESENCE_PUBLISH/SUBSCRIBE calls ===
Changes included in Version 9 hotfix2
[http://mantis.innovaphone.com/view.php?id=66593 Definition]
 
== New Features ==
 
 
   
=== X.509: Add key usage to certificate requests ===


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|}
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The Microsoft CA (standard) does not write the key usage into the certificate if it is not specified in the request.<!---->
This was unexpected behaviour. You want to see the presence status of the configured user and not the presence status of the destination to which this user has configured a call forwarding<!---->
      
      
=== configurable DHCP client  hostname to be sent in DHCP requests ===
=== SIP: Interoperability with MX-ONE ===


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|}
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By default the hostname sent in DHCP client requests is the catenation of the short device name and the last 3 bytes of the mac address, for example IP241-2a-00-c6.<br/>The MS DHCP server creates a DNS entry from this name and the assigned IP address.<br/>A configurable name may better meet existing naming conventions. <!---->
A semi-attended transfer fails if MX-ONE sends INVITE(Replaces)<br/>instead of 200/OK when connecting a call.<!---->
      
      
=== PBX-SOAP: Support for Fax Server interface ===
=== Phone: Presence-Fkey did not always show presence set by myPBX ===


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|}
|}
UserFindDestination function added<br/><br/>UUI Protocol Discriminator fixed<!---->
Problem:<br/>1) Set presence A with IP phone (fkey shows A)<br/>2) Set presence B with myPBX (fkey shows B)<br/>3) Delete presence with IP phone (fkey shows no presence)<br/><br/>Now Fkey shows presence B.<br/><!---->
''Status:''
pbx_xml.cpp<br/>pbx_xml.h
      
      
=== Relay: Forward facilities to local destinations ===
=== PBX: Tooltip on "PBX/Config/Log Calls" checkmark wrong ===


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The relay forwards all facility events if interworking is disabled and the destination is local. This is needed for some interfaces.<br/><!---->
It read "If not checked PBX calls are logged", should be "If checked PBX calls are logged"<!---->
      
      
=== PBX-SOAP: Support receiving of UUI ===
=== IP241: Added flag to hide background-image configuration from phone menu ===


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needed for Fax Server<!---->
See http://wiki.innovaphone.com/index.php?title=Reference9:Concept_Fine_grained_function_hiding#More_Information<br/>for more information<!---->
      
      
=== SoftwarePhone: Muting ringer from context menu ===
=== Media: Discard RTP packets from wrong source ===


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|}
|}
Now it is possible to mute the ringer with the context menu item 'Mute ringer' of the tray icon. This option is not saved for the next application start.<!---->
Packets arriving at RTP port must be discarded if the source if not the expected one.<br/>To be save against DOS attack and for interop with Lync.<br/>In some scenarios Lync starts sending RTP packets while having the call set to 'inactive'.<!---->
      
      
=== Allow configuration of SRTP crypto suite, to be used for media proposals ===
=== PBX: Blind transfer with consultation to BC-Conference failed ===


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This is needed, because some equipment (e.g. Microsoft Lync) is not able to adapt to whatever proposal it receives<!---->
The call was disconnected<!---->
      
      
=== DHCP-client  monitors ethernet link down/up events and revalidates current lease after link up ===
=== H.323: No Alarm/Event should be generated by shutting down registration due to reset ===


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|}
|}
This prevents problems when a device is hot plugged to another network.<br/>Further this helps to overcvome a problem with certain cable modems.    <!---->
no event is sent to the application about this kind of unregistration<!---->
      
      
=== Fax: Progress informations ===
=== phone-orchid: micro-speaker loop of embedded test did not work ===


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|}
|}
The FAX interface provides progress and error informations, now.<!---->
Der Test war im Treiber nicht freigeschaltet<!---->
      
      
=== SIP: Presence: Support for presence/person/activities/note ===
=== myPBX: Send existing invitation links to users that join the chat session later ===


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|}
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Not only decode presence/person/note<!---->
Users A and B have a chat session.<br/><br/>1. A starts application sharing or audio conference<br/>-> A and B receive an invitation link<br/><br/>2. A adds a third user C to the chat and C accepts<br/>-> C should also receive the invitation link <br/><br/>This worked only for audio conference but not for application sharing.<br/><!---->
      
      
=== additional choices for 'loop current' on ip22/24 ip302 and ip28 gateways ===
=== IP-DECT: Ring back tone after transfer ===


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for special applications higher loop currents above 26mA are requested. Implementation adds 30mA, 35mA and 40mA values.<!---->
If in-band ring back tone is sent and the call is transferred to a new destination with no in-band ring back tone, a local ring back tone must be played to the DECT handset. This is fixed now.<!---->
''Status:''
si3210_drv.cpp si3241_drv.cpp
      
      
=== SIP: Web UI for cause code mapping ===
=== Ring Back tone missing after transfer when in-band tone was provided before but not after transfer ===


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|}
|}
Configuration of mappings between ISDN cause codes and SIP response codes.<br/>http://x.x.x.x/SIP/mod_cmd.xml?xsl=sip.xsl<br/>http://x.x.x.x/TSIP/mod_cmd.xml?xsl=sip.xsl<br/>http://x.x.x.x/SIPS/mod_cmd.xml?xsl=sip.xsl<!---->
The initial local ring back tone must be restarted when no data is received after transfer.<!---->
      
      
== Bug Fixes ==
=== PBX: Send Name Identification with CLIR calls if "No CLIR on internal Calls" ===
 
 
   
=== IP2x2x: Reset USB Hub, OHCI and LCD during softreset ===


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|}
|}
was missing on softreset.<br/>Also enable/disable USB power added<!---->
The feature "No CLIR on internal Calls" did not work completetly. The number was sent, but the Name Id was still suppressed<!---->
      
      
=== H.323 Media Re-Negotiation Problem if reverse Media call to Media-Relay/exclusive Coder Interface ===
=== phone_orchid: Ring Back tone missing  when Silence Compression is enabled ===


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|}
|}
This could happen if a call from an mobile endpoint was initiated to a SIP Trunk with media-relay/exclusive coder and re-negotiation happens, e.g. because the call was sent to a TONE interface before<!---->
When Silence Compression was enabled the tone generation was not triggered<!---->
      
      
=== PBX-SOAP: When monitoring a Gateway object, dialing information received with overlap receiving was missing ===
=== IP232: Fix for touch handling ===


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|}
|}
Applications, which monitor a gateway object, and want to connect the call depending on the number dialed, need this information<!---->
Do not open touch keyboard on controls with CTRL_READONLY.<br/>Do not open touch keyboard on controls without CTRL_ACTIVATE.<br/>Using KEY_SHIFT has modified key to uppercase permanently.<br/>Cursor positioning on text controls did not work.<br/>Multi-line editor control was not displayed after hiding touch keyboard.<br/>Hide overlay keyboard after next touched key.<br/>Move and resize editor control when activating touch keyboard.<!---->
      
      
=== PBX CF Filter for external calls did not work as expected in case of chained CFs ===
=== SIP: Trap on timer expiration during call release ===


{|
{|
Line 3,473: Line 3,332:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66599 66599]
|[http://mantis.innovaphone.com/view.php?id=71699 71699]
|}
|}
A filter for external calls did not match if the external call was forwarded already by an internal user<!---->
Media negotiation watchdog timer expired after final SIG_REL went to app.<br/>But before app deleted the call object.<!---->
''Status:''
pbx.cpp
      
      
=== SIP: SIP/TLS registration was not accepted ===
=== phone: display info provided by SETUP or CONNECT was ignored ===


{|
{|
Line 3,486: Line 3,343:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66625 66625]
|[http://mantis.innovaphone.com/view.php?id=71727 71727]
|}
|}
Inbound SIP/TLS connection was not accepted since client certificate was checked.<br/>Usually client certificates are self signed and not trusted by innovaphone PBX.<br/>Mutual TLS was done.<!---->
only the display info provided by an INFO event was handled  <!---->
''Status:''
siptrans.cpp
      
      
=== PBX-CDR: Generate CDRs for Broadcast objects ===
=== phone_orchid: continuous dialing tone stops after 10 seconds and is not restarted anymore ===


{|
{|
Line 3,499: Line 3,354:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66628 66628]
|[http://mantis.innovaphone.com/view.php?id=71753 71753]
|}
|}
No CDR was generated from Broadcast object.<!---->
the continuous dialing tone as used in most countries must be played as long as no digit is entered.<!---->
      
      
=== Gateway: Trap in case of collision of hold and clearing from remote ===
=== IP230/IP240: Sporadic traps during manufacturing programming ===


{|
{|
Line 3,510: Line 3,365:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66642 66642]
|[http://mantis.innovaphone.com/view.php?id=71778 71778]
|}
|}
This could happen on gateways with analog interfaces if the R-Key was pressed right when the other side hung up<!---->
Inefficient loop during mac address programming<!---->
''Status:''
flash_firmware.cpp<br/>
      
      
=== H.323 potential trap if AlertingNumber is received ===
=== IPVA, Reset didn't work on VMware Player 4.0 ===


{|
{|
Line 3,521: Line 3,378:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66710 66710]
|[http://mantis.innovaphone.com/view.php?id=71818 71818]
|}
|}
is no problem with existing equipment, because we don't know of any sending an AkertingNumber. Could become an problem if we do this sometimes in the future<!---->
IPVA, Reset didn't work on VMware Player 4.0<!---->
      
      
=== phone: local/default coder config was overriden by unsupported or wrong  DHCP config ===
=== IP241: Slow screen update when changing fkey type on fkey configuration screen ===


{|
{|
Line 3,532: Line 3,389:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66716 66716]
|[http://mantis.innovaphone.com/view.php?id=71831 71831]
|}
|}
DHCP provided coder config was not checked before overriding the locally defined configuration or the default values.<br/>Now a ';' separated list of coder configurations can be specified as well for the general as for the local coder, for example<br/>  G729A,60,x;G723-53,60,x/G722,30;G711,30<br/>If no coder in the list is supported by the device the respective default coder and frame size is used together with the SRTP-flag(x) and the DTMF-flag(n) from the last entry in list.<!---->
Removing and adding config controls is very slow.<!---->
      
      
=== PBX-CDR: In CDRs generate for a Gateway object with prefix, the prefix was removed from the number ===
=== ip6010 - calling &amp;amp; dtmf tone timing did not  work ===


{|
{|
Line 3,543: Line 3,400:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66719 66719]
|[http://mantis.innovaphone.com/view.php?id=71966 71966]
|}
|}
For a call to a Gateway object with Prefix checkmark set, the local number was provided with out the prefix (the number of the Gateway object itself). This was different from the behaviour of a Trunk object and the number could not be used without the knowledge of the object.<!---->
the tone time was calculated much too short<!---->
      
      
=== SIP: DNS resolving failed on no-reg interfaces if DHCP takes longer than 10sec from boot ===
=== IP232: Redesign of touch keyboard ===


{|
{|
Line 3,554: Line 3,411:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66720 66720]
|[http://mantis.innovaphone.com/view.php?id=71968 71968]
|}
|}
DNS resolving does not work as long as local IP config is not complete.<br/>DNS resolving is part of initialization process of SIP interfaces.<br/>SIP interfaces give up on DNS after 10 secs of retrying.<br/>Local IP config may take longer if DHCP is used.<!---->
Redesign of touch keyboard<!---->
      
      
=== Release not forwarded in quick dial object ===
=== IPVA, Trap After Failed FW Upload ===


{|
{|
Line 3,565: Line 3,422:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66728 66728]
|[http://mantis.innovaphone.com/view.php?id=71978 71978]
|}
|}
If the called party released the call, the remote party didn't get the release.<!---->
If a firmware upload failed with "wrong checksum" the box trapped occasionally, leaving the boot disk in an inconsistent state.<!---->
   
= V9 Hotfix 6 (90600.06) =
Changes included in Version 9 hotfix6
[http://mantis.innovaphone.com/view.php?id=71745 Definition]
 
== New Features ==
 
 
      
      
=== PBX-SOAP: Don't show dummy call, which is generated by UserCall ===
=== IP232: Presence fkey shows current activity, but not current note ===


{|
{|
Line 3,576: Line 3,441:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66748 66748]
|[http://mantis.innovaphone.com/view.php?id=71611 71611]
|}
|}
The UserCall (make outgoing call) function initiates a call to the user phone first and then transfers this call to the outgoing call. This initial dummy call should not show up as call, when monitoring the user from a different application<!---->
Show current note in second line of fkey label.<!---->
      
      
=== SIP: Lync interoperability ===
=== myPBX: Cause codes for instant messaging ===


{|
{|
Line 3,587: Line 3,452:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=66750 66750]
|[http://mantis.innovaphone.com/view.php?id=71878 71878]
|}
|}
Lync uses same Rseq number on several provisional responses.<br/>Must send PRACK for each of it.<!---->
Display causes why a chat session has been terminated.<br/><br/>For example: Unassigned number, rejected, no response...<!---->
      
      
=== phone orchid: DTMF digits typed in handsfree mode sometimes duplicated when speaker volume is high ===
=== myPBX: Make call state icons clickable ===


{|
{|
Line 3,598: Line 3,463:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67092 67092]
|[http://mantis.innovaphone.com/view.php?id=71886 71886]
|}
|}
an audible feedback for the typed digit is sent to the speaker. when the speaker volume is high this feedback is received by the microphone and detected as if it was generated for example by a DTMF tone generator.<br/>Now this detection is disabled while typing DTMF digits. <!---->
Clicking a call state icon should have the same effect as clicking the text.<!---->
      
      
=== phone orchid: Audible Signal after alerting for partner/pickup key did not work ===
=== myPBX: Notification on disconnected chat sessions ===


{|
{|
Line 3,609: Line 3,474:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67094 67094]
|[http://mantis.innovaphone.com/view.php?id=71918 71918]
|}
|}
audible signal was triggered but not played<!---->
Display a text that the chat has been terminated.<!---->
      
      
=== myPBX launcher: Accept language code from configured URLs ===
=== Debug information on assertion ===


{|
{|
Line 3,620: Line 3,485:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67137 67137]
|[http://mantis.innovaphone.com/view.php?id=71961 71961]
|}
|}
Use the language code from the URL (lang=it), if configured. Otherwise use the language of the current windows login, like in prior versions.<br/><br/><!---->
More debug information on default event handler.<!---->
      
      
=== myPBX: Workaround for redrawing problems in IE9 ===
=== myPBX: Display reason why a browser is not supported ===


{|
{|
Line 3,631: Line 3,496:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67138 67138]
|[http://mantis.innovaphone.com/view.php?id=71972 71972]
|}
|}
IE9 does not redraw the presence note when the activity image stays the same.<br/><br/>This can be fixed by avoiding to replace an image or text by the same image or text.<!---->
Currently it only says "Browser version not supported".<br/><br/>It should also tell what feature isn't supported:<br/>- XmlHttpRequest<br/>- WebStorage<br/>- PostMessage<br/><!---->
      
      
=== IP241: embedded test: Function key LED test didnt work ===
=== myPBX: Forward invitation links to incoming chat calls that are added to a chat conference ===


{|
{|
Line 3,642: Line 3,507:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67142 67142]
|[http://mantis.innovaphone.com/view.php?id=72280 72280]
|}
|}
init was missing<!---->
The same way like the invitation links (for application sharing and audio conferences) are forwarded to additional outgoing chat calls.<!---->
''Status:''
forms_color.cpp
      
      
=== IP110 enters boot loop after power cycle with v9hf1 ===
=== myPBX: Display invitation links in the chat window of the organizer ===


{|
{|
Line 3,655: Line 3,518:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67158 67158]
|[http://mantis.innovaphone.com/view.php?id=72282 72282]
|}
|}
... other phone_incas like 200a 230 240 are not affected.<br/><br/>To recover from this gwload is needed.<!---->
The organizer of an application sharing session or a audio conference wants to see all corresponding links in the chat window.<!---->
''Status:''
phone_inca.cpp
      
      
=== PBX CF Filter worked different if previous divertion with CFU or CFNR ===
=== SoftwarePhone: Password encryption tool ===


{|
{|
Line 3,668: Line 3,529:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67212 67212]
|[http://mantis.innovaphone.com/view.php?id=72431 72431]
|}
|}
This was confusing<!---->
A password encryption tool was added.<!---->
      
      
=== PBX: Syslog for registration attempt to unknown user wrong ===
=== SoftwarePhone: Crash dump ===


{|
{|
Line 3,679: Line 3,540:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67224 67224]
|[http://mantis.innovaphone.com/view.php?id=72433 72433]
|}
|}
The reason for the reject was provided as "Missing Authentication". It is now changed to "User unknown".<!---->
Now, the SoftwarePhone writes a crash dump file.<!---->
      
      
=== IP241: Disable LINK LEDs not implemented ===
=== myPBX: Display shortened links in chat messages ===


{|
{|
Line 3,690: Line 3,551:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67225 67225]
|[http://mantis.innovaphone.com/view.php?id=72445 72445]
|}
|}
on IP240 this function is available<!---->
Shorten the displayed part of the URL in order to avoid text overflow.<br/>Cut overflowing texts at the right boundary of the chat box.<!---->
''Status:''
orchid_drv.cpp .h .xsl
      
      
=== possible noise in PRI connections with ip6010 ip3010 ip1060 ===
=== PBX: New presence access flag 'online' ===


{|
{|
Line 3,703: Line 3,562:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67302 67302]
|[http://mantis.innovaphone.com/view.php?id=72538 72538]
|}
|}
some few gateways may produce noise when using the PRI ports. This can be fixed with a new CPLD code contained in future firmware.<!---->
With this flag a user allows access to his online status<!---->
''Status:''
cpld.h
      
      
=== ip22/24/28/302/305: Reduce memory footprint ===
=== myPBX: Configure visibility of online state and presence separately ===


{|
{|
Line 3,716: Line 3,573:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67304 67304]
|[http://mantis.innovaphone.com/view.php?id=72632 72632]
|}
|}
The new DSP code increases the needed memory, on boxes PBX (IP302/IP305) <br/>the memory may be already low. So we split the binary with the extra IP28 DSP code to save some bytes.<!---->
Phone status and IM status are now called online status.<br/>Activity and note are now called presence.<br/><br/>The visibility of both is now configured separately.<!---->
''Status:''
ac_494.cpp<br/>ip24.mak<br/>ac_dsp3.mak<br/>
      
      
=== Web-UI: Vertical alignment of ui elements ===
=== PBX Mobility: Better support for Opticaller ===


{|
{|
Line 3,729: Line 3,584:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67313 67313]
|[http://mantis.innovaphone.com/view.php?id=72701 72701]
|}
|}
Diagnostics/Counters and others<!---->
HTTP request to initiate call for mobile phone<!---->
''Status:''
work.css<br/>log_counters.xsl<br/>reg_edit_fkey.xsl
      
      
=== IPxx10 traps when clicking on Linux stop ===
=== myPBX: New translations ===


{|
{|
Line 3,742: Line 3,595:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67314 67314]
|[http://mantis.innovaphone.com/view.php?id=72768 72768]
|}
|}
wenn Linux gestopped wird, trapt das Gerät<!---->
For the other v9hotfix6 items regarding myPBX.<!---->
      
      
=== RTCP: Interval reduced to 5 secs (from 10 secs) ===
=== Support for more automated Fax Tests ===


{|
{|
Line 3,753: Line 3,606:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67322 67322]
|[http://mantis.innovaphone.com/view.php?id=72826 72826]
|}
|}
For interop with Lync.<!---->
Switching from Audio to Fax to Audio to Fax. Special case which could happen if switch to Fax happened before connect.<!---->
      
      
=== SIP: Trap when receiving unsolicited NOTFIY with dialog-info ===
=== myPBX: Highlight incoming alerting calls ===


{|
{|
Line 3,764: Line 3,617:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67368 67368]
|[http://mantis.innovaphone.com/view.php?id=73093 73093]
|}
|}
Trap when not subscribed for dialog-info<!---->
Highlight the alerting icon using an orange border.<!---->
      
      
=== SIP: Handling if transport errors on connection oriented transport types ===
=== SIP: Interworking with KIRK Wireless Server 300 ===


{|
{|
Line 3,775: Line 3,628:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67398 67398]
|[http://mantis.innovaphone.com/view.php?id=73110 73110]
|}
|}
Handling of transport error was added for SIP/TCP and SIP/TLS.<br/>Transaction timeout (32secs) is no longer required to elaps.<!---->
If "KIRK Wireless Server 300 PCS10__ r3327"<br/>calls into PBX and is connected with Voicemail,<br/>Voicemail may send re-INVITE with SRTP key.<br/>Instead of accepting or ignoring the SRTP option,<br/>KIRK Wireless Server 300 rejects the whole SDP offer.<br/><br/>Now we retry the re-INVITE w/o offering SRTP key.<!---->
      
      
=== SIP: NOTIFY sent after 302 moved temporarily ===
=== PBX: Forward original received ISDN display element to picking up or forwarded call ===


{|
{|
Line 3,786: Line 3,639:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67450 67450]
|[http://mantis.innovaphone.com/view.php?id=73278 73278]
|}
|}
After processing "302 moved temporarily" on an outbound call a NOTIFY (sipfrag) was sent.<!---->
In the display element from ISDN there could be vital information from equipment like crypto gateways. This should be available also if the call was picked or forwarded.<!---->
      
      
=== Compact Flash: possible trap when deleting directories ===
=== SoftwarePhone: Support for Jabra SPEAK 410 USB with product id 0x0410 ===


{|
{|
Line 3,797: Line 3,650:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67456 67456]
|[http://mantis.innovaphone.com/view.php?id=73424 73424]
|}
|}
There has been a chance for a trap when deleting directories.<br/><!---->
Now, the SoftwarePhone supports Jabra SPEAK 410 USB with product id 0x0410.<!---->
      
      
= V9 Hotfix 3 (90600.03) =
=== SIP: Overriding cause code mappings did not work ===
Changes included in Version 9 hotfix3
[http://mantis.innovaphone.com/view.php?id=67520 Definition]
 
== New Features ==
 
 
   
=== SIP: Send OPTIONS at configurable interval ===


{|
{|
Line 3,816: Line 3,661:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67519 67519]
|[http://mantis.innovaphone.com/view.php?id=73477 73477]
|}
|}
For keep alive pruposes on interfaces without registration.<br/>Required for Lync interoperability.<br/>(config change TSIP /options-interval 30)<!---->
Mapping of SIP response codes into Q.931 cause codes could not be overwritten.<!---->
''Status:''
sip.cpp/h<br/>siptrans.cpp/h
      
      
=== SIP: Fast re-routing on gateway interface w/o registration ===
=== Linux: Empty file check ===


{|
{|
Line 3,829: Line 3,672:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67593 67593]
|[http://mantis.innovaphone.com/view.php?id=73554 73554]
|}
|}
OPTIONS is used to check availability of remote peer.<br/>If OPTIONS fails, not trying to send INVITE.<!---->
Now, an empty file is not accepted as a successful file upload.<!---->
   
== Bug Fixes ==
 
 
      
      
=== SIP: New config file option /no-ms-acceptedby ===
=== myPBX: Make more obvious that closing the chat window terminates the chat ===


{|
{|
Line 3,840: Line 3,687:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67665 67665]
|[http://mantis.innovaphone.com/view.php?id=71893 71893]
|}
|}
Some IP phones get confused by this Microsoft extension.<br/>Cisco SPA 303 and Cisco SPA 962 don't stop ringing anymore<!---->
Display warning message when the user tries to close the chat window.<br/>Show disconnect icon instead of close icon in the window.<!---->
      
      
=== ip200a/230/240:  handset conversations can be monitored in a directly connected  headset ===
=== IPVA, Optimize Flash-Related I/O Operations ===


{|
{|
Line 3,851: Line 3,698:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67666 67666]
|[http://mantis.innovaphone.com/view.php?id=71954 71954]
|}
|}
This feature is required for a special application and is supported only for ip200a/230/240 phones with a directly connected headset (non DHSG).<br/>It is enabled via<br/>  config add INCA_DSP /handset-spy <volume><br/>whith <volume> in the range from 1..8  <!---->
Specifically to reduce boot-time. Utilizing 64KB I/O accesses where possible.<!---->
      
      
=== PBX: Translation of Cause "Call Rejected" to Cause "User Busy" for endpoint objects only ===
=== H.323 RAS: Registration with authentication to account without authentication failed in a strange way ===


{|
{|
Line 3,862: Line 3,709:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67668 67668]
|[http://mantis.innovaphone.com/view.php?id=71958 71958]
|}
|}
If the cause is received from a gateway, it is forwarded transparently now<!---->
Information was missing that no authentication was supported, so it was continued to retry instead of giving up right away<!---->
      
      
=== myPBX: Show active sessions on the admin interface ===
=== TCP: Roundtrip measurement wrong in case of packet loss ===


{|
{|
Line 3,873: Line 3,720:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67708 67708]
|[http://mantis.innovaphone.com/view.php?id=71985 71985]
|}
|}
The active myPBX sessions are shown on the page PBX/myPBX.<!---->
In case of packet loss, way to high round trip values were measured. If the packet-loss was to high, this could result in a constantly increasing re-transmission timeout value.<!---->
      
      
=== Phone: Added new function keys to Phone-UI (Toggle &amp;amp; Prepare Override) ===
=== SIP: Workaround for delays in media negotiation ===


{|
{|
Line 3,884: Line 3,731:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67729 67729]
|[http://mantis.innovaphone.com/view.php?id=72189 72189]
|}
|}
Description: Phone: Added new function keys to Phone-UI (Toggle &amp; Prepare Override)<br/><br/><!---->
Workaround for delays in media negotiation caused by delayed Admission on H.323 calls.<!---->
      
      
=== PBX: New presence activity "do not disturb" ===
=== SIP: Trap on IP-DECT when re-configuring PBX link ===


{|
{|
Line 3,895: Line 3,742:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67777 67777]
|[http://mantis.innovaphone.com/view.php?id=72190 72190]
|}
|}
If a users sets this presence activity no calls are passed.<!---->
85:2195:425:7 - REG_PRI.4 default(8102be48): serial_timeout<br/>85:2195:425:7 - Assertion failed line 748 in common/os/os.cpp, object deleted<br/><!---->
''Status:''
Merged to 09-80500
      
      
=== PBX: Allow CFB on Gateway Type Objects ===
=== IP232: Input of capital Latin Extended-A ===


{|
{|
Line 3,906: Line 3,755:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67827 67827]
|[http://mantis.innovaphone.com/view.php?id=72226 72226]
|}
|}
A CFB is triggered by a User Busy. If a CFB is used for example at a Trunk, the CFB is executed when the called remote user returns busy. Because this may be unexpected the CFB was not executed at a Gateway Type Object.<br/><br/>It is now enabled again, because it is useful when connecting external systems which return busy to indicate an out of channels situation<!---->
Input of capital Latin Extended-A<!---->
      
      
=== PBX Trunk/Gateway: Round robin within registrations to same device, different devices sequentially ===
=== myPBX: Queue chat messages for im calls that are not yet alerting  ===


{|
{|
Line 3,917: Line 3,766:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67835 67835]
|[http://mantis.innovaphone.com/view.php?id=72237 72237]
|}
|}
This way both round-robin or sequential usage of gateways can be configured<!---->
The following fix did not work across PBXes:<br/>#71608: myPBX: Send existing invitation links to users that join the chat session later <br/><br/>The problem was that messages were only queued in alerting state. When the call is made across PBXes this state isn't reached, yet.<!---->
      
      
=== improved test for Timeslot Switch Chip of ip6010 ip3010 ip0010 ip1060 and ip6000 ===
=== Scheduling improved to avoid processes not being scheduled during long flashman operations ===


{|
{|
Line 3,928: Line 3,777:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67840 67840]
|[http://mantis.innovaphone.com/view.php?id=72243 72243]
|}
|}
intension is better analysis of hardware problems<!---->
In version 7 it could happen, that IP and other processes were not scheduled any more during periods of long flashman operations (e.g. bootcode update or reorganizing flash).<br/><br/>In version 8 and higher there was already a fix for this problem, but this included special handling of the flashman priority level, which was not a good solution even if it worked.<!---->
''Status:''
idt72_drv.cpp, idt72_drv.h
      
      
=== ip22/24/28/302/305/6010/3010/1060/241/222/232: False DTMF detects ===
=== IP232: Touch keyboard key highlighting ===


{|
{|
Line 3,941: Line 3,788:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67844 67844]
|[http://mantis.innovaphone.com/view.php?id=72268 72268]
|}
|}
Seen on IP222 and IP6010<!---->
Added highlighting for key like SHIFT, BACKSPACE, etc.<!---->
''Status:''
ac_dsp3.cpp
      
      
=== phone_orchid: builtin color display test ===
=== PBX: H.323 Names in some places (e.g. Waiting Queue Maps) containing non-ASCII charcters could break the user interface ===


{|
{|
Line 3,954: Line 3,799:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67845 67845]
|[http://mantis.innovaphone.com/view.php?id=72277 72277]
|}
|}
After the builtin test function has been started the display test mode is entered when the 'Esc' key is pressed. Numeric keys trigger a full screen test display, all other keys stop the display test mode.<br/>To the keys 0..9 the following patterns are assigned: <br/>DarkGray, White, Grey, Black, Red, Green, Blue, Yellow, Cyan, Magenta<!---->
This happened esspecially for Names as destinations for Waiting Queue maps, but it could happen at other places as well.<!---->
      
      
=== IP-DECT: Configuration XML data for OEM device ===
=== SIP: Cleanup failed (resources leaking) ===


{|
{|
Line 3,965: Line 3,810:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67858 67858]
|[http://mantis.innovaphone.com/view.php?id=72284 72284]
|}
|}
A new configuration XML attribute is added for changing a OEM GUI.<!---->
Call and channel objects were not freed sometimes<br/>when INVITE was followed by CANCEL very fast.<!---->
      
      
=== Phone: "Function keys not modifiable on the phone" mask should disable creation of new function keys of masked type ===
=== IP241: Disable Link LED not working ===


{|
{|
Line 3,976: Line 3,821:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67905 67905]
|[http://mantis.innovaphone.com/view.php?id=72338 72338]
|}
|}
Description: Phone: "Function keys not modifiable on the phone" mask should disable creation of new function keys of masked type. Currently, only modification of preset function keys is disabled, but the creation of new ones enabled and possible.<!---->
Timing problem with access to the paged MDIO registers<!---->
''Status:''
orchid_drv.cpp
      
      
=== Phone: Added command line option to hide Administration Menu and/or MAC/Serial completely ===
=== Trap: When accessing web interface ===


{|
{|
Line 3,987: Line 3,834:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67943 67943]
|[http://mantis.innovaphone.com/view.php?id=72362 72362]
|}
|}
Description: Phone: Added command line option to hide Administration Menu and/or MAC/Serial completely. See /hide-mask option to PHONE ADMIN-UI in wiki for more information.<br/><br/><!---->
Seen once.<br/>Not reproducable yet.<!---->
      
      
=== IP-DECT: OEM registration string ===
=== Ip6010 DSP Disconnect timeout after fax session ===


{|
{|
Line 3,998: Line 3,845:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68058 68058]
|[http://mantis.innovaphone.com/view.php?id=72403 72403]
|}
|}
The Radio registration string of an OEM device is changed to check a OEM license in the DECT Master.<!---->
Debugs added<br/>* enable with http://addr/debug.xml  DSP trace and DSP control message trace to printout  all packets to the DSP with a descriptive string. That allows to analyse the message flow to the DSP after a trap.<br/>* for further testing old fax disconnect procedure can be enabled with http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl  with "t38 skip fax close".<!---->
''Status:''
ac_dsp3.cpp<br/>ac_dsp3.h<br/>ac_491.h<br/>dsp.xsl
      
      
=== SIP: Diagnostics of transport error "Remote server certificate mismatch" ===
=== IP2x2: Fixed fkey handling of builtin test ===


{|
{|
Line 4,009: Line 3,858:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68142 68142]
|[http://mantis.innovaphone.com/view.php?id=72442 72442]
|}
|}
If remote sverer certificate does not match the destination domain name<br/>signaling connection is refused by client.<br/><!---->
Fkey test did not work for touchkeys, key symbols were misplaced on screen<!---->
      
      
=== myPBX: Support for WebEx meeting passwords ===
=== myPBX: Improve presentation of chat conferences ===


{|
{|
Line 4,020: Line 3,869:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68172 68172]
|[http://mantis.innovaphone.com/view.php?id=72444 72444]
|}
|}
In WebEx a meeting can have a password that must be entered by the attendees when they join. Some WebEx accounts can only create meetings with passwords.<br/><br/>The possibility to configure a global meeting password is added to the PBX/Config/myPBX page.<!---->
Chat conferences should not be displayed as multiple calls but as a single item in the main window.<!---->
      
      
=== Phone: Message function key ===
=== IP232: Parameters for touch sensor fixed for latest sensor ===


{|
{|
Line 4,031: Line 3,880:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68208 68208]
|[http://mantis.innovaphone.com/view.php?id=72455 72455]
|}
|}
Description: Phone: Message function key. Multifunctional depending on number of unread messages. Stores one prepared message (with destination and message text) and presents the new message screen when invoked. If incoming messages pending, display the letter/message icon and jump to incoming-messages subscreen upon invocation.<!---->
latest touch was treated like the very old version that needed other parameters.<br/>Old touch is not supported any more ( only 3 were build)<!---->
''Status:''
edt_touch.cpp
      
      
=== phone: Finnish translations updated ===
=== IP241,IP222,IP232: Make password configuration more convenient ===


{|
{|
Line 4,042: Line 3,893:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68395 68395]
|[http://mantis.innovaphone.com/view.php?id=72456 72456]
|}
|}
no english placeholders anymore, Texts begin with upper case letter<!---->
When entering a passwort on the phone's ui content is displayed as "****".<br/>Show last enered character in plain text.<!---->
      
      
=== IP150: dimming of key LEDs and LCD backlight ===
=== Ip6010 DSP Allow coder change from T38 to voice and back to T38 ===


{|
{|
Line 4,053: Line 3,904:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68414 68414]
|[http://mantis.innovaphone.com/view.php?id=72457 72457]
|}
|}
to save power in special environments the key LEDs can be dimmed by<br/>  config add KEYS0 /light-off<br/>The lcd backlight can be configured the usual menu way on the phone. <br/><br/><!---->
..<!---->
''Status:''
ac_dsp3.cpp<br/>
      
      
=== HTTP-Client: Allow user names longer than 16 characters ===
=== IP222: Take center key of 4-way-navigation control an OK key ===


{|
{|
Line 4,064: Line 3,917:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68499 68499]
|[http://mantis.innovaphone.com/view.php?id=72477 72477]
|}
|}
Now user names with up to 64 characters are allowed.<!---->
Center key should enter selected menu item.<br/>Not leaving the current menu screen.<br/>ESC key can be used to leave current menu screen.<br/>Phone app will ask whether to save changes.<!---->
      
      
=== Phone: Call forwarding (always, busy, no reply) destination now choosable from dial-menu ===
=== myPBX: Name and Number Display not correct on IM sessions across PBXs ===


{|
{|
Line 4,075: Line 3,928:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68504 68504]
|[http://mantis.innovaphone.com/view.php?id=72491 72491]
|}
|}
Phone: Call forwarding (always, busy, no reply) destination now choosable from dial-menu. Usage: enter number or search for phonebook entry, press menu-key, scroll down to choose call-forwarding (always, busy or no-reply) and acknowledge choice in CF-screen.<br/><br/><!---->
for local sessions (same PBX) the number and the Long Name was displayed for the remote party and this should be the same for remote session.<br/><br/>Sometimes the number was missing, sometimes the Name was displayed instead of Long Name.<br/><br/>The number was not adjusted correctly if different Nodes were involved<!---->
      
      
=== IP-DECT: Allow setting empty text for idle display ===
=== SoftwarePhone: Trap and media channel after conference ===


{|
{|
Line 4,086: Line 3,939:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68553 68553]
|[http://mantis.innovaphone.com/view.php?id=72511 72511]
|}
|}
Some handsets will not show signal strength and battery symbols if idle display is defined, they must be set to empty string.<!---->
If a call is disconnected during conference mode, the SoftwarePhone traps or there is no voice channel to the remaining party. This is fixed now.<!---->
      
      
=== ac_dsp3: support echo canceller trace ===
=== HTTP: Accept authentication with both Latin1 and UTF-8 coding ===


{|
{|
Line 4,097: Line 3,950:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68649 68649]
|[http://mantis.innovaphone.com/view.php?id=72512 72512]
|}
|}
required to analyze echo canceller problems<!---->
The server does not know what encoding is used in Basic- or Digest-Authentication. Therefore both encodings shall be tried.<!---->
      
      
=== Phone: On IP240, make OK key a headset activation key (along with Space Key) ===
=== ip28 codec hang when TEL port configured to '600Ohms' ===


{|
{|
Line 4,108: Line 3,961:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68774 68774]
|[http://mantis.innovaphone.com/view.php?id=72521 72521]
|}
|}
Phone: On IP240, make OK key a headset activation key (along with Space key).<br/>Enabled from Phone/Preferences : Use Newline/OK Key as Headset Key.<!---->
this option experienced problems in the past and should therefore not be selected. If selected the firmware will switch to default CTR21.<!---->
      
      
== Bug Fixes ==
=== PBX: Cause for release/reject of a IM session not signaled ===
 
 
   
=== IPxx10: error handling in sata driver ===


{|
{|
Line 4,123: Line 3,972:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67229 67229]
|[http://mantis.innovaphone.com/view.php?id=72533 72533]
|}
|}
Old cards are producing DMA errors that were not handled properly. Try again read/write operation after error recovery.<!---->
Better feedback to the User<!---->
      
      
=== DECT: IP6000/IP6010/... default config Master mode off ===
=== myPBX: New icons for unknown phone and im status ===


{|
{|
Line 4,134: Line 3,983:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67479 67479]
|[http://mantis.innovaphone.com/view.php?id=72536 72536]
|}
|}
Now the Dect Master is in mode off by default for the IP6000/IP6010/...<!---->
The difference between open, closed and unknown was not understood well.<!---->
      
      
=== SoftwarePhone: DTMF to voice mail ===
=== SoftwarePhone: Dialing outbound calls temporarily failed ===


{|
{|
Line 4,145: Line 3,994:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67563 67563]
|[http://mantis.innovaphone.com/view.php?id=72628 72628]
|}
|}
Now, sending DTMF with myPBX is correctly working. This fixes sending DTMF to voice mail.<!---->
When there was a held call and a consultation call and the consultation call was released by the remote peer the SoftwarePhone did not accept further outbound calls until the held call had been released. This is fixed now.<!---->
      
      
=== VM: Trap while processing self-forwarded call ===
=== IP232: Fix for touch keyboard handling ===


{|
{|
Line 4,156: Line 4,005:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67570 67570]
|[http://mantis.innovaphone.com/view.php?id=72651 72651]
|}
|}
VM: Trap while processing self-forwarded call<!---->
When editing call diversions, one of the on/off controls may render across touch keyboard.<br/>Content of multi-line-edit-control was mis-placed when touch-keyboard was activated.<br/>Display of first matching directory entriy on indirect dialing screen.<!---->
      
      
=== SIP: Uninitialized data in SDP offer/answer ===
=== ip241 - monitor mode (handset + speaker) did not work in V9hotfix5 ===


{|
{|
Line 4,167: Line 4,016:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67617 67617]
|[http://mantis.innovaphone.com/view.php?id=72702 72702]
|}
|}
Applies to G.726 exclusive calls only.<!---->
monitor mode (handset + speaker) was displayed but the speaker was mute<!---->
      
      
=== Phone: Main menu scrolling below last item broken ===
=== update - scfg command could hang when the HTTP session was broken or prematurely closed by the server ===


{|
{|
Line 4,178: Line 4,027:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67618 67618]
|[http://mantis.innovaphone.com/view.php?id=72708 72708]
|}
|}
Phone: Main menu scrolling below last item broken.<br/>1st item hould be activated upon down arrow press (done) and screen focus moved up (not being done - bug).<!---->
in consequence update script processing was stopped until reboot<!---->
      
      
=== cpld update not working on ip241 ===
=== Web-UI: Wider input fields for domain names ===


{|
{|
Line 4,189: Line 4,038:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67629 67629]
|[http://mantis.innovaphone.com/view.php?id=72723 72723]
|}
|}
-<!---->
When configuring<br/> - Remote Domain<br/> - Local Domain<br/> - Proxy<br/> - STUN Server<br/>on a Gateway interface there's need for more space.<!---->
''Status:''
phone_orchid.cpp phone_orchid:config.h xilinx.cpp<br/>and as consequence also ip6010:config.h
      
      
=== SIP: Interoperability with Lync and media-bypass ===
=== Trap: When Dectmaster registers user at PBX using SIP protocol ===


{|
{|
Line 4,202: Line 4,049:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67645 67645]
|[http://mantis.innovaphone.com/view.php?id=72729 72729]
|}
|}
Ack contained wrong To-Tag when calling a lync client in media-bypass scenario.<br/>Results into call drop after 30 seconds.<!---->
When Dectmaster registers user at PBX using SIP protocol<!---->
      
      
=== Phone: Automatic key repeat did not work on IP222/IP232 ===
=== SIP: Add payload type for RTP-DTMF in case of media-relay ===


{|
{|
Line 4,213: Line 4,060:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67661 67661]
|[http://mantis.innovaphone.com/view.php?id=72732 72732]
|}
|}
Automatic key repeat did not work on phone w/o alpha keyboard<!---->
Add payload type for RTP-DTMF (telephone-event) to SDP offer in case of media-relay.<br/>Should help on Mobility scenarios.<!---->
      
      
=== Call to Voicemail did not work anymore with SRTP ===
=== PBX: Called Name displayed when calling an object with forking was wrong ===


{|
{|
Line 4,224: Line 4,071:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67672 67672]
|[http://mantis.innovaphone.com/view.php?id=72735 72735]
|}
|}
This was a collateral damage from fix<br/><br/>http://wiki.innovaphone.com/index.php?title=Support:DVL-Roadmap_Firmware_V9#Allow_configuration_of_SRTP_crypto_suite.2C_to_be_used_for_media_proposals<!---->
The name of the forking destination was displayed instead of the name of the called object<!---->
      
      
=== Phone: long function key titles hide idle screen information ===
=== IP4001/IP6000 -UART driver did not work ===


{|
{|
Line 4,235: Line 4,082:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67681 67681]
|[http://mantis.innovaphone.com/view.php?id=72743 72743]
|}
|}
Description: Phone: long function key titles hide idle screen information.<br/>Fixed: important idle screen information now shortens the amount of displayed function key name. Following information is now displayed over the function key text: a) crossed bell icon on do-not-disturb (lines 2+3) b) CFU + CFU-destination (lines 2+3) c) missed calls, unread messages and waiting callbacks (line 4)<br/><br/> <!---->
the hardware requires the registers to be written 32-bit wise which was not respected by a fix one year before.  <!---->
      
      
=== PBX Waiting: Call forwarded with DTMF mapping was shown in myPBX for each registration ===
=== PBX: No Audio if call thru Waiting Queue DTMF destination, was transfered to BC-Conf ===


{|
{|
Line 4,246: Line 4,093:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67682 67682]
|[http://mantis.innovaphone.com/view.php?id=72746 72746]
|}
|}
The call was not sent with the original conferenceID, so myPBX could not detect that it was in fact the same call<!---->
Problem caused by call state management error in PBX for calls connected without alert if alert was received later<!---->
      
      
=== PBX: License accounting in centralized licensing scenario wrong if master not available ===
=== IP241,IP222,IP232: Wrong call state displayed ===


{|
{|
Line 4,257: Line 4,104:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67698 67698]
|[http://mantis.innovaphone.com/view.php?id=72748 72748]
|}
|}
When the master is available the slave stores the licenses from the master including the usage. This stored usage included the licenses used by the slave itself, so if after a reset the master was not available the local usage just added to this.<br/><br/>Now from the stored usage the local usage is subtracted.<!---->
"Destination reached" displayed instead of "subscriber busy" if consultation call was rejected.<br/><!---->
      
      
=== phone_orchid: microphone is not mute on a call intrusion in  silent monitoring mode / microphone cannot be muted in a conference ===
=== PBX Waiting: Ringback missing when using DTMF to dial from one WQ to another which is alerting ===


{|
{|
Line 4,268: Line 4,115:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67704 67704]
|[http://mantis.innovaphone.com/view.php?id=72766 72766]
|}
|}
On a call intrusion in silent monitoring mode the microphone of the intruding party must be mute. In a conference the micro should be muted when the micro key is pressed and unmuted when the micro key is pressed again.<br/>Muting the microphone did work when only one call was active but not when two calls were active as in a intrusion/conference.<!---->
This was a collateral damage for a fix for Waiting Queue announcements from a Boolean Object<!---->
      
      
=== Phone: CLIR on text messaging did not work ===
=== IP222,IP232: Long key press on numeric block while being offhook does not open directory search ===


{|
{|
Line 4,279: Line 4,126:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67710 67710]
|[http://mantis.innovaphone.com/view.php?id=72831 72831]
|}
|}
CLIR on text messaging did not work<!---->
Long key press during offhook is expected to open directory search.<!---->
      
      
=== SIP: Interoperability with Lync: Handling of REFER from Lync ===
=== IP232: New touch control in menu header for screen exit ===


{|
{|
Line 4,290: Line 4,137:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67713 67713]
|[http://mantis.innovaphone.com/view.php?id=72833 72833]
|}
|}
REFER for blind transfer was rejected with "406 Not Acceptable",<br/>due to absence of user part in refer-to URI.<!---->
Re-design<!---->
      
      
=== PBX Trunk: Problem with Forking to trunk if multiple GWs are registered to Trunk ===
=== IP241,IP222,IP232: Support for unicode 0x308 (diaeresis/umlaut) ===


{|
{|
Line 4,301: Line 4,148:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67720 67720]
|[http://mantis.innovaphone.com/view.php?id=72859 72859]
|}
|}
If one of the gateways rejected the call (no channel, not connected, ...), the original call from which was forked was disconnected<!---->
When ä,ö,ü are encoded as a,o,u followed by diaeresis from unicode block 'Combining Diacritical Marks' only a,o,u where displayed.<br/>Now ä,ö,ü are displayed.<!---->
      
      
=== Phone: Calls received with CLIR appear in call list with an empty entry ===
=== IP241,IP222,IP232: Support for hebrew and arabic presence notes ===


{|
{|
Line 4,312: Line 4,159:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67722 67722]
|[http://mantis.innovaphone.com/view.php?id=72905 72905]
|}
|}
Description: Phone: Calls received with CLIR appear in call list with an empty entry; now fixed to display the CLIR text: anonymous/unbekannt/...<br/><br/><!---->
Hebrew and arabic text passages are rendered frmo right to left.<!---->
      
      
=== PBX: License accounting did not work with Unknown Registrations under some special conditions ===
=== PBX: Conference no media ===


{|
{|
Line 4,323: Line 4,170:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67731 67731]
|[http://mantis.innovaphone.com/view.php?id=72915 72915]
|}
|}
It could happen that a registration to a user was not accounted for if the endpoint used for this already had an unknown registration at the time the user was created<!---->
There is sometimes no media for a conference member. This is fixed, now.<!---->
      
      
=== User Interface: Alignment of tables fixed ===
=== SIP: Memory leak during transfer ===


{|
{|
Line 4,334: Line 4,181:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67734 67734]
|[http://mantis.innovaphone.com/view.php?id=73003 73003]
|}
|}
Some strange alignments were introduced with the last hotfix<!---->
Occured on internal testing only (002-conf-with-bcast.xml)<!---->
      
      
=== Gateway: Interface Name with multiple "Umlaute" did not work ===
=== IP241,IP222,IP232: Update for CFU indication on idle screen ===


{|
{|
Line 4,345: Line 4,192:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67737 67737]
|[http://mantis.innovaphone.com/view.php?id=73013 73013]
|}
|}
There was a length limitation of the URL encoded output, which was already exceeded if three "Umlaute" (or any character which is encoded in more the one byte with utf-8), were used<!---->
Not displaying "cfu:241" on header of idle screen<br/>but "Diverted to 241"<br/>or "Umgeleitet zu 241"<br/>or "Renvoyé sur 241"<br/>etc.<!---->
      
      
=== IP-DECT: Packetization could change after handover ===
=== myPBX: Simplify adding multiple users to a chat session ===


{|
{|
Line 4,356: Line 4,203:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67738 67738]
|[http://mantis.innovaphone.com/view.php?id=73016 73016]
|}
|}
On the new radio the RTP should be sent using the same packetization as was negotiated with the original call<!---->
Do not unselect the chat session, after a person has been added. Replace "start chat" buttons by "add to chat" buttons when a chat session is selected.<!---->
      
      
=== SIP: Fix for early media from Waitng Queue ===
=== RTP-DTMF: Digit may get lost during media re-negotiation ===


{|
{|
Line 4,367: Line 4,214:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67775 67775]
|[http://mantis.innovaphone.com/view.php?id=73037 73037]
|}
|}
PROGRESS after ALERT was not handled by SIP stack.<br/>Now 183 Session Progress with SDP is send after 180 Ringing w/o SDP.<!---->
Receiver starts handling on END event.<br/>Sender may stop sending before END event was sent.<!---->
      
      
=== Leak checking improved ===
=== myPBX: Do not show bubble for outgoing chat calls and messages ===


{|
{|
Line 4,378: Line 4,225:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67783 67783]
|[http://mantis.innovaphone.com/view.php?id=73039 73039]
|}
|}
sometimes leaks were falsely detected. Problem if objects are about to be deleted, which were not owned by any module anymore. This happend esspecially with httpclient.<!---->
Notifications are only needed for incoming calls and messages.<!---->
      
      
=== SIP: Generate/add SRTP key on media-relay interfaces ===
=== myPBX: Skript errors in chat window ===


{|
{|
Line 4,389: Line 4,236:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67789 67789]
|[http://mantis.innovaphone.com/view.php?id=73040 73040]
|}
|}
This enables even partial SRTP (SRTP on one side of media-relay).<!---->
Closing the chat window caused different script errors in Internet Explorers and Firefox.<!---->
      
      
=== Phone: Enable "Activate Registration" without user/password authentication if "Protect Configuration at Phone" set ===
=== LDAP replication from v7 could stop ===


{|
{|
Line 4,400: Line 4,247:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67791 67791]
|[http://mantis.innovaphone.com/view.php?id=73052 73052]
|}
|}
Description: Phone: Enable "Activate Registration" without user/password authentication if "Protect Configuration at Phone" set. Activating a registration is a state change, and not a configuration modification, so allow this option.<br/><br/> <!---->
LDAP replication from v7 could stop<!---->
      
      
=== H.323: A name_id of length 0 resulted in invalid H.450 coding ===
=== SIP: Don't write SRTP key into T.38 part of media description ===


{|
{|
Line 4,411: Line 4,258:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67796 67796]
|[http://mantis.innovaphone.com/view.php?id=73112 73112]
|}
|}
An empty name identification received was forwarded in H.323 as invalid H.450. Such a name is now forwarded as 'name not available'.<!---->
v=0<br/>o=- 1295 1 IN IP4 10.120.55.3<br/>s=-<br/>c=IN IP4 10.120.55.6<br/>t=0 0<br/>m=audio 20026 RTP/SAVP 8 101 13<br/>a=rtpmap:101 telephone-event/8000<br/>a=fmtp:101 0-15<br/>a=ptime:20<br/>a=silenceSupp:off - - - -<br/>a=inactive<br/>a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9xHxSg836505XOXwdIHfQ8Cm2ZYezNPpjvHNPCvb<br/>m=image 0 udptl t38<br/>a=T38FaxVersion:0<br/>a=T38MaxBitRate:9600<br/>a=T38FaxFillBitRemoval:0<br/>a=T38FaxTranscodingMMR:0<br/>a=T38FaxTranscodingJBIG:0<br/>a=T38FaxRateManagement:transferredTCF<br/>a=T38FaxUdpEC:t38UDPRedundancy<br/>a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9xHxSg836505XOXwdIHfQ8Cm2ZYezNPpjvHNPCvb<br/><!---->
      
      
=== H.323 Malformed packet ===
=== PBX External UC: Support of multisite setup ===


{|
{|
Line 4,422: Line 4,269:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67803 67803]
|[http://mantis.innovaphone.com/view.php?id=73115 73115]
|}
|}
The ASN.1 encoder had a bug under one special condition: For a constrained character string with a maximum length of more or equal to 16bits, with an effective length of zero, the padding for octett alignment was missing for the zero length bitfield containing the string.<br/><br/>In H.323 this only happens for the CallIdentity used for H.450 call transfer message in case of blind transfer without consultation.<br/><br/>This fix breaks compatibility with earlier versions, for this reason this fix is available for version 9,8,7 and 6.<br/><br/>If phones and PBX with versions containing and not containing this fix are mixed  the following problems will occur:<br/>- A blind transfer without consultation (initiated with the redial key) is not possible<br/>- A call which was transfered without consultation is not displayed at the transfered-to phone as transfered<br/><!---->
This requires all calls to the External UC to be routed to one interface. Before this fix the calls were routed within the local PBX only.<br/><br/>Now a Name (not Long Name as before) has to be configured for the External UC interface.<!---->
      
      
=== SIP: Unwanted media-relay sessions when using forking/broadcast/multi-reg ===
=== IP241,IP222,IP232: Show number of missed calls in status bar ===


{|
{|
Line 4,433: Line 4,280:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67819 67819]
|[http://mantis.innovaphone.com/view.php?id=73140 73140]
|}
|}
If in incoming SIP was routed to multiple destinations<br/>the final session could be media-relay although not configured.<!---->
New extended symbol is more eye-catching.<!---->
      
      
=== ip22/24/28/302/305: DSP affected by certain fax tones ===
=== PBX: Called party number wrong at original called phone with forking ===


{|
{|
Line 4,444: Line 4,291:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67821 67821]
|[http://mantis.innovaphone.com/view.php?id=73145 73145]
|}
|}
Brother fax problem, the first fax is transfered, the next fax transfers fail.<br/>Switch to fax from remote is now done without reopening the channel.<br/>Closing the channel waits until t38 is switched off.<br/><!---->
If the forking destination and the original called phone are assigned to different nodes<!---->
''Status:''
ac_dsp3.cpp ac_dsp3.h
      
      
=== IP30x, IP1060, IP2010, IP6010: Fax did not work if rerouted from ISDN interface to a Voip destination ===
=== Media Negotiation: v9 XPARENT not compatible to v8 or earlier XPARENT ===


{|
{|
Line 4,457: Line 4,302:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67823 67823]
|[http://mantis.innovaphone.com/view.php?id=73153 73153]
|}
|}
DSP was not configured correctly<!---->
In v9 a dynamic payload type is used for XPARENT to be compatible to SIP, whereas in v8 an earlier payload type 0 was used. Within the media negotiation this should be detected and switched back to payload type 0.<!---->
      
      
=== IP2x2: Handset microphone gain increased ===
=== Phones: Update of phone text data ===


{|
{|
Line 4,468: Line 4,313:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67843 67843]
|[http://mantis.innovaphone.com/view.php?id=73211 73211]
|}
|}
changed from 22.5db to 30db<!---->
Some translations missing (Polski,Eesti).<br/>Support for language typical letters (Polski).<br/>Some abbreviation removed (for color phones).<!---->
''Status:''
ac_codec3.cpp
      
      
=== IP241: Remote party information truncated more than necessary ===
=== PBX: Conference GUI update with IE ===


{|
{|
Line 4,481: Line 4,324:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67872 67872]
|[http://mantis.innovaphone.com/view.php?id=73256 73256]
|}
|}
Line 3 of call ctrl<!---->
The PBX broadcast conference object window is not closed if IE is used. This fixed now.<!---->
      
      
=== IP-DECT: Changed channel option SRTP crypto suite ===
=== SIP: support for "Content-Type: multipart/alternative" ===


{|
{|
Line 4,492: Line 4,335:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67904 67904]
|[http://mantis.innovaphone.com/view.php?id=73260 73260]
|}
|}
Now DECT system channel configuration option 'Secure RTP' is a drop down box. The DECT Master correctly transmits the changed option. This feature was changed in V9 Hotfix 2, related case #66810.<!---->
Lync interoperability<!---->
      
      
=== SIP: DNS problem when SRV response provides no additional records ===
=== IP2x2: Keyboard handling on call reroute ===


{|
{|
Line 4,503: Line 4,346:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67907 67907]
|[http://mantis.innovaphone.com/view.php?id=73268 73268]
|}
|}
If 2-step resolving is required (SRV and A) the service port<br/>of the SRV response got lost and default SI Pport 5060 was used.<!---->
Redial key launches "Transfer" screen.<br/>After entering the destination number, Redial key should cause execution of call reroute.<!---->
      
      
=== SIP: Trap when configuring STUN server on a SIP/TCP or SIP/TLS interface ===
=== PBX Waiting: When using a waiting queue for outgoing dialing, the announcement was cut off at the beginning ===


{|
{|
Line 4,514: Line 4,357:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67923 67923]
|[http://mantis.innovaphone.com/view.php?id=73274 73274]
|}
|}
STUN is for SIP/UDP only.<!---->
This was because the announcement was started already when the call was sent out and not when the call was connected<!---->
      
      
=== myPBX: Always send an ID with CT_INITIATE ===
=== myPBX: Allow spaces in last LDAP attribute ===


{|
{|
Line 4,525: Line 4,368:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67932 67932]
|[http://mantis.innovaphone.com/view.php?id=73276 73276]
|}
|}
This is needed for interoperability with phones using v9hotfix2 or older. (See #67803)<!---->
Currently "j doe innovaphone" finds all internal users starting with j. It should only find internal users that start with "j doe innovaphone".<br/><!---->
      
      
=== SIP: Must answer every request - even unknown/unsupported methods ===
=== IP232: Cursor positioning when touching ip address control ===


{|
{|
Line 4,536: Line 4,379:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67935 67935]
|[http://mantis.innovaphone.com/view.php?id=73407 73407]
|}
|}
Lync sends proprietary NEGOTIATE request and waits for response.<br/>Otherwise transport connection is blocked for any upcoming request.<!---->
Make sure input cursor is at right side of touched octet.<!---->
      
      
=== PBX Waiting: A call rejected on a primary operator was never sent to all operators ===
=== SIP: Configuration of SIP response code mappings did not work on IPxx10 and IPVA ===


{|
{|
Line 4,547: Line 4,390:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67945 67945]
|[http://mantis.innovaphone.com/view.php?id=73453 73453]
|}
|}
if again and again rejected on all primary operators. Problem happend also if rejected by "Do not disturb" on operator phone.<!---->
http://wiki.innovaphone.com/index.php?title=Howto:Tweak_SIP_Response_Code_to_Q.931_Cause_Code_Mapping<!---->
      
      
=== PBX: Master/Slave compatibility problem with version 9 and version 8 and non-ascii characters in PBX name ===
=== PBX: Reporting licenses counting could be wrong if config templates were used ===


{|
{|
Line 4,558: Line 4,401:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67956 67956]
|[http://mantis.innovaphone.com/view.php?id=73466 73466]
|}
|}
In version 8 only latin1 characters were allowed, which means in unicode the high byte was always 0. So it could be ignored and when sending location information between master and slave sometimes the high byte contained 0xff.<br/><br/>In version 9 this non-ascii location information was not correct unicode at all.<br/><br/>The problem happened only if non-ascii characters were used when naming a PBX.<!---->
It could happen that a reporting license was used up by a config template object, if reporting was checked and other config templates where referenced<!---->
      
      
=== SIP: Generate new SRTP key on every incoming re-negotiation ===
=== myPBX: Allow overriding implicit allows ===


{|
{|
Line 4,569: Line 4,412:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67958 67958]
|[http://mantis.innovaphone.com/view.php?id=73471 73471]
|}
|}
Required to have different encryption after transfer.<!---->
It was not possible to define an allow with the same name. This is needed to override allows that are inherited from templates or group memberships.<!---->
      
      
=== PBX-CDR: Mobility calls to Trunk, external number was missing in CDR ===
=== IP241,IP222,IP232: Trap when scrolling down long call list ===


{|
{|
Line 4,580: Line 4,423:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67978 67978]
|[http://mantis.innovaphone.com/view.php?id=73494 73494]
|}
|}
only the number of the trunk itself was available<!---->
Out of memory trap when scrolling down long call list.<!---->
      
      
=== IP150: OEM specific WEB GUI modifications did not work ===
=== Trap: When using Diagnostics/Ping ===


{|
{|
Line 4,591: Line 4,434:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=67992 67992]
|[http://mantis.innovaphone.com/view.php?id=73536 73536]
|}
|}
manufacturer specific stylesheets have to be be adjusted to the new GUI style  <!---->
When leaving Diagnostics/Ping or Diagnostics/Traceroute web screen a trap may occur.<!---->
      
      
=== PBX: End of call intrusion was not signaled to the phone ===
=== PBX: Navigating in locations and groups if non-ascii charaters were used, failed with internet explorer ===


{|
{|
Line 4,602: Line 4,445:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68007 68007]
|[http://mantis.innovaphone.com/view.php?id=73567 73567]
|}
|}
The call intrusion tone was generated even if the intrusion was terminated<!---->
Internet explorer buggy concerning url encoding of href attributes<!---->
      
      
=== PBX: User Interface Inconsistency. Same thing was sometimes call 'Response Timeout'/' and 'No Response Time' ===
=== IPVA, crash-dump without backtrace, if  tracing was switched off internally ===


{|
{|
Line 4,613: Line 4,456:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68008 68008]
|[http://mantis.innovaphone.com/view.php?id=73574 73574]
|}
|}
Only Response Timeout used anymore<!---->
IPVA, crash-dump without backtrace, if  tracing was switched off internally<!---->
      
      
=== PBX-SOAP: Support UserClear for pending outgoing calls on Waiting Queues ===
=== ISDN: Do not provide tones and no signaling of inband info for unrestricted digital information calls ===


{|
{|
Line 4,624: Line 4,467:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68009 68009]
|[http://mantis.innovaphone.com/view.php?id=73603 73603]
|}
|}
UserClear was ignored in this state<!---->
There is equipment, which is doing unrestricted digital information ISDN calls, which gets confused if there is an ALERT message indicating inband tones (ringback).<!---->
      
      
=== SIP: Fix for media negotiation with SRTP ===
=== IP241,IP222,IP232: Symbol "new messages" and symbol "headset" do overlap in status bar ===


{|
{|
Line 4,635: Line 4,478:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68067 68067]
|[http://mantis.innovaphone.com/view.php?id=73608 73608]
|}
|}
Exchange of SRTP key may fail after hold/retrieve<!---->
Moved "headset" symbol to right side of status bar.<!---->
      
      
=== phone_orchid: DTMF Tones detected in voice data from microphone were propagated to remote side ===
=== IP-DECT: DECT endpoint update for multicast identities ===


{|
{|
Line 4,646: Line 4,489:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68075 68075]
|[http://mantis.innovaphone.com/view.php?id=73613 73613]
|}
|}
DTMF tones may be detected from audible feedback on pressing a dial key in connected state and also from some other source. It's better to propagate only tones requested explicitely via a dial key and not from some external source. <!---->
The update information to DECT is changed if the user logs out a handset. This is used to update the multicast identities in some OEM handsets.<!---->
      
      
=== SIP: Trap handling 491 response on reliable transport ===
=== Ip6010 DSP Allow calls to start with T.38 ===


{|
{|
Line 4,657: Line 4,500:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68093 68093]
|[http://mantis.innovaphone.com/view.php?id=73626 73626]
|}
|}
Trap with MAX_BUSY_TICKS<!---->
Needed for some T38 fax applications<!---->
''Status:''
ac_dsp3.cpp
      
      
=== phone_inca: "ETH0/Isolate PC Link" checkmark could not be cleared via WEB UI once set ===
=== IP241,IP222,IP232: Missing "Recall possible" display text on call completion ===


{|
{|
Line 4,668: Line 4,513:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68098 68098]
|[http://mantis.innovaphone.com/view.php?id=73628 73628]
|}
|}
Only a WEB UI problem, a "config rem ETH0 /isolate-pc" did help.<!---->
Missing display text on 'recall possible' notification.<!---->
      
      
=== SIP: Trap when outgoing SIP call is cancelled while DNS resolving is ongoing ===
=== SIP: Media negotiation problem on outbound call from BC Conference object ===


{|
{|
Line 4,679: Line 4,524:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68099 68099]
|[http://mantis.innovaphone.com/view.php?id=73652 73652]
|}
|}
Trap when outgoing SIP is cancelled while DNS resolving is ongoing<!---->
INVITE was sent without SDP offer.<br/>test/9.00/pbx/conf/SIP/002-conf-with-bcast.xml<!---->
      
      
=== Gateway: Allow configuration of username and password for ENUM/SIP interfaces ===
=== IP241,IP222,IP232: Extra digits dialed when using RIGHT key on screnn 'indrect dialling' ===


{|
{|
Line 4,690: Line 4,535:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68147 68147]
|[http://mantis.innovaphone.com/view.php?id=73694 73694]
|}
|}
For rare where remote destination server asks for authentication.<br/>(And all remote destination servers ask for same auth or remote destination server s always the same.)<!---->
'123123' instead of '123' was dialed.<!---->
      
      
=== SIP: Interoperability with LinkSys SPA3102 ===
=== ISDN Trunk: Transfer to ISDN Trunk with TONE interface failed ===


{|
{|
Line 4,701: Line 4,546:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68174 68174]
|[http://mantis.innovaphone.com/view.php?id=73695 73695]
|}
|}
LinkSys SPA3102 gives "g729a" as RTP payload type mapping:<br/><br/>    v=0<br/>    o=- 510843041 510843041 IN IP4 192.168.10.20<br/>    s=-<br/>    c=IN IP4 192.168.10.20<br/>    t=0 0<br/>    m=audio 16404 RTP/AVP 18 100 101<br/>    a=rtpmap:18 G729a/8000<br/>    a=fmtp:18 annexb=no<br/>    ...<br/><br/>Needs to be handled.<!---->
There was not media after the transfer<!---->
      
      
=== IP241: Fix for display rendering ===
=== H.323: Renegotiation to SRTP after dialtone failed ===


{|
{|
Line 4,712: Line 4,557:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68181 68181]
|[http://mantis.innovaphone.com/view.php?id=73698 73698]
|}
|}
Display of diversion destination was corrupt.<br/>Pickup fkey labeling was wrong when party information needed truncation.<!---->
Happens with Escape Dialtones configured in PBX and calls with SRTP enabled<!---->
      
      
=== Gerneral/Admin page was broken if too many authentication servers were configured ===
=== Option added to wait  for the DSP to disconnect fax, needed only for some fax devices. ===


{|
{|
Line 4,723: Line 4,568:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68231 68231]
|[http://mantis.innovaphone.com/view.php?id=73748 73748]
|}
|}
The number of authentication servers is now restricted to 10.<!---->
Use the prodedure of <br/>Mantis 67821: ip22/24/28/302/305: DSP affected by certain fax tones<br/>only if the option /t38-wait-fax-close set.<br/><br/>This avoids traps caused by disconnect timeouts if the DSP response is not received in time.<br/><!---->
''Status:''
ac_dsp3.cpp<br/>ac_dsp3.h<br/>dsp.xsl
      
      
=== SIP: Interoperability with Lync ===
=== SIP: Using wrong remote port when registering ===


{|
{|
Line 4,734: Line 4,581:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68232 68232]
|[http://mantis.innovaphone.com/view.php?id=73784 73784]
|}
|}
Media negotiation problem on calls coming from on Lync client<br/>and getting forwarded to another Lync client.<br/><!---->
Only affects IP-DECT when handset is switched OFF and ON and if the SIP runs on non-standard port.<!---->
      
      
=== IP241: Updated some display text information ===
=== IP232: Memory leak in display rendering ===


{|
{|
Line 4,745: Line 4,592:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68234 68234]
|[http://mantis.innovaphone.com/view.php?id=73864 73864]
|}
|}
No need to strongly abbreviate disconnect cause text.<!---->
Memory leak in display rendering<!---->
   
= V9 Hotfix 7 (90600.08) =
Changes included in Version 9 hotfix7
[http://mantis.innovaphone.com/view.php?id=75932 Definition]
 
== New Features ==
 
 
      
      
=== Phone: Added loud note to web-ui that bool funnction key can only toggle if boolean object addressed by number ===
=== IP241,IP222,IP232: Some diagnostics to check phone's ui memory consumption ===


{|
{|
Line 4,756: Line 4,611:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68240 68240]
|[http://mantis.innovaphone.com/view.php?id=74049 74049]
|}
|}
Description: Phone: Added loud note to web-ui that bool funnction key can only toggle if boolean object addressed by number <br/><br/><!---->
Where have all the memory gone.<!---->
      
      
=== phone: intrusion call started in handset mode is not terminated when going on hook when TAPI or operator run on PBX ===
=== SIP: Support for "Content-ID" in "multipart/alternative" bodies ===


{|
{|
Line 4,767: Line 4,622:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68249 68249]
|[http://mantis.innovaphone.com/view.php?id=74161 74161]
|}
|}
With TAPI or operator running on the PBX the the signaling of a busy condition is changed such that a disconnect instead of a release is sent. The disconnect was not handled correctly, the hookswitch state was lost and the next on-hook signal was ignored. TThus teh call could be terminated with the disc-key only.<br/><!---->
Support for "Content-ID" in "multipart/alternative" bodies<!---->
      
      
=== Phone: Switch presence fkey light on if presence activity is set ===
=== IP241,IP222,IP232: Added drop shadow to lower end of menu screens ===


{|
{|
Line 4,778: Line 4,633:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68256 68256]
|[http://mantis.innovaphone.com/view.php?id=74278 74278]
|}
|}
Switch presence fkey light on if presence activity is set<!---->
To make end of list of menu entries more discernible.<!---->
      
      
=== myPBX: Wrong charset was used when authenticating against reporting or changing user password ===
=== myPBX: Support static URLs for application sharing ===


{|
{|
Line 4,789: Line 4,644:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68258 68258]
|[http://mantis.innovaphone.com/view.php?id=74389 74389]
|}
|}
Javascript uses Unicode. The PBX and the reporting use UTF8. Therefore strings have to be converted to UTF8 before encrypting them using RC4.<!---->
In some application sharing solutions a fixed link can be used to create and join meetings (GoMeetNow, BeamYourScreen).<br/><br/>There are two URLs configured, one for the presenter and one for attendees. When the user clicks the aplication sharing button the links are sent using chat messages.<br/><!---->
      
      
=== IP241: New fkey backgrounds ===
=== PBX: Tracing flag turns on tracing in all dyn PBX's as well ===


{|
{|
Line 4,800: Line 4,655:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68304 68304]
|[http://mantis.innovaphone.com/view.php?id=74390 74390]
|}
|}
Less transparency to avoid interference with personalized background images<!---->
Helpfull to debug dyn PBX setups<!---->
      
      
=== myPBX: Chunked response from application sharing provider did not work ===
=== debug method raw_ethernet ===


{|
{|
Line 4,811: Line 4,666:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68311 68311]
|[http://mantis.innovaphone.com/view.php?id=74897 74897]
|}
|}
After receiving HTTPCLIENT_RECV_RESULT, pbx_client did not send another HTTPCLIENT_RECV.<!---->
A new debug method to trace raw ethernet data.<!---->
      
      
=== phone_orchid: call pickup via partner/pickup key fails when the key is pressed immediately after audible signal ===
=== Support for new hardware ===


{|
{|
Line 4,822: Line 4,677:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68313 68313]
|[http://mantis.innovaphone.com/view.php?id=74990 74990]
|}
|}
When "Audible Signal after alerting" was configured on a partner/pickup key and the key was pressed while or a short time after the audible signal was played then the picked call was mute. <!---->
.<!---->
''Status:''
files: ac_dsp3.cpp
      
      
=== IP241: Activate external background image from phone menu ===
=== possible use of push-pull drivers for new hardware ===


{|
{|
Line 4,835: Line 4,688:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68351 68351]
|[http://mantis.innovaphone.com/view.php?id=75051 75051]
|}
|}
External background image source can be configured on web ui.<br/>Background image can be selected on phone menu.<br/>Now also external background image can be selected.<!---->
not visible to customers<!---->
      
      
=== myPBX: Hide passwords for application sharing and reporting in config ===
=== PBX/IP-DECT: User password length ===


{|
{|
Line 4,846: Line 4,699:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68352 68352]
|[http://mantis.innovaphone.com/view.php?id=75453 75453]
|}
|}
The passwords are moved from the config line to the VARS.<br/><br/>PBX0/MY-A - Application sharing password<br/>PBX0/MY-R - Reporting password<!---->
The maximum user password length in the PBX and IP-DECT is increased to 23.<!---->
      
      
=== SIP: Failed to register on dynamic PBX ===
=== H.323: Support for switch back to local media on endpoint ===


{|
{|
Line 4,857: Line 4,710:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68362 68362]
|[http://mantis.innovaphone.com/view.php?id=75522 75522]
|}
|}
Attempt to register on a dyn PBX with SIP protocol was rejected with "301 Moved Permanently".<!---->
needed to resume normal call after a conference using external conference unit<!---->
      
      
=== IP-DECT: Adding OEM radios to Kerberos realm did not work with passwords containing special characters ===
=== IP232: Touch gesture to open directory-search/indirect-dialing screen ===


{|
{|
Line 4,868: Line 4,721:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68377 68377]
|[http://mantis.innovaphone.com/view.php?id=75549 75549]
|}
|}
The password was not URL-decoded when reading it from the UI.<!---->
Vertical upwards on idle screen opens directory-search/indirect-dialing screen.<br/>Vertical downwards on idle screen opens main-menu screen.<!---->
      
      
=== SIP: Pending control calls on gateway ===
=== PBX-SOAP: Wsdl Versioning mechanism fixed ===


{|
{|
Line 4,879: Line 4,732:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68378 68378]
|[http://mantis.innovaphone.com/view.php?id=75552 75552]
|}
|}
Incoming unsolicited NOTIFY(message-summary) may cause pending control call on Gateway.<br/>Control calls are calls (signaling connections) without media channel.<br/>These calls are now released.<!---->
Better mechanism, to avoid constantly adding arguments to the Version function with new Versions.<!---->
      
      
=== WEB GUI page cannot be  scrolled completely when height of left hand logo is too big ===
=== different port naming convention for new hardware ===


{|
{|
Line 4,890: Line 4,743:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68382 68382]
|[http://mantis.innovaphone.com/view.php?id=75575 75575]
|}
|}
Height of FHF logo is bigger than that of the default logo, this must be considered when computing size for iframe below tab lines. <!---->
BRI1..5 instead of TEL1..4+PPP<!---->
      
      
=== DTMF user configuration with invalid checkbox check for presence setting ===
=== PBX-SOAP: UserHold without MOH to local User ===


{|
{|
Line 4,901: Line 4,754:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68383 68383]
|[http://mantis.innovaphone.com/view.php?id=75577 75577]
|}
|}
The check of the checkmark of the presence setting was wrong.<!---->
UserHold was sending MOH to the local and the remote User. With the argument remote=true, the MOH is sent to the remote user only<!---->
      
      
=== IP222: Alpha input using the num block ===
=== myPBX launcher: Use default configuration from local machine registry hive ===


{|
{|
Line 4,912: Line 4,765:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68398 68398]
|[http://mantis.innovaphone.com/view.php?id=75634 75634]
|}
|}
Automatic switching back to numeric mode when leaving input ctrl.<!---->
If no user configuration is given, the following values are copied from the local machine hive.<br/><br/>"Software\\innovaphone\\myPBX\\URL"<br/>"Software\\innovaphone\\myPBX\\secondaryURL"<br/><br/>Also the autostart checkmark is disabled, if autostart is activated, globally. Autostart is controlled with the following registry key.<br/><br/>"Software\\Microsoft\\Windows\\CurrentVersion\\Run\\innovaphone myPBX"<!---->
      
      
=== SIP: Trap when using TLS as transport ===
=== SIP: support for proprietary message header ===


{|
{|
Line 4,923: Line 4,776:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68410 68410]
|[http://mantis.innovaphone.com/view.php?id=75864 75864]
|}
|}
Only if remote side closes transport connection while requests are pending.<!---->
Support for proprietary message header (CAL)<!---->
      
      
=== phone_orchid: begin of voice mail prompt was cut off sometimes ===
=== H.323: Call to SIP Trunk with MediaRelay and exclusive coder renegotiation from TONE to early media did not work ===


{|
{|
Line 4,934: Line 4,787:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68416 68416]
|[http://mantis.innovaphone.com/view.php?id=75959 75959]
|}
|}
Connection of Voip to DSP channel was delayed and thus the first packets of the RTP stream were lost.<!---->
No ringback and no audio after connect<!---->
      
      
=== body onload attribute can be extended and tab_active method doesn't crash anymore ===
=== SIP: Added Call-Info header to re-INVITE for hold/retrieve ===


{|
{|
Line 4,945: Line 4,798:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68430 68430]
|[http://mantis.innovaphone.com/view.php?id=76011 76011]
|}
|}
Custom PBX object XSL had no method anymore to set the onload attribute of the body. This can be now extended with a XSL template parameter.<br/>Additionally the tab_active method has been called by default and the default value caused the method to crash.<!---->
Call-Info: <urn:X-cisco-remotecc:hold><br/>Call-Info: <urn:X-cisco-remotecc:resume><!---->
      
      
=== X509: Fix for reading innovaphone info from flash ===
=== SIP: Support for active call pickup acc. to RFC-3891 ===


{|
{|
Line 4,956: Line 4,809:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68435 68435]
|[http://mantis.innovaphone.com/view.php?id=76035 76035]
|}
|}
Parsing the innovaphone info text was incorrect<!---->
Sending INVITE with Replaces header to do pickup at SIP PBX's<!---->
      
      
=== License: Be safe against factory reset during license invalidation ===
=== phone ip222, ip232 : USB headset support (beta) ===


{|
{|
Line 4,967: Line 4,820:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68447 68447]
|[http://mantis.innovaphone.com/view.php?id=76463 76463]
|}
|}
If factory reset is done before license invalidation procedure is complete,<br/>will keep you from completing the license invalidation.<br/>Now the procedure can be completed even after factory reset.<!---->
general support for USB headsets added, some headsets are tested and working, others will follow soon <!---->
      
      
=== IP241 - handsfree speaker volume to low ===
=== phone - Headset Function Key can be configured now as enable/disable or as call control key ===


{|
{|
Line 4,978: Line 4,831:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68451 68451]
|[http://mantis.innovaphone.com/view.php?id=76468 76468]
|}
|}
The handsfree speaker volume was too low even when configured to maximum. Now the general output volume is increased by 3 dB. In case of problems the general output volume can be changed by<br/>    config add AC-DSP0 RINGER /VoiceOutputGain n<br/>with n = 1..63 -> (-32 + n)db, n = 32 -> 0dB, n = 0 -> mute<!---->
a headset key mode can be configured to use the key either to enable/disable the headset (Mode: Enable) or to start/accept/clear calls via headset (Mode: Control)<!---->
   
== Bug Fixes ==
 
 
      
      
=== wrong link to PPP Interface State Info Help ===
=== Ip6010 DSP Allow coder change from T38 to voice and back to T38 with local DSP ===


{|
{|
Line 4,989: Line 4,846:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68463 68463]
|[http://mantis.innovaphone.com/view.php?id=73193 73193]
|}
|}
Link out of V9 namespace<!---->
..<!---->
''Status:''
ac_dsp3.cpp<br/>
      
      
=== phone: DHSG headset not reset to idle after a hookswitch signal in idle state ===
=== ISDN: Enable fax detection only after connect ===


{|
{|
Line 5,000: Line 4,859:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68567 68567]
|[http://mantis.innovaphone.com/view.php?id=73213 73213]
|}
|}
most DHSG headsets generate a hookswich signal and enter voice mode when taken out of basestation. This hookswitch signal was simply ignored.<br/>Now the voice mode is cleared after one second if there is no other DHSG event before.<!---->
On ISDN networks it can happen that the Connect message is delayed. This way fax tones are forwarded to the caller before the caller has received this Connect.<br/><br/>This way a renegotiation on voip to fax could happen before the connect, which is not supported by sip.<!---->
      
      
=== SIP/TCP: Transport error when connection is closed by client ===
=== Phones: Update of phone text data ===


{|
{|
Line 5,011: Line 4,870:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68578 68578]
|[http://mantis.innovaphone.com/view.php?id=73775 73775]
|}
|}
If transaction client closes connection before final response has been sent,<br/>the server tries to open a new connection toward ephemeral port of closed connection.<!---->
External input<!---->
      
      
=== SIP: Fix for Dialog-Info notification ===
=== SIP: Handling of collision of transfer and release ===


{|
{|
Line 5,022: Line 4,881:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68581 68581]
|[http://mantis.innovaphone.com/view.php?id=73936 73936]
|}
|}
Send an empty dialig-info XML after inbound subscription.<br/>Required for interop with Grandstream GXP2010.<!---->
If one end releases a call while the other initiates an attended transfer, a "ghost call" may remain.<br/>Resource leak.<br/><!---->
      
      
=== SIP: Use secondary proxy address on interfaces without registration ===
=== IP241,IP222,IP232: Limitation of background image size ===


{|
{|
Line 5,033: Line 4,892:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68635 68635]
|[http://mantis.innovaphone.com/view.php?id=73957 73957]
|}
|}
When secondary proxy address resolved by DNS it should be used to send INVITE to.<!---->
Limit is 512 kBytes now.<br/>To keep phone from out of memory trap.<!---->
      
      
=== SIP: Handling of 491 response on reliable transport ===
=== H.323: Renegotiation after PBX Node dialtone failed sometimes ===


{|
{|
Line 5,044: Line 4,903:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68652 68652]
|[http://mantis.innovaphone.com/view.php?id=74020 74020]
|}
|}
Re-try of re-INVITE was missing.<!---->
This depended on timing. This happened if the call was sent after the dialtone to a master over a slow link.<!---->
      
      
=== SIP: Treat domain part of SIP URI case-insensitive ===
=== IP232: Open phone menu by touch gesture ===


{|
{|
Line 5,055: Line 4,914:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68653 68653]
|[http://mantis.innovaphone.com/view.php?id=74046 74046]
|}
|}
According to RFC3261(19.1.4)<!---->
Touching header bar and moving vertically down opens the phone menu.<br/>No need to use center key of 4-way control.<!---->
      
      
=== SIP: Fix for REFER handling ===
=== Dect User Gui: AD-replicated objects weren't listed under certain conditions ===


{|
{|
Line 5,066: Line 4,925:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68663 68663]
|[http://mantis.innovaphone.com/view.php?id=74051 74051]
|}
|}
Support for attended transfer between two calls from/to different SIP endpoints.<br/>As long as both calls belong to the same signaling interface.<!---->
Deleted object thate were re-replicated werent't visible in the Dect user GUI. A mandatory Ldap attribute was missing under such circumstances.<br/><!---->
      
      
=== SIP: Problem decoding INFO(application/dtmf-relay) ===
=== IP222: Handset Microphone level too low ===


{|
{|
Line 5,077: Line 4,936:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68667 68667]
|[http://mantis.innovaphone.com/view.php?id=74073 74073]
|}
|}
DTMF digit was not decoded from message body if whitespace between EQUAL and DIGIT.<br/>E.g. Signal= 5<!---->
Handset micro increased by 6db<br/>Handsfree micro unchanged<br/><!---->
''Status:''
ac_codec3.cpp<br/>
      
      
=== Phone: Changing config option /sip-hold does not call for reset ===
=== IP222,IP232: Language configuration not saved ===


{|
{|
Line 5,088: Line 4,949:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68691 68691]
|[http://mantis.innovaphone.com/view.php?id=74087 74087]
|}
|}
Reset is required and 'reset required" must be displayed.<!---->
Language can be selected on phone menu, but change is neither saved nor applied.<!---->
      
      
=== SIP: Web UI for cause code mapping (fix) ===
=== SIP: Don't take NOTIFY(application/simple-message-summary) with "Do-Not-Disturb: no" as FTY_MWI_DEACTIVATE ===


{|
{|
Line 5,099: Line 4,960:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68772 68772]
|[http://mantis.innovaphone.com/view.php?id=74162 74162]
|}
|}
Did not work as expected<!---->
Interop with Nortel CS1k PBX<!---->
      
      
=== SIP: Out-Of-Memory trap if Group Indications are enabled ===
=== Phones: Fine grained function hiding did not disable Fkey configuration ===


{|
{|
Line 5,110: Line 4,971:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68804 68804]
|[http://mantis.innovaphone.com/view.php?id=74174 74174]
|}
|}
If 'Group Indications' are enabled on a PBX object where a SIP client registers,<br/>an OOM trap may occur in case of heavy call activity on PBX.<br/><!---->
Value 0x00000800 (PHONE_HIDE_US_FUNCTION_KEYS) hides function keys entry from user-list,<br/>but fkeys can still be (re)configured by long fkey press.<br/><br/>For more information see:<br/>http://wiki.innovaphone.com/index.php?title=Reference9:Phone/Protect<br/><!---->
      
      
=== Phone: "Keep Calling Party Info on Pickup Key" option from Phone-&gt;Preferences not working, fixed ===
=== myPBX: Allow asterisk as a wildcard in LDAP search strings ===


{|
{|
Line 5,121: Line 4,982:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68820 68820]
|[http://mantis.innovaphone.com/view.php?id=74200 74200]
|}
|}
Phone: "Keep Calling Party Info on Pickup Key" option from Phone->Preferences not working, fixed<!---->
As it is done on our phones.<!---->
      
      
=== Phone: DnD Absence Message containing newline dumped to config-file breaks phone upon config-file upload ===
=== IP232: Pixel noise with some color values ===


{|
{|
Line 5,132: Line 4,993:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68898 68898]
|[http://mantis.innovaphone.com/view.php?id=74221 74221]
|}
|}
Solution: All flashdir entries cotaining newline now dumped as binary.<br/><br/><!---->
- changed to lower pixel clock<!---->
''Status:''
orchid_lcd.cpp
      
      
=== IP30x V9 hf2 media problem: Echo canceller not reliable ===
=== PBX: Conference to VM ===


{|
{|
Line 5,143: Line 5,006:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68901 68901]
|[http://mantis.innovaphone.com/view.php?id=74222 74222]
|}
|}
Sometimes voice from ISDN/Analog to the IP is muted after some time.<br/>Seems to be a problem in the latetest echocanceller.<br/>Change to old DSP code until fixed DSP code is available.<br/><!---->
Now a call to the VM as a conference member works again.<!---->
''Status:''
      
ip24.mak
=== RTP-DTMF: Digit may get lost during media re-negotiation ===
      
=== Denial of Service filter in ethernet library did not work ===


{|
{|
Line 5,156: Line 5,017:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68907 68907]
|[http://mantis.innovaphone.com/view.php?id=74232 74232]
|}
|}
this filter can be useful to prevent DOS attacks on non routing devices <!---->
Receiver starts handling on END event.<br/>Sender may stop sending before END event was sent.<!---->
      
      
=== SIP: Trap when handling SUBSCRIBE on federation interfaces ===
=== No media for calls from SIP provider forwarded back to SIP provider in case of NAT and MediaRelay ===


{|
{|
Line 5,167: Line 5,028:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68976 68976]
|[http://mantis.innovaphone.com/view.php?id=74290 74290]
|}
|}
SIP client object may be deleted while DNS query is pending.<br/>But only under critical timing conditions.<br/><!---->
NAT traversal depends on a packet being sent from inside the NAT to outside, to fix the RTP destination of the outside endpoint. This does not happen if both endpoints are outside.<br/><br/>Dummy packets are sent from the Media Relay function in this case to achieve this.<!---->
      
      
=== myPBX: Use display name for sending meeting urls and conference numbers ===
=== ISDN: Send HLC with mobility calls ===


{|
{|
Line 5,178: Line 5,039:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68992 68992]
|[http://mantis.innovaphone.com/view.php?id=74296 74296]
|}
|}
For normal chat messages the DN is used as the sender name. For sending the links for application sharing or audio conferences the CN was used.<!---->
Some ISDN networks refuse the forwarding of a call to a mobile network if no HLC (High Layer Compatibility) Information Element indicating Telephony is included in the call.<!---->
      
      
=== Phone: Hide calling party on Pickup key fixed ===
=== IP241,IP222,IP232: Improved phone screen rendering ===


{|
{|
Line 5,189: Line 5,050:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68995 68995]
|[http://mantis.innovaphone.com/view.php?id=74301 74301]
|}
|}
Phone: Hide calling party on Pickup key fixed<br/><!---->
Reduce map copy operations.<!---->
      
      
=== SoftwarePhone: No CT setup with remote connected calls ===
=== SIP: Accept INVITE's from alternative proxy ===


{|
{|
Line 5,200: Line 5,061:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68996 68996]
|[http://mantis.innovaphone.com/view.php?id=74310 74310]
|}
|}
The call transfer setup facility is removed in the call setup if the call is a by remote control connected call used in case of outgoing calls with myPBX. This fixes an empty diverting party number information element in the PBX. Now it can be used with a trunk PBX object with the enabled option 'Set Calling=Diverting No', otherwise the calling party number was removed within this object.<!---->
If alternative proxy is configured, calls should accepted from alternative proxy,<br/>even if primary proxy is alive and kicking.<!---->
      
      
=== PBX: Conference trap ===
=== Flash Directory: Substring match could miss results ===


{|
{|
Line 5,211: Line 5,072:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68998 68998]
|[http://mantis.innovaphone.com/view.php?id=74320 74320]
|}
|}
A zero pointer trap in the broadcast conference PBX object is fixed.<!---->
Searching by substring worked case sensitive in some cases, instead of functioning case insensitive.<!---->
      
      
=== IP222/IP232: Hookswitch on some PCB not working ===
=== PBX: CF at Gateway Type objects - additional dialed digits should be added to the destination ===


{|
{|
Line 5,222: Line 5,083:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69001 69001]
|[http://mantis.innovaphone.com/view.php?id=74348 74348]
|}
|}
Reduce LED frequency from 12kHz to 6 kHz<!---->
This way a CFNR at a trunk object can be used to reroute the call to another trunk.<!---->
''Status:''
orchid_lcd.cpp
      
      
=== phone_orchid - builtin speaker test did not work ===
=== Gateway: Deleting of routes could result in duplication of routes ===


{|
{|
Line 5,235: Line 5,094:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69026 69026]
|[http://mantis.innovaphone.com/view.php?id=74356 74356]
|}
|}
works only over channel 0<!---->
This happened if an interface registration was disabled, for which automatic routes have been generated and then a route was deleted. The last route was duplicated.<!---->
      
      
=== myPBX: Login did not work with system names containing special characters ===
=== IP241/222/232: Monitormode (Lauthören): Level too low ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69033 69033]
|[http://mantis.innovaphone.com/view.php?id=74362 74362]
|}
|}
Missing URL encoding when sending the connect-request message.<!---->
Use independent analog codec channels for speaker and headset receiver.<br/>Speaker volume in monitor mode is configured as in handsfree mode.<br/><br/>To change to speaker level in monitor mode the gain of the speaker can be configured with<br/> config change AC-DSP0 RINGER /DualOutputModeGain level<br/> config activate<br/><br/>level is from 0..63  0<br/>0 -> -32dm <br/>32 -> 0db<br/>63 -> 31db<br/><br/><!---->
''Status:''
ac_codec3.cpp<br/>ac_codec3.h
      
      
=== PBX Mobility: Mobility object hanging if call establishment with SOAP/myPBX canceled ===
=== IP241,IP222,IP232: Could not activate first builtin background image from phone menu ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69052 69052]
|[http://mantis.innovaphone.com/view.php?id=74367 74367]
|}
|}
If a call for a mobile phone is initiated by SOAP or myPBX, a call is first sent to the mobile phone. If the mobile phone accepts the outgoing call to the destination is initiated. If the mobile phone did not accept the initial call, no other calls could be done from then on.<!---->
Could not activate first builtin background image from phone menu if external background image is currently active.<!---->
      
      
=== IP-DECT: Configuration of Media preferences did not work anymore ===
=== IP241,IP222,IP232: Show multiple diversions on incoming calls ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69056 69056]
|[http://mantis.innovaphone.com/view.php?id=74371 74371]
|}
|}
Media preference were ignored. This was a collateral damage introduced with fixes from the last hotfix<!---->
If a call was diverted more than once, the phone shows first diverting party (original called number) and last diverting party.<br/>(Not only the last diverting party)<!---->
      
      
= V9 Hotfix 4 (90600.04) =
=== myPBX launcher: Unhandled exception when accessing browser object ===
Changes included in Version 9 hotfix4
[http://mantis.innovaphone.com/view.php?id=69495 Definition]
 
== New Features ==
 


{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=74380 74380]
|}
The myPBX launcher could crash when accessing the browser document because of an unhandled exception.<!---->
      
      
=== Improved protection against Denial of Service attacks ===
=== IP232: Pickup fkey displays no information in active state ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69166 69166]
|[http://mantis.innovaphone.com/view.php?id=74392 74392]
|}
|}
flooding a box with different kinds of packets may lead to out of memory conditions. The Denial of Service filter in the ethernet layer is activated where required. TCP listening sockets have a backlog limit now. The http service restricts the number of half-open sessions and limits the number of concurrent sessions according to the total memory available on a box.<!---->
Pickup fkey displays no information in active state<!---->
      
      
=== ISDN interop issue with SecuGATE LI 30 from Sirrix ===
=== Phones: Only one diverting party displayed on incoming calls ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69168 69168]
|[http://mantis.innovaphone.com/view.php?id=74406 74406]
|}
|}
The SecuGATE LI30 is sending/receiving ISDN INFO messages in Call Proceeding State (State 3 and state 9), which was not supported<!---->
Only original called party displayed, not the last diverting party.<br/>Phones with non-color display only (IP240,IP230,IP200,IP11,IP150)<br/><br/><!---->
      
      
=== PBX Mobility: Support of transfer on mobile side, not using mobility ===
=== NAT: Permanent UDP forwarding did not work ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69275 69275]
|[http://mantis.innovaphone.com/view.php?id=74549 74549]
|}
|}
In case that the mobile phone transfers the call to another destination, this call must be removed from the mobility function, so that the mobility function is available for another call<!---->
Permanent UDP forwarding was unusable after first forwarded datagram.<!---->
      
      
=== USB Driver merge from v10 to v9 ===
=== PBX Waiting: If used as outgoing dialing device, coder list sent with the call was not correct ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69288 69288]
|[http://mantis.innovaphone.com/view.php?id=74593 74593]
|}
|}
First step to support USB headsets. This is not functional yet.<!---->
Should be based on the coders paramter in the configured URL<!---->
      
      
=== SIP: New config file option /hold-notify-as-inactive ===
=== Phone_orchid: Residual echo in handset ( sporadic) ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69293 69293]
|[http://mantis.innovaphone.com/view.php?id=74671 74671]
|}
|}
If set, holdNotific is interworked into "a=inactive".<br/>If not, holdNotific is interworked into "a=sendonly".<!---->
Echo canceller NLP sensitivity Mode changed to  2 <!---->
''Status:''
ac_dsp3.cpp
      
      
=== SIP: New config file option /prefer-pai2 ===
=== IP222,IP232: Function key "Hotdesk" did not work ===


{|
{|
Line 5,342: Line 5,208:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69459 69459]
|[http://mantis.innovaphone.com/view.php?id=74723 74723]
|}
|}
Interoperability with Telepo:<br/>When receiving INVITE, get calling party id from second P-Asserted-Id header.<!---->
Function key "Hotdesk" did not work<!---->
      
      
=== new: DHCP manufacturer specific option 'boot-cfg' provides an URL to read config from after any reset except 'creset' ===
=== IP232: Fix for long touch on fkey ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69472 69472]
|[http://mantis.innovaphone.com/view.php?id=74743 74743]
|}
|}
This option permits to boot boxes with a fresh config provided via TFTP/HTTP without storing the config on the device.<br/>It is intended to be used as follows:<br/><br/>1. the box is started with DHCP enabled (no initial configuration)<br/>2. the box contacts the DHCP server and gets the ip-address and also the Vendor Specific Information in option 43.<br/>  Suboption 249 of the Vendor Specific Information specifies the URL of the boot config file.<br/>3. the box polls the TFTP/HTTP server for the config file.<br/>4. the box reads the config file and executes the commands provided in the file<br/><br/>The URL may contain the same meta-character strings an Update Server URL, for example #m (mac-address)<br/>The length of the URL in the DCHCP suboption is restricted to 127 characters.<br/><br/>The URL is polled in 5 second intervals.<br/>The config file is read and executed by the update process in the usual way.<br/>A 'creset' commmand as last command of the file will restart the box with the new configuration without writing any 'config' command options to the flash.<br/>After a restart by the 'creset' commmand the boot-cfg URL is ignored.<br/>After a restart by any other of the 'reset' commands or by a power cycle the boot-cfg URL is processed again.<br/><br/>On an Innovaphone DHCP-Server configuration of a boot-cfg URL and providing it to clients via suboption 249 must be explicitely enabled by<br/>    config add DHCP0 /boot-cfg<br/>    config write<br/>    config activate<br/>Once enabled the URL may be entered under "IP4/ETH0/DHCP-Server/Boot Config URL" and is provided to all clients then.<br/><br/>If an Innovaphone DHCP-Client receivess a boot-cfg URL it is displayed under "IP4/ETH0/DHCP/Boot Config URL".<br/><br/><!---->
Opens fkey config, but mist not toggle control on touch-off.<!---->
      
      
=== auto complete dtmf feature codes with '#' after 2 seconds ===
=== Web-UI: Configuration of "Sync Server" does not return with HTML page ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69561 69561]
|[http://mantis.innovaphone.com/view.php?id=74778 74778]
|}
|}
Optional feature for phones, which are not able to send a '#', e.g. the iPhone.<br/>They dial a feature like a cfu with a destination number and after two seconds, the feature code is automatically completed with a '#'.<!---->
Configuration of "Sync Server" does not return with HTML page<!---->
      
      
=== PBX: HTTP request to initiate call for mobile phone ===
=== IP222,IP232: changes not saved when leaving menu screen with LEFT key ===


{|
{|
Line 5,375: Line 5,241:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69570 69570]
|[http://mantis.innovaphone.com/view.php?id=74792 74792]
|}
|}
To improve GSM client functionality<!---->
Even after confirming to save<!---->
      
      
=== PBX: WSDL for SOAP API readable from box ===
=== IPxx10: Flashdir Segments Default to 129(was 51) ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69688 69688]
|[http://mantis.innovaphone.com/view.php?id=74826 74826]
|}
|}
Simplifies Application development in some environments<!---->
IPxx10: Flashdir Segments Default to 129(was 51)<!---->
      
      
=== SIP: Get display information from Call-Info header in register refresh response ===
=== Interop: Don't add an error log "SRTCP authentication failed" on non-RTCP packets ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69758 69758]
|[http://mantis.innovaphone.com/view.php?id=74903 74903]
|}
|}
Get display information from Call-Info header in 200/OK<!---->
Microsoft Lync send invalid RTP and RTCP packets at the beginning of each call.<br/>In case of encrypted media, decryption of these packets fails.<!---->
      
      
=== IP-DECT: Display update ===
=== phone_orchid: ethernet statics counters sometimes displayed as negative numbers ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69770 69770]
|[http://mantis.innovaphone.com/view.php?id=74913 74913]
|}
|}
Now, the endpoint's display is updated if the registration forwards a display update.<!---->
the unsigned counters were printed as signed integers<!---->
      
      
=== Voicemail &lt;pbx-query-obj&gt; queries new properties ===
=== IP241,IP222,IP232: Volume control not displayed when already at maximum ===


{|
{|
Line 5,419: Line 5,285:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69781 69781]
|[http://mantis.innovaphone.com/view.php?id=74927 74927]
|}
|}
"user_type": returns "1"(endpoint) or "2"(other object like gateway)<br/><br/>"pseudo_type": returns types alike "vm", "bc_conf", "trunk"<br/><br/>"pseudo_text": returns friendly texts alike "Voicemail", "Bc Conference", "Trunk Line"<br/><br/><!---->
Volume control is not displayed when trying to increase while being at maximum.<!---->
      
      
== Bug Fixes ==
=== IP6000 crypto driver: Trap when buffers are depleted ===
 
 
   
=== Kerberos: Protect against ping pong attacks ===


{|
{|
Line 5,434: Line 5,296:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68822 68822]
|[http://mantis.innovaphone.com/view.php?id=74935 74935]
|}
|}
Do not answer with an error message to unexpected or malformed messages.<br/><br/>This protects against the "Kerberos Server Spoofed Packet Amplification DoS" attack. The attack causes two Kerberos servers to send each other error messages in a ping pong style.<!---->
Avoid the trap and log an Event when the buffers are depleted.<!---->
      
      
=== RTP-DTMF: Start handling of RTP-DTMF on reception of END event ===
=== IP241,IP222,IP232: Mute status indication does not work ===


{|
{|
Line 5,445: Line 5,307:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=68938 68938]
|[http://mantis.innovaphone.com/view.php?id=74964 74964]
|}
|}
Workaround for Bug in MS Lync / Mediation server.<br/>Mediation server changes destination port while sending RTP-DTMF redundancy retransmissions.<!---->
"MICROPHONE OFF" is displayed on status bar now.<!---->
      
      
=== SIP: Presence interoperability with ESTOS UC server ===
=== SIP: Rare problem with blind transfer ===


{|
{|
Line 5,456: Line 5,318:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69050 69050]
|[http://mantis.innovaphone.com/view.php?id=74991 74991]
|}
|}
'Do Not Disturb' signaling without presence/tuple/contact element.<!---->
callIdentity must be zero when giving ctInitiate to PBX on blind transfer.<!---->
      
      
=== SIP: Support for Mediation Server Cluster ===
=== TLS: Flow control for incoming data ===


{|
{|
Line 5,467: Line 5,329:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69051 69051]
|[http://mantis.innovaphone.com/view.php?id=75004 75004]
|}
|}
Load balancing and fail-over acc. to spec.<!---->
The TLS socket has to wait for the application to process incoming data before sending the next RECV.<!---->
      
      
=== Memory leak with each new telnet session ===
=== H.323: Renegotiating to Fax was rejected, if PCM switch and local networks were used ===


{|
{|
Line 5,478: Line 5,340:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69157 69157]
|[http://mantis.innovaphone.com/view.php?id=75006 75006]
|}
|}
Memory leak with each new telnet session (last packet upon exit not cleaned up)<br/><br/><!---->
If the side which initiated a switch to T.38 has configured PCM and the media address was classified as local due to local network configuration, the T.38 was rejected.<!---->
      
      
=== SIP: Media negotiation problem ===
=== SIP: Interworking of divertingLegInformation1 improved ===


{|
{|
Line 5,489: Line 5,351:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69159 69159]
|[http://mantis.innovaphone.com/view.php?id=75025 75025]
|}
|}
On media-relay with exclusive codec we can answer an incoming SDP offer right away without passing to app.<!---->
Honour subscriptionOption<!---->
      
      
=== PPP connection fails after LCP renegotiation with different authentication methods (PAP -&gt; CHAP) ===
=== IP222,IP232: Changes not saved in some cases ===


{|
{|
Line 5,500: Line 5,362:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69167 69167]
|[http://mantis.innovaphone.com/view.php?id=75088 75088]
|}
|}
the active authentication entity was not stopped when another authentication was started and and signalled layer down after some timeout <!---->
Some config screens did not write changed settings directly after "Save Changes" dialog.<br/>If menu was left with DISC key (instead of ESC key) the changes have been discarded.<br/><br/>Also affects other phones: IP241,IP240,IP230,IP110<br/>Changes are saved immediately when leaving the current screen.<br/>Not when leaving "User Settings" or "Phone Setting" screen.<!---->
      
      
=== PBX Twinning: Blind transfer to other phone in a twin phone configuration was not possible ===
=== PBX Waiting: Diverting leg1 info not correct when diverting to a Waiting Queue ===


{|
{|
Line 5,511: Line 5,373:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69170 69170]
|[http://mantis.innovaphone.com/view.php?id=75091 75091]
|}
|}
Only normal call to other phone or consulation call was supported<!---->
If Waiting queue not defined in root node and escapes are used<!---->
      
      
=== Web-UI: Fixed layout of DynPBX configuration ===
=== PBX: Adjusting received leg1 info wrong ===


{|
{|
Line 5,522: Line 5,384:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69197 69197]
|[http://mantis.innovaphone.com/view.php?id=75097 75097]
|}
|}
Use min-width for fieldset.left<!---->
In context of nodes with escapes<!---->
      
      
=== IPVA didn't run on vSphere5 ===
=== VM: &lt;pbx-upd-obj type="cfu"..&gt; without effect when invoked multiple times ===


{|
{|
Line 5,533: Line 5,395:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69232 69232]
|[http://mantis.innovaphone.com/view.php?id=75121 75121]
|}
|}
IPVA didn't run on vSphere5<!---->
Statement <pbx-upd-obj type="cfu"..> failed to work properly after being used for diversion manipulation multiple times within a single script session.<!---->
      
      
=== SIP: Interoperability of INFO(application/dtmf-relay) with Polycom ===
=== IP28 Watchdog leads to endless interrupt and trap ===


{|
{|
Line 5,544: Line 5,406:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69247 69247]
|[http://mantis.innovaphone.com/view.php?id=75186 75186]
|}
|}
Fail to decode DTMF signal, since "application/dtmf-relay" body does not contain any CRLF.<br/>While CRLF is required according to "SIP INFO Package for DTMF".<br/><!---->
.<!---->
      
      
=== IP222/IP232: residual echo  in handset mode ===
=== IP800/6000: Problem mit FAX und CNG detect: Option to disable CNG detect added ===


{|
{|
Line 5,555: Line 5,417:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69249 69249]
|[http://mantis.innovaphone.com/view.php?id=75278 75278]
|}
|}
Change Codec config <!---->
<br/>config change AC-DSP0 /t38-cng-detect-disable 1 <br/>config write<br/>config activate<br/><br/><!---->
''Status:''
''Status:''
ac_codec3.cpp
ac_dsp3.cpp/h<br/>ac_fax3.cpp/h
      
      
=== ip22/24/28/302/305: DSP affected by certain fax tones causing a trap ===
=== SIP: Decrement Max-Forwards and interwork to H.323 ===


{|
{|
Line 5,568: Line 5,430:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69250 69250]
|[http://mantis.innovaphone.com/view.php?id=75288 75288]
|}
|}
Assert removed for testing<br/><!---->
Instead of sending always "Max-Forwards: 70"<!---->
''Status:''
ac_dsp3.cpp
      
      
=== PBX: Phone config was not sent to phone, if phone was power cycled shorty after registration ===
=== H.323: Renegotiation to T.38 did not work for slowstart call thru media-relay to efc endpoint ===


{|
{|
Line 5,581: Line 5,441:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69280 69280]
|[http://mantis.innovaphone.com/view.php?id=75305 75305]
|}
|}
The new registration after the power cycle was not detected as new registration but as re-transmission of the previous registration, so it was not reported to the PBX and no phone config was sent<!---->
A common scenario when this happend is an XCAPI (slowstart) calling thru a PBX with media-relay configured to an ISDN interface<!---->
      
      
=== CX0-Wave-Encoding sometimes produced white noise ===
=== Fix for MIPS counter ===


{|
{|
Line 5,592: Line 5,452:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69286 69286]
|[http://mantis.innovaphone.com/view.php?id=75310 75310]
|}
|}
CX0-Wave-Encoding sometimes produced white noise.<br/>Also a memory leak was eliminated.<!---->
MIPS counter was incorrect<!---->
      
      
=== IP-DECT: Hanging call after call transfer ===
=== PBX: Allow 'Max Calls' configuration of 0 ===


{|
{|
Line 5,603: Line 5,463:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69309 69309]
|[http://mantis.innovaphone.com/view.php?id=75362 75362]
|}
|}
The hanging call in dectradio is fixed which occurred if a user does an unattended call transfer to an unassigned number and the transferred call is not disconnected.<!---->
So that no call at all is sent to the object but a CFB (if configured) is executed<!---->
      
      
=== SRTP: Bad key exchange on H.323/SIP interworking ===
=== SIP: Send "305 Use Proxy" if INVITE is received from unexpected source ===


{|
{|
Line 5,614: Line 5,474:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69353 69353]
|[http://mantis.innovaphone.com/view.php?id=75380 75380]
|}
|}
MKI was added in SDP due to uninitialized data object.<!---->
Applies to registered interfaces only (e.g. phones).<!---->
      
      
=== Pass DNS-TTL to SIP ===
=== TLS: Possible trap when restoring sessions ===


{|
{|
Line 5,625: Line 5,485:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69374 69374]
|[http://mantis.innovaphone.com/view.php?id=75394 75394]
|}
|}
In order to do load-balancing SIP needs to ge hold on the TTL of DNS resource records.<!---->
Only in v9, depending on the certificate.<!---->
      
      
=== SIP: Support of maddr parameter in redirect response for REGISTER ===
=== PBX Exec Object: Leg2 information send to secretary not adjusted correctly for nodes ===


{|
{|
Line 5,636: Line 5,496:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69386 69386]
|[http://mantis.innovaphone.com/view.php?id=75431 75431]
|}
|}
Set when redirecting REGISTER.<br/>Read when processing redirect response.<!---->
The leg2 information is used to display at the secretary the number of the exec which was called. This number was not correct if nodes with escapes were used<!---->
      
      
=== G726 codec obsolete ===
=== Gateway Interface Maps: Should be applied to leg1 info also ===


{|
{|
Line 5,647: Line 5,507:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69388 69388]
|[http://mantis.innovaphone.com/view.php?id=75437 75437]
|}
|}
The G726 codec was rarely used (if ever) in real life. In addition there are signaling problems specially with DECT peers when G726 is selected. Thus G726 is removed from the list of supported coders in all products.<!---->
The same rules used for a Calling Party Number are applied to leg1 info<!---->
      
      
=== IP222: Input ip addresses does not work ===
=== SoftwarePhone: Timer tick resolution ===


{|
{|
Line 5,658: Line 5,518:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69393 69393]
|[http://mantis.innovaphone.com/view.php?id=75447 75447]
|}
|}
Entering an ip address on the phone's menu does not work.<br/>All digits are doubled.<!---->
The timer tick resolution is increased. This fixes the call RTCP calculation transmitted to the PBX.<!---->
      
      
=== IP222: Call reroute did not work ===
=== IP241,IP222,IP232: Indirect dialing of names not possible ===


{|
{|
Line 5,669: Line 5,529:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69455 69455]
|[http://mantis.innovaphone.com/view.php?id=75451 75451]
|}
|}
Using redial key on in incoming not-connected call did not work.<!---->
Could not dial a name from 'indirect dialing' screen<br/>by moving the focus to input field and going offhook.<br/>(Only fkey worked)<!---->
      
      
=== PBX: Call Diversion/Forward to '-' did not always show expected result ===
=== PBX: CFNR Loop check detected loops that weren't ===


{|
{|
Line 5,680: Line 5,540:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69457 69457]
|[http://mantis.innovaphone.com/view.php?id=75465 75465]
|}
|}
A call diversion to a destination'-' can be used to explicitly no execute a diversion of this type. So if a user has an CFU to '-' and this diversion is valid for a given call (Filter, Boolean), the phone should ring.<br/><br/>In fact the call was rejected.<br/><br/>There was also a problem with CFB in case of "busy on ... calls"<!---->
Esspecially a CFNR at a Waiting Queue was not executed if the destination of the CFNR has transfered the call to the Waiting Queue<!---->
      
      
=== IP-DECT: New radio BMC firmware PCS05Ak ===
=== Gateway: Event "No Media Data Received" was created for each SWITCH-PCM call ===


{|
{|
Line 5,691: Line 5,551:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69468 69468]
|[http://mantis.innovaphone.com/view.php?id=75483 75483]
|}
|}
The new radio BMC firmware PCS05Ak for the IP1200 fixes a trap by the DECT system if more than 255 DECT users without an endpoint subscription are sent to it.<!---->
There should be no event in this case, this is normal.<!---->
      
      
=== PBX: No CLIR on internal calls did not work for SOAP ===
=== PBX: Conference id prefix/suffix configuration ===


{|
{|
Line 5,702: Line 5,562:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69474 69474]
|[http://mantis.innovaphone.com/view.php?id=75507 75507]
|}
|}
If the features "No CLIR on internal Calls" is activated on a PBX a CLI is sent to the called phone even if the call was sent with "CLI presentation restricted". The same should be case on SOAP/TAPI when monitoring this user.<br/><br/>Now when "No CLIR on internal Calls" is enabled all number information available is provided on SOAP.<!---->
The broadcast conference PBX configuration is changed: now, the third party conference unit option is saved and must be enabled to use the configured id prefix and suffix. Otherwise they are ignored now and default values for the innovaphone conference interface of the current device firmware version are used. This fixes the configuration if the firmware is updated from V8 to V9 and the innovaphone conference interface is used.<br/>Disabling the "Create Dynamic Conference Id" option in firmware V9 hotfix 5 and 6 is also fixed now.<!---->
      
      
=== PBX: Reject calls without media, if no known facility ===
=== IP241,IP222,IP232: Call duration display wraps after 100 minutes ===


{|
{|
Line 5,713: Line 5,573:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69477 69477]
|[http://mantis.innovaphone.com/view.php?id=75508 75508]
|}
|}
Fixes compatibility issues between versions. For example presence subscription sessions from v8 phones being forwarded to voicemail<!---->
Now after 60 minutes the display changes from mode [mm:ss] to [hh:mm].<!---->
      
      
=== PBX Waiting: Not possible to send DTMF to Waiting Queue from myPBX Numeric Keyboard ===
=== TLS: Duplicate alert message on malformed ClientHelloV2 ===


{|
{|
Line 5,724: Line 5,584:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69482 69482]
|[http://mantis.innovaphone.com/view.php?id=75509 75509]
|}
|}
The Waiting Queue prohibited the facilities used for this to be sent to calling phone<!---->
Only one alert should be sent per session.<!---->
      
      
=== PBX: Filter for internal or external calls at CFs did not work CFB or CFNR if call already diverted ===
=== TLS: Improved negotiation of protocol version ===


{|
{|
Line 5,735: Line 5,595:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69483 69483]
|[http://mantis.innovaphone.com/view.php?id=75510 75510]
|}
|}
Problem:<br/><br/>User A has CFU to User B<br/>User B has CFNR for ext. Calls only to User C<br/><br/>An internal call to A was diverted to B (ok) and after no response diverted to C (nok)<!---->
TLS server unnecessarily rejected ClientHello messages with TLS 1.1 and higher. Instead of rejecting it should tell the client that it wants to use TLS 1.0.<!---->
      
      
=== PBX Waiting: In case of "Announcemen w/o Connect" together with Alert-Timeout 0, DTMF dialing was not possible ===
=== TLS: Skip empty records ===


{|
{|
Line 5,746: Line 5,606:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69496 69496]
|[http://mantis.innovaphone.com/view.php?id=75511 75511]
|}
|}
In this case as only response to the incoming SETUP a PROGRESS was sent. This meant, that the caller was still in overlap dialing state, so a phone does not send DTMF, but translates input keys to INFO dialing messages.<br/><br/>A CALL-PROC is now sent before PROGRESS, which terminates the dialing.<!---->
TLS record layer should ignore records with zero length without doing anything.<!---->
      
      
=== SIP: Send BYE with Reason header with "Q.850 Recovery on timer expiry" ===
=== myPBX: Script error after waking the PC up from hibernation ===


{|
{|
Line 5,757: Line 5,617:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69500 69500]
|[http://mantis.innovaphone.com/view.php?id=75532 75532]
|}
|}
If session refresh is outstanding the call is released with BYE with "Reason: Q.850;cause=16".<br/>Better send BYE with "Reason: Q.850;cause=102"<!---->
Sometimes window.open throws an unhandled exception in this case.<!---->
      
      
=== PBX Waiting: No ringback when doing two-stage dialing to a Gateway/Trunk object ===
=== SIP: Check proxy availability did not work in any scenario ===


{|
{|
Line 5,768: Line 5,628:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69531 69531]
|[http://mantis.innovaphone.com/view.php?id=75542 75542]
|}
|}
A local ringback is now switched on, when receiving ALERT from called party<!---->
OPTIONS can be used to poll remote proxy's availablity<br/>to avoid TCP timeout when INVITE is to be sent.<br/>Signaling interface is marked as down and not used anymore.<!---->
      
      
=== SIP: Bug when decoding SRTP keys from SDP ===
=== IP232: Hiding touch keyboard by touching a control ===


{|
{|
Line 5,779: Line 5,639:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69545 69545]
|[http://mantis.innovaphone.com/view.php?id=75576 75576]
|}
|}
Bug in decode base64.<!---->
Touching a text control activates the touch keyboard.<br/>Touching the text control again now deactivates the touch keyboard.<!---->
      
      
=== Web-UI: Input field to small for SIP-URI ===
=== IP241,IP222,IP232: CFU information in header bar ===


{|
{|
Line 5,790: Line 5,650:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69578 69578]
|[http://mantis.innovaphone.com/view.php?id=75654 75654]
|}
|}
Input control on "UC" tab of External UC properties dialog was too small.<br/>Only part of SIP-URI was visible.<!---->
CFU information in header bar is now displayed even if there's not is enough space between name and number.<br/>Either name or number is omitted is required.<!---->
      
      
=== phone: assume an outbound call to be an external call if connected number info is missing in connect event ===
=== IP241,IP222,IP232: Replace triangle by arrow to display diversion/transfer information ===


{|
{|
Line 5,801: Line 5,661:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69581 69581]
|[http://mantis.innovaphone.com/view.php?id=75689 75689]
|}
|}
In certain ISDN configurations the PBX can not provide the connected number info in the connect event for an outbound call. In this case the the call was assumed to be an internal call and consequently was not recorded when transparent recording of external calls was configured.<br/>Now an external call is assumed in this case.<br/> <br/><!---->
Replace quite heavy 'BLACK RIGHT-POINTING POINTER'<br/>by much lighter 'RIGHTWARDS ARROW'<br/>to display diversion/transfer information<br/>on call control, fkeys and call lists.<!---->
      
      
=== IP-DECT: Reset link ===
=== H.323: A forwarded HopCount&gt;32 could result in a very small HopCount ===


{|
{|
Line 5,812: Line 5,672:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69584 69584]
|[http://mantis.innovaphone.com/view.php?id=75765 75765]
|}
|}
The reset link of DECT System Config GUI page is fixed.<!---->
There are only 5 bits for transmitting a HopCount in H.323. A HopCount from SIP is typically 70 and this value was not reduced to 32 but only the 5 lower bits were transmitted, which resulted in a HopCount of 6<!---->
      
      
=== SIP: Do not check remote certificate name when calling PBX client with TLS ===
=== IP: Minor memory management change ===


{|
{|
Line 5,823: Line 5,683:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69598 69598]
|[http://mantis.innovaphone.com/view.php?id=75770 75770]
|}
|}
Check is reqired only when calling an unregistered client or when giving INVITE to registrar.<!---->
no use of malloc in ipproc<!---->
      
      
=== phone: VLAN signaling priority could not be configured via phone menu ===
=== IP241,IP222,IP232: Two waiting calls are now displayed (instead of one) ===


{|
{|
Line 5,834: Line 5,694:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69633 69633]
|[http://mantis.innovaphone.com/view.php?id=75789 75789]
|}
|}
Under "Menu/Administration/IP Settings/VLAN" there was only a "VLAN Priority" menu item. This menu item did override the 'Priority RTP Data' value but not the 'Priority Signaling' value as entered via WEB configuration.<br/>Now the items "Prio. RTP Data" and "Prio. Signaling" replace the "VLAN Priority" item.<!---->
Waiting calls are displayed below the two main calls.<br/>Only two lines of information and grey background.<!---->
      
      
=== IPxx10-sata: trap after config /trace /track activation ===
=== SIP: Handling of some more proprietary Alert-Info and Call-Info in INVITE ===


{|
{|
Line 5,845: Line 5,705:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69642 69642]
|[http://mantis.innovaphone.com/view.php?id=75824 75824]
|}
|}
Instruccion was accessing uninitialized pointer.<!---->
Alert-Info: <Bellcore-dr3>;info=alert-recall<br/>and<br/>Alert-Info: <http://not_used.com>;info=alert-autoanswer<br/>and<br/>Call-Info: <sip:127.0.0.1>;answer-after=0<!---->
      
      
=== IP222: Alpha input using the num block ===
=== PBX Waiting: Leg2 information sent with calls to operators not adjusted correctly for nodes ===


{|
{|
Line 5,856: Line 5,716:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69646 69646]
|[http://mantis.innovaphone.com/view.php?id=75830 75830]
|}
|}
Input mode changes back to numeric in screen "Wahlvorbereitung".<br/>Should stay in alpha mode.<!---->
If a Waiting Queue was configured in a Node not the root node, the leg2 info was not adjusted corrcectly. The leg2 information is used to signal to the operator which Waiting Queue is forwarding the call<!---->
      
      
=== IP200: Manufacturing test fails is hwbuild is reprogrammed ===
=== SIP: Decoding of RFC-4412 definitions ===


{|
{|
Line 5,867: Line 5,727:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69662 69662]
|[http://mantis.innovaphone.com/view.php?id=75848 75848]
|}
|}
Old hwbuild was not cleared, because on INCA devices the hwbuild is located on a higher address was not cleared.<!---->
The 'Resource-Priority' Header Field<br/>The 'Accept-Resource-Priority' Header Field<br/>The 'resource-priority' Option Tag<br/>417 Unknown Resource-Priority response<!---->
''Status:''
flash_firmware.cpp
      
      
=== H.323: Media Negotiation problem with Lync interop and SRTP ===
=== SIP: Handling of 422 Session Interval Too Small ===


{|
{|
Line 5,880: Line 5,738:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69687 69687]
|[http://mantis.innovaphone.com/view.php?id=75873 75873]
|}
|}
With a retrieve from the lync after hold (which is signaled as a reinvite with sendrevc) new media parameters were sent, containing new SRTP keys. These new media parameters cannot be used, on the PBX which is initiating new end to end media negotiation at the same time. These media parameter were not ignored properly.<!---->
Get Min-SE and re-try INVITE<!---->
      
      
=== phone_orchid: wrong volume setting when monitor mode is entered ===
=== SIP: Send b=TIAS attribute in media description ===


{|
{|
Line 5,891: Line 5,749:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69734 69734]
|[http://mantis.innovaphone.com/view.php?id=75881 75881]
|}
|}
when monitor mode ise entered by pressing the speaker key in a handset conversation the handsfree speaker is enabled in addition to the handset speaker.<br/>the volume was reconfigured with the wrong value.<!---->
Send b=TIAS attribute in media description<!---->
      
      
=== phone_orchid: Calls received with CLIR appear in call list with an empty entry ===
=== remove indirect calls to os_mem_alloc() via malloc() ===


{|
{|
Line 5,902: Line 5,760:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69737 69737]
|[http://mantis.innovaphone.com/view.php?id=75886 75886]
|}
|}
Calls received with CLIR or without a number/name appeared in call list with an empty entry; now either "anonymous" (CLIR) or "unknown" is displayed instead of a name<br/><br/><!---->
malloc() shall not be used anymore in sources dedicated to run on innovaphone hardware<!---->
      
      
=== SIP: Take SDP id and version as 64bit integer ===
=== IPv6: Memory leak if packets received, which are not handled locally ===


{|
{|
Line 5,913: Line 5,771:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69738 69738]
|[http://mantis.innovaphone.com/view.php?id=75920 75920]
|}
|}
o-line of SDP offer/answer is defined as 64bit integer<br/>Interoperability with MX-ONE<!---->
E.g. an IPv6 multicast UDP packet, for some other application was not deleted.<!---->
      
      
=== Relay: Forward facilities to local destinations ===
=== SIP: Fix for video negotiation ===


{|
{|
Line 5,924: Line 5,782:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69764 69764]
|[http://mantis.innovaphone.com/view.php?id=75954 75954]
|}
|}
This fix is related to the previous fix #66629 for V9 hotfix2. Now, facilities are only forwarded, if the destination is a physical interface, not e.g. a SIP provider.<!---->
Fix for video negotiation<!---->
      
      
=== SIP: Trap when handling NOTIFY(application/qsig) ===
=== phone_orchid: after leaving a conference the active call was mute sometimes ===


{|
{|
Line 5,935: Line 5,793:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69771 69771]
|[http://mantis.innovaphone.com/view.php?id=75955 75955]
|}
|}
Traps if no progress indicator present in tunneled DISCONNECT message.<!---->
The conference was not stopped internally in all cases. <!---->
      
      
=== WEB-UI Config Upload screen blank after upload of a file with a lot of failing lines ===
=== Flashman: Show meaningful result after OEM certificate upload in production ===


{|
{|
Line 5,946: Line 5,804:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69812 69812]
|[http://mantis.innovaphone.com/view.php?id=75958 75958]
|}
|}
When uploading a file with about 50 failing lines the screen was left blank without any info about the failing operation.<!---->
"Certificate" instead of "-unknown-"<!---->
      
      
=== phone: "Keep Calling Party Info on Pickup Key" option from Phone-&gt;Preferences did not work with very long numbers/names ===
=== SIP: Fix for Supported header ===


{|
{|
Line 5,957: Line 5,815:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69816 69816]
|[http://mantis.innovaphone.com/view.php?id=76016 76016]
|}
|}
speciall when using this option in conjunction with the "Display Name on Pickup/Partner Key" option the calling party info was not correctly displayed<!---->
Re-implemented to easily add more tags<!---->
      
      
=== IP6010: SRTP using AES-192 and AES-256 did not work ===
=== IP222,IP232: Alpha mode did not work on multi-line edit controls ===


{|
{|
Line 5,968: Line 5,826:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69828 69828]
|[http://mantis.innovaphone.com/view.php?id=76165 76165]
|}
|}
Due to a bug in the encryption driver of the IP6010, only AES-128 worked on this platform.<!---->
Could no enter alpha chars using the num block<!---->
      
      
=== IP-DECT: Location update (OEM) ===
=== SIP: Trap when parsing very large History-Info header ===


{|
{|
Line 5,979: Line 5,837:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69863 69863]
|[http://mantis.innovaphone.com/view.php?id=76198 76198]
|}
|}
Location update with message waiting information is fixed if the endpoint roams. This is only used by OEM devices.<!---->
Trap when parsing very large History-Info header<!---->
      
      
=== Logging: Threshold for error event "SRTP authentication failed" ===
=== SIP: SIP message may exceed available buffer ===


{|
{|
Line 5,990: Line 5,848:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69920 69920]
|[http://mantis.innovaphone.com/view.php?id=76203 76203]
|}
|}
Error event was triggered at the very first decrypt failure.<br/>Some decrypt failure are expected during media re-negotiation.<br/>Trigger this error event after 10 decrypt failures in line.<br/><!---->
Trace message like this can occur:<br/>  ERROR: SIP message buffer (1025) exceeded! (9,14,801d781c)<br/> SIP: CSeq missing<br/> SIP: Message encoding failed!<br/><!---->
      
      
=== PBX Calls Page/SOAP wrong number ===
=== SIP: Contact not allowed in BYE/CANCEL/PRACK request ===


{|
{|
Line 6,001: Line 5,859:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69921 69921]
|[http://mantis.innovaphone.com/view.php?id=76210 76210]
|}
|}
In a configuration with escapes for calls from a slave and a node not the root node and the call forwarded to the master, because the number could not be resolved locally, wrong escapes were added to the called number<!---->
Acc. to RFC-3261 there must not be Contact header line in BYE/CANCEL request<br/>Acc. to RFC-3262 there must not be Contact header line in PRACK request<!---->
      
      
=== phone ip222: MWI LED not working ===
=== reset could fail, if issued during firmware update ===


{|
{|
Line 6,012: Line 5,870:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69924 69924]
|[http://mantis.innovaphone.com/view.php?id=76266 76266]
|}
|}
The MWI LED was not switched on for pending voice mails and/or pickups<!---->
and even worse any following reset command were not accepted as well<!---->
      
      
=== Trap during fax transfer ===
=== IP241,IP222,IP232: Show "New message from ..." on active phone screen ===


{|
{|
Line 6,023: Line 5,881:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=69950 69950]
|[http://mantis.innovaphone.com/view.php?id=76283 76283]
|}
|}
Seen on Ip28, can happen on AC_DSP3 ( IP22/24/28/302/305/1060/301/6010 )<!---->
Show "New message from ..." on phone screen for 3 seconds when active with calls (non-idle).<br/>Like on b/w phones.<!---->
''Status:''
ac_dsp3.h
      
      
=== phone_orchid: checkmark "Phone/User-x/General/Options/No DTMF Detection" has no effect ===
=== IP241,IP222,IP232: Toggle of display-name/name-alias/number was buggy ===


{|
{|
Line 6,036: Line 5,892:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70114 70114]
|[http://mantis.innovaphone.com/view.php?id=76290 76290]
|}
|}
If this checkmark is set DTMF digits entered via keyboard in a connected call shall be sent in-band as voice data, not encoded in RTP-DTMF packets as usual.<br/> <!---->
String termination was missing.<br/>Garbage data was displayed.<!---->
      
      
= V9 Hotfix 5 (90600.05) =
=== IP232: Open call details when touching entry in call list ===
Changes included in Version 9 hotfix5
[http://mantis.innovaphone.com/view.php?id=70208 Definition]
 
== New Features ==
 
 
   
=== IP-DECT: Default frame size to 30ms ===


{|
{|
Line 6,055: Line 5,903:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70140 70140]
|[http://mantis.innovaphone.com/view.php?id=76292 76292]
|}
|}
Now the default frame size is 30ms.<!---->
Instead of initiating call to remote party in call list entry.<!---->
      
      
=== SIP: Support for transparent message headers on transfer ===
=== H.323: Merges from v10 for Media Negotiation fixes and special OEM protocol features ===


{|
{|
Line 6,066: Line 5,914:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70163 70163]
|[http://mantis.innovaphone.com/view.php?id=76301 76301]
|}
|}
Interoperability with Telepo.<br/><!---->
- Problem with transparent forwarding of SDP<br/>- special SRTP key exchange<br/>- Video fixes<br/>- Unit Testing support<!---->
      
      
=== added support for USB tracing with Wireshark ===
=== IP-DECT: Memory leak with rejected radio registration ===


{|
{|
Line 6,077: Line 5,925:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70373 70373]
|[http://mantis.innovaphone.com/view.php?id=76308 76308]
|}
|}
USB traces are now supported by the debug class and can be opened with Wireshark and a new innovaphone.dll.<!---->
Memory leak occurred with a rejected radio registration is fixed, now.<!---->
      
      
=== SIP: Prefer P-Asserted-Identity with tel-URI ===
=== PBX-SOAP: When monitoring Waiting queue, the peer number was not indicated ===


{|
{|
Line 6,088: Line 5,936:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70417 70417]
|[http://mantis.innovaphone.com/view.php?id=76315 76315]
|}
|}
Prefer P-Asserted-Identity with tel-URI over that with sip-URI.<br/>In case there are two P-Asserted-Identity headers in INVITE.<!---->
The number was considered presentation restricted, because of some bit manipulation error<!---->
      
      
=== Gateway: Forward Display Info received from ISDN Setup to H.323 ===
=== ip22/24/28/302/305/6010/3010/1060: Fax failed if tones are sent during the fax call ===


{|
{|
Line 6,099: Line 5,947:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70562 70562]
|[http://mantis.innovaphone.com/view.php?id=76342 76342]
|}
|}
needed for compatibility with SecuGATE LI30<!---->
In some cases the Fax messages were missing due to tones send during the fxx call<!---->
''Status:''
ac_dsp3.cpp
      
      
=== &lt;pbx-getcallinfo&gt; returning diversion reason ===
=== Gatway: Transfer handling may cause re-routing after regular hang-up ===


{|
{|
Line 6,110: Line 5,960:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70697 70697]
|[http://mantis.innovaphone.com/view.php?id=76357 76357]
|}
|}
as<br/><pbx-getcallinfo out-leg2-reason="..."/><!---->
Re-routing is expected to take place on non-connected calls only.<!---->
      
      
=== myPBX: Interface for IM provider ===
=== PBX Number Map: Not possible to use overlap dialing to Number map with incomplete destination ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70867 70867]
|[http://mantis.innovaphone.com/view.php?id=76377 76377]
|}
|}
Interface between the JavaScript client and the myPBX launcher.<!---->
If a Number Object with incomplete destination was called and the number was to be completed with overlap dialing a wrong number was called.<br/><br/>This is a usefull feature to use Number Maps as quick dial to other nodes. In this case Number Maps are used with a destination of the remote node, so the number is incomplete, the number within this node has to be dialed in addition to the Number of the Number Map object.<!---->
''Status:''
IM to JS:<br/>prepare_dial_name(value)<br/>prepare_im(value)<br/>ep_request(name,number)<br/><br/>JS to IM:<br/>innovaphone_updateXml(xml)<br/>innovaphone_sessionInfo(domain,name,number,cn,dn)
      
      
=== SIP: Diagnostics of transport error "SIP Overload" ===
=== SIP: Support for media recording ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70883 70883]
|[http://mantis.innovaphone.com/view.php?id=76396 76396]
|}
|}
Limitation of buffer allocation<!---->
Media recording to HTTP URL<!---->
''Status:''
siptrans.cpp/h<br/>sipmsg.h
      
      
=== IP-DECT: License restriction for OEM device ===
=== Gateway: Conference interface, no voice ===


{|
{|
Line 6,147: Line 5,993:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70887 70887]
|[http://mantis.innovaphone.com/view.php?id=76419 76419]
|}
|}
A license restriction for an OEM device is added. It is not used in the IP1200.<!---->
The ADSP firmware is changed to version 122. This fixes a bug in the conference interface of IP6000/IP6010/... which results in conference calls without voice in one direction for a single member.<!---->
      
      
=== SIP: Diagnostics of error "Registration expired" ===
=== phone_orchid: spurious trap in long conference calls ===


{|
{|
Line 6,158: Line 6,004:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70938 70938]
|[http://mantis.innovaphone.com/view.php?id=76445 76445]
|}
|}
Missing registration refresh<!---->
in long confernce calls the phone may trap because the DSP delivers an unexpected zero size packet.<!---->
''Status:''
sip.cpp/h<br/>
      
      
=== Possibility to clear the call list in mypbx ===
=== IP241,IP222,IP232: Call held indication ===


{|
{|
Line 6,171: Line 6,015:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71202 71202]
|[http://mantis.innovaphone.com/view.php?id=76454 76454]
|}
|}
A new button is now available in mypbx to clear the call list.<!---->
If remote side has put the call on hold<br/>the user should get a visible indication<br/>(in addition to hearing music on hold).<br/><br/>New call status "held" is displayed.<!---->
      
      
=== phone: LED mode of Join Group function key can be set both for idle and for active state ===
=== LDAP-Expert, Edit Object Dialog: Could show more than one object ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71247 71247]
|[http://mantis.innovaphone.com/view.php?id=76464 76464]
|}
|}
sometimes the "not in group" state must be signaled as the exception<!---->
LDAP-Expert, Edit Object Dialog: Could show more than one object<!---->
      
      
=== myPBX: Show implicit visibility in groups ===
=== Qsig: Connected Number could be encoded wrong ===


{|
{|
Line 6,193: Line 6,037:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71250 71250]
|[http://mantis.innovaphone.com/view.php?id=76472 76472]
|}
|}
Active members of groups can see the presence and the calls of other group members. In order to make that clear to the user, now the visibility settings of myPBX show in what groups the user is visible.<!---->
Qsig: Connected Number could be encoded wrong<!---->
      
      
=== IP-DECT: CSS changes for new OEM device ===
=== Gateway: Better handling of call-reroute requests ===


{|
{|
Line 6,204: Line 6,048:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71282 71282]
|[http://mantis.innovaphone.com/view.php?id=76515 76515]
|}
|}
For a new OEM device with a changed style some CSS classes are added.<!---->
Apply interface maps to numbers in reroute request<!---->
      
      
=== PBX: New User property 'Do not Disturb' ===
=== PBX-SOAP: Better error handling when using a Waiting Queue for outgoing calls ===


{|
{|
Line 6,215: Line 6,059:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71439 71439]
|[http://mantis.innovaphone.com/view.php?id=76544 76544]
|}
|}
No calls are sent to the user if set.<br/><br/>Can be set by 'External UC' applications (e.g. Estos Procall)<!---->
- call was hanging if an invalid number or user without registration was called<br/>- trap if outgoing call was done on operator connect<!---->
      
      
=== PBX: Support up to six devices for a user ===
=== Spurious leaks in leak checking ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71506 71506]
|[http://mantis.innovaphone.com/view.php?id=76589 76589]
|}
|}
Sometimes 4 devices are to little.<br/>A little bit of cleanup on the user interface was done as well with this change<!---->
A problem which mostly shows up in automated release testing and then requires manual investigation.<!---->
      
      
=== phone: Mic Off/On controllable via Soap:UserRc(&lt;call&gt;,14/15) ===
=== SoftwarePhone: Auto start configuration ===


{|
{|
Line 6,237: Line 6,081:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71721 71721]
|[http://mantis.innovaphone.com/view.php?id=76595 76595]
|}
|}
To allow Soap app's control of the mute key<!---->
The duplicate back slash occurred with the install directory is fixed now. Used by the run mode configuration 'At login'.<!---->
      
      
=== SoftwarePhone: New Tray Icon ===
=== myPBX launcher: Window position was not always remembered ===


{|
{|
Line 6,248: Line 6,092:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71969 71969]
|[http://mantis.innovaphone.com/view.php?id=76626 76626]
|}
|}
The SoftwarePhone has got a new tray icon differs from the myPbx one, now.<!---->
When the user just moved the window without resizing it, the window position was not remembered for the next session.<!---->
      
      
== Bug Fixes ==
= V9 Hotfix 9 (90600.11, withdrawn) =
Changes included in Version 9 hotfix9
[http://mantis.innovaphone.com/view.php?id=76991 Definition]
 
== New Features ==




      
      
=== SIP: Message decoding error ===
=== PBX-SOAP: UserCall new option diversion override ===


{|
{|
Line 6,263: Line 6,111:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70116 70116]
|[http://mantis.innovaphone.com/view.php?id=75773 75773]
|}
|}
Only concerns Message headers whose value starts and with quotes, but are not quoted.<br/>E.g.<br/> Referred-By: "Huvudnummer"<sip:400@abcdef.ghi;fnrid=1759>;from-tag=5decdf1a;to-tag=2515833546;org-cid="6afa95ede909d311906f00013e11cdb3@192.168.2.115"<br/><!---->
To initiate calls, which are not diverted by a configured CFU.<br/><br/>Use Info[] element with type="fty-no-cf"<!---->
      
      
=== Linux: VLAN GUI removed ===
=== SIP: Support for "Allow-Events: ccnr" and "Allow-Events: ccbs" ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70131 70131]
|[http://mantis.innovaphone.com/view.php?id=76658 76658]
|}
|}
It is not necessary to configure a VLAN to Linux. Now the GUI page Linux VLAN is removed.<!---->
For interoperability:<br/>Support for "Allow-Events: ccnr" and "Allow-Events: ccbs"<!---->
      
      
=== ip24/ ip6010/phone_orchid: in-band ring back tone generation into voice channel did not work ===
=== Phone-&gt;PBX LDAP Search returns normalised number to be dialled by phone ===


{|
{|
Line 6,285: Line 6,133:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70146 70146]
|[http://mantis.innovaphone.com/view.php?id=77261 77261]
|}
|}
sometimes a ring back tone must be sent in-band to an ISDN channel. the tone was generated but passed to the wrong destination. <!---->
1)Phone->PBX LDAP Search returns normalised number to be dialled by phone.<br/>2)Phone receives info about escape digits when registering at its PBX.<br/><br/>With 1) the PBX includes an object's normalized number into the LDAP search result.<br/>With 1) the phone is able to dial that normalized number.<br/>With 2) the phone is able to prefix required escape digits to the received normalized number.<!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
=== SIP: Problems with CLEARMODE ===
=== PBX soap method SetPresence ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70149 70149]
|[http://mantis.innovaphone.com/view.php?id=78092 78092]
|}
|}
CLEARMODE was not offered as prefered codec outgoing INVITE.<br/>Destination accepted G711a instead of CLEARMODE:<br/><br/>Offer:<br/>  v=0<br/>  o=- 18 1 IN IP4 172.16.66.77<br/>  s=-<br/>  c=IN IP4 172.16.66.77<br/>  t=0 0<br/>  m=audio 16422 RTP/AVP 4 18 8 0 96 97 101 13<br/>  a=rtpmap:96 G726-32/8000<br/>  a=rtpmap:97 CLEARMODE/8000<br/>  a=rtpmap:101 telephone-event/8000<br/>  a=fmtp:18 annexa=yes<br/>  a=fmtp:18 annexb=no<br/>  a=fmtp:101 0-15<br/>  a=ptime:20<br/><br/>Answer:<br/>  v=0<br/>  o=cp10 131833822144 131833822144 IN IP4 172.16.66.175<br/>  s=SIP Call<br/>  c=IN IP4 172.16.67.132<br/>  t=0 0<br/>  m=audio 33526 RTP/AVP 8 0 97<br/>  b=AS:75<br/>  a=rtpmap:8 PCMA/8000/1<br/>  a=rtpmap:0 PCMU/8000/1<br/>  a=rtpmap:97 CLEARMODE/8000/1<br/>  a=ptime:30<br/><!---->
PBX soap method SetPresence added to set the presence of a PBX user.<!---->
      
      
=== IP-DECT: Master radio list sync highlighting ===
=== phone ip222, ip232 : USB headsets need not to be configured anymore ===


{|
{|
Line 6,307: Line 6,157:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70150 70150]
|[http://mantis.innovaphone.com/view.php?id=78114 78114]
|}
|}
In the IP-DECT Master radio list the active sync highlighting is fixed, now.<!---->
If an USB headset with a known signature (vendor/product id) is plugged it is automatically enabled. This is indicated by the headset symbol in the status line.<br/>"Phone/Preferences/Start Outbound Call on Electronic Hook Switch (EHS) Signal" is implied in this case because some headsets will loose state if a hoook signal is ignored. <br/><!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
=== SIP: No update of name presentation after transfer on QSIG interworking ===
=== phone: ip222, ip232: USB headset support - Plantronics C420,  GN2000 USB - MS OC Version ===


{|
{|
Line 6,318: Line 6,170:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70155 70155]
|[http://mantis.innovaphone.com/view.php?id=78201 78201]
|}
|}
Fix for QSIG/SIP interworking.<br/>Display name of callTransferComplete was not passed to SIP.<!---->
.<!---->
''Status:''
checked in to 9.00, 90600
      
      
=== IP222/IP232: Handset  volume with 90772 andHF4 too low ===
=== PBX Mobility: No-Alert checkmark as workaround for provider which do not provide Alerting ===


{|
{|
Line 6,329: Line 6,183:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70226 70226]
|[http://mantis.innovaphone.com/view.php?id=78410 78410]
|}
|}
This is a side effect of the echo canceller setting, volume in IP222/232 increased. IP241 uses old settings.<!---->
Some SIP provider do not provide an Alerting signal when a mobile phone is called. This could result in no ringback signal to the caller or the min/max-alert feature not working.<br/><br/>This new checkmark provides a fake Alerting in case Progress is received<!---->
      
      
=== SIP: Cannot use SIP phone with myPBX ===
=== show linux shutdown warning on firmware reset page ===


{|
{|
Line 6,340: Line 6,194:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70252 70252]
|[http://mantis.innovaphone.com/view.php?id=78665 78665]
|}
|}
Blind transfer (sending REFER) did not work since v9hotfix3.<!---->
If linux is running, a warning is now shown, that linux should be shutdowned before a firmware reset is performed.<br/><!---->
      
      
=== IP241: Use menu-key to enter tftp mode ===
=== Linux: Enable/Disable support link ===


{|
{|
Line 6,351: Line 6,205:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70263 70263]
|[http://mantis.innovaphone.com/view.php?id=78786 78786]
|}
|}
tftp was activated with alt-key, on on IP240 menu-key is used<!---->
Now the Linux menu is always shown and a link is provided to enable or disable the Linux support (RAM reservation). The support state is also saved in the downloaded configuration file and restored with the upload.<br/>Update: The support state is only saved in the downloaded configuration with password. The state is not saved in the configuration file with standard password or if downloaded by the update server. Please use the next or a later hotfix instead, see also fix #78836.<!---->
''Status:''
   
platform_orchid.c<br/>boot241.y
== Bug Fixes ==
 
 
      
      
=== IP222/IP232: Sporadic Problem with the LCD Display after soft-reset ===
=== H.323: Media Negotiation problem with conferences on IP-DECT ===


{|
{|
Line 6,364: Line 6,220:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70269 70269]
|[http://mantis.innovaphone.com/view.php?id=76314 76314]
|}
|}
It seem to be problematic to reset all orchid modules, e.g. the DMA module during software reset.<br/>Now only USB and ENET modules are reset, the display gets also reset.<br/>The display reset is released in the firmware.<!---->
A channel was not switched to the conference after a hold/retrieve cycle<!---->
''Status:''
start_orchid.S<br/>platform_orchid.c<br/>phone_orchid.cpp<br/>boot222.y<br/>boot232.y<br/>boot241.y
      
      
=== IP241: Missing fkey icons ===
=== PBX Mobility: Dialed digits could get lost, when using Opticaller ===


{|
{|
Line 6,377: Line 6,231:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70353 70353]
|[http://mantis.innovaphone.com/view.php?id=76598 76598]
|}
|}
Icons types list, face, mask-white and mask-black are available now.<!---->
There could be a collision of a dialed digit with media renegotiation. For example if with the first digit a media was switched to inband information from a carrier.<!---->
      
      
=== PBX: Conference trap ===
=== phone: Ring Tone Titles containing apostrophes  garble phone configuration ===


{|
{|
Line 6,388: Line 6,242:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70362 70362]
|[http://mantis.innovaphone.com/view.php?id=76745 76745]
|}
|}
Some reworks of the PBX conference object. Fixes traps with call transfers of conference calls and conference calls to other PBX objects or mobility. Object update is also possible without call and chat clearing, now. Set maximum call number takes effect for maximum incoming calls, now.<!---->
When under "Phone/Ring Tones/Add Ring Tone" a title containing apostrophes is entered the page "Phone/User-x/Preferences" cannot be edited anymore because of a XML-Error.<!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
=== IP241: Status icon for conditional call diversion ===
=== SIP: Record-Route handling on outbound subscriptions ===


{|
{|
Line 6,399: Line 6,255:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70363 70363]
|[http://mantis.innovaphone.com/view.php?id=76831 76831]
|}
|}
Now having 2 different status icons for diversion.<br/>One for unconditional and one for conditional forwarding.<!---->
Processing of Record-Route in SUBSCRIBE response.<br/>Simply missing.<!---->
      
      
=== IP241: Moving focus on screen "Call Diversion" activates diversion ===
=== phone: Pickup fuction key is not displayed while all alerting calls are displayed on a Partner key ===


{|
{|
Line 6,410: Line 6,266:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70377 70377]
|[http://mantis.innovaphone.com/view.php?id=77393 77393]
|}
|}
Using UP/DOWN/LEFT/RIGHT keys on the "Call Diversion" screen makes ON/OFF changing it state.<!---->
alerting calls displayed on a Partner key are not displayed on the Pickup key.<br/>if nothing has to be displayed on the pickup key the key should not disappear but display the 'idle' label<!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
=== Webmedia: Recording of G.722 did not work ===
=== H.323: Media Negotiation problem with transfer in Gateway (not PBX) ===


{|
{|
Line 6,421: Line 6,279:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70382 70382]
|[http://mantis.innovaphone.com/view.php?id=77683 77683]
|}
|}
Empty file was created.<!---->
Under special conditions a blind transfer happend in the Gateway could result in a call without media. This only happened if the call was transfered twice and the destination of the first transfer was a physical interface.<!---->
      
      
=== myPBX: Closing a chat window sometimes causes a Java Script error ===
=== PBX Twinning: When calling another (twin) phone, the call was sent to the original phone also ===


{|
{|
Line 6,432: Line 6,290:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70439 70439]
|[http://mantis.innovaphone.com/view.php?id=77829 77829]
|}
|}
client.js, line 2033<br/><br/>The Problem is a collision of closing the window and incoming messages for that window.<!---->
It still works, but this waiting call could be confusing<!---->
      
      
=== SIP: DNS resolving _stun._udp.xten.com did not work ===
=== PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX ===


{|
{|
Line 6,443: Line 6,301:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70449 70449]
|[http://mantis.innovaphone.com/view.php?id=77874 77874]
|}
|}
Because of unusual DNS response.<br/>Target attribute of SRV answer records did not contain host domain name, but ip address.<br/><br/><result val="0" title="SUCCESS"><br/><br/><answer_rrs title="Answer Records"><br/><rr rr-type="33" rr-name="SRV" name="_stun._udp.xten.com" ttl="3407" priority="10" weight="0" port="3478" target="216.93.246.16"/><br/><rr rr-type="33" rr-name="SRV" name="_stun._udp.xten.com" ttl="3407" priority="10" weight="0" port="3478" target="216.93.246.14"/><br/></answer_rrs><br/><br/><ns_rrs title="Authoritative Nameserver Records"><br/></ns_rrs><br/><br/><ar_rrs title="Additional Records"><br/></ar_rrs><br/><br/></result><br/><!---->
A call from a object within a node with escapes on a slave PBX was not routed to the master if the destination was within the same node and not known on this slave but was sent to the node-extern destination directly<!---->
      
      
=== H.323: Renegotiation to Fax did not work under some conditions ===
=== PBX Waiting: Name Id missing in calls initiated with SOAP ===


{|
{|
Line 6,454: Line 6,312:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70462 70462]
|[http://mantis.innovaphone.com/view.php?id=77911 77911]
|}
|}
Problem happened<br/>- If switch to fax was done right after connect. This is typically done by IP Fax Servers<br/>- If multiple signaling hops (e.g. multiple PBXs) were used<br/>- If connect to a tone interface happened during dialing<!---->
When a Waiting Queue is used by applications to initiate outgoing calls, the name of the waiting queue should be sent with these calls as calling name. This name id was missing<!---->
      
      
=== myPBX: Norwegian translation updated ===
=== RTP: Potential random trap when closing channels ===


{|
{|
Line 6,465: Line 6,323:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70477 70477]
|[http://mantis.innovaphone.com/view.php?id=77918 77918]
|}
|}
Sent => Send<br/>Logginn => Pålogging <br/>Permanent logginn => Forbli pålogget<!---->
Happens if there is a collision with a received packet and closing of the channel. Window for this is very small, so it should happen very rarely. Probability can increase with high load.<!---->
      
      
=== ethernet broadcasts bearing unicast IP packets with an arbitrary destination address were passed to the local IP-stack ===
=== Web-UI: Visually separated commands ===


{|
{|
Line 6,476: Line 6,334:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70503 70503]
|[http://mantis.innovaphone.com/view.php?id=77954 77954]
|}
|}
some load-balancing implementations send unicast IP packets (specially TCP-SYN) as ethernet broadcast packets.<br/>Such packets must be silently discarded if the IP destination address is not the address of one of the local interfaces.<br/><!---->
Visually separated commands 'Clear' and 'Save' on Maintenance/diagnostics/Events.<br/>Also 'download' and 'delete' on General/License.<!---->
      
      
=== IP222: Cannot leave menu screen with ESC when IP address ctrl was active ===
=== IP241,IP222,IP232: Show special symbol for call completion entries in call list ===


{|
{|
Line 6,487: Line 6,345:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70569 70569]
|[http://mantis.innovaphone.com/view.php?id=77966 77966]
|}
|}
ESC was taken as BS (backspace).<br/>ESC must be handled as ESC when BS is no longer possible (empty input).<!---->
Show special symbol for call completion entries in call list to separate from usual missed calls.<!---->
      
      
=== PBX: Presence update for PBX objects ===
=== IP241,IP222,IP232: Status message "Recording" not displayed ===


{|
{|
Line 6,498: Line 6,356:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70577 70577]
|[http://mantis.innovaphone.com/view.php?id=77992 77992]
|}
|}
Now the presence is updated in myPBX for other PBX objects if the access is changed to allowed.<!---->
Status message "Recording" needs to be displayed during recording.<!---->
      
      
=== PBX: Conference presence info ===
=== Gateway: Wrong Media info sent for calls forwarded from a VOIP interface to another VOIP interface ===


{|
{|
Line 6,509: Line 6,367:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70581 70581]
|[http://mantis.innovaphone.com/view.php?id=78024 78024]
|}
|}
Now the broadcast conference object updates the presence info and is shown as callable in myPBX.<!---->
This could cause wrong information to be displayed on a PBX calls page. Esspecially SRTP was not indicated.<br/><br/>Better to not send any Media Info in this case.<!---->
      
      
=== myPBX launcher: Hotkey does not work with some applications ===
=== IP241,IP222,IP232: Wrong display name in call list ===


{|
{|
Line 6,520: Line 6,378:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70623 70623]
|[http://mantis.innovaphone.com/view.php?id=78047 78047]
|}
|}
Try to get the selected phone number using WM_COPY before sending CTRL-C. This should help in some of the cases where the hotkey didn't work before.<!---->
Name identification of remote party is to be displayed,<br/>but name identification of diverting party was displayed instead<br/>on incoming calls list.<!---->
      
      
=== SIP: Do not try to map local listen port with STUN when SIP/TCP is used ===
=== phone: the call which was the active call at start of a conference was not automatically cleared  on a release from remote ===


{|
{|
Line 6,531: Line 6,389:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70654 70654]
|[http://mantis.innovaphone.com/view.php?id=78086 78086]
|}
|}
Mapping of local listen port is only required fur SIP/UDP<!---->
The call which was the active call at start of a conference had to be cleared manually but the call which was on hold at start of conference was cleared automatically.<br/>Now any call will be automatically cleared when relesed from remote.<!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
=== IP222: Use R-key as BACKSPACE on edit fields ===
=== phone_orchid: remaining call mute after remote relase for the call which was the active call at start of a conference ===


{|
{|
Line 6,542: Line 6,402:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70664 70664]
|[http://mantis.innovaphone.com/view.php?id=78091 78091]
|}
|}
Not ESC-key.<br/>ESC-key is for leaving screen only.<!---->
did not happen after a remote release for the call which was on hold at start of the conference.<!---->
''Status:''
checked in to 10.00, 9.00
      
      
=== Trap: When Dectmaster registers user at PBX using SIP protocol ===
=== phone: ip222,ip232: Muting USB headset micro via microphone key did not work but MICROPHONE OFF was displayed ===


{|
{|
Line 6,553: Line 6,415:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70675 70675]
|[http://mantis.innovaphone.com/view.php?id=78159 78159]
|}
|}
After closing regstration Dectmaster starts another call.<br/>Call is rejected, but signaling enity is deleted before call object.<!---->
.<!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
=== IP-DECT: Hanging call after failed call transfer ===
=== IP241,IP222,IP232: Trap in font rendering ===


{|
{|
Line 6,564: Line 6,428:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70756 70756]
|[http://mantis.innovaphone.com/view.php?id=78210 78210]
|}
|}
The hanging call in dectradio is fixed which occurred if a user does an unattended call transfer and the call transfer fails.<!---->
Some characters at beginning of line may cause trap.<!---->
      
      
=== SIP: Allow STUN to be used to map local media ports on SIP/TCP interfaces ===
=== Gateway: Trap when interworking Call Completion ===


{|
{|
Line 6,575: Line 6,439:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70809 70809]
|[http://mantis.innovaphone.com/view.php?id=78228 78228]
|}
|}
STUN cannot map signaling TCP port, but UDP media ports.<!---->
Trap when interworking Call Completion.<br/><br/>LOG CALL 6 A:Call    ->                        / PRI2::->*::<br/>R_CALL free error c18a59b8<br/><!---->
''Status:''
medialib.h<br/>media.cpp<br/>h323ch.h/cpp<br/>sip.h/cpp<br/>siptrans.h/cpp<br/>
      
      
=== IPVA, Keyboard Console, Credentials With Special Chars Couldn't Be Entered ===
=== phone: ip241: memory leak when ip230x extension module is attached and used  ===


{|
{|
Line 6,588: Line 6,450:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70873 70873]
|[http://mantis.innovaphone.com/view.php?id=78311 78311]
|}
|}
The Scancode table wasn't finished<!---->
one packet lost per keystroke on extension module<!---->
      
      
=== SIP: Reduce memory footprint of SIP stack ===
=== PBX: Dynamic group function keys did not work with non-ascii characters in group name ===


{|
{|
Line 6,599: Line 6,461:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70886 70886]
|[http://mantis.innovaphone.com/view.php?id=78322 78322]
|}
|}
Free INVITE request buffer when receiving ACK. No need to keep any longer.<br/>Delete INVITE client transaction when cancelling.<!---->
Name was not correctly converted to internal utf-8 representation<!---->
      
      
=== SIP: No route processing if neither Record-Route header nor Contact header is present ===
=== phone: ip222,ip232: trace USB media data only if explicitely requested ===


{|
{|
Line 6,610: Line 6,472:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=70971 70971]
|[http://mantis.innovaphone.com/view.php?id=78326 78326]
|}
|}
Misleading trace message:<br/>  sip_call::process_routing(0xA8) Unsupported transport protocol: sip:user@domain.com;user=phone<!---->
use "config add USB-HOST AUDIO /trace" if data is needed<!---->
      
      
=== IP6010: Wrong timer under high load ===
=== H.323: Potential Trap in case of special malformed RAS registration ===


{|
{|
Line 6,621: Line 6,483:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71001 71001]
|[http://mantis.innovaphone.com/view.php?id=78368 78368]
|}
|}
-Clear IRQ in handle-interrupt after os_interrupt is too late, since IRQ´s a enabled again and e.g. the timer irq is called again if a lower level IRQ like the enet occurs.<br/>-The IRQ needs to be cleared in the serial-irq handler, in all case. After the serial-irq other interrupts are enabled.<br/><!---->
A NULL pointer access could happen<!---->
''Status:''
ip6010.cpp<br/>ip6010.h
      
      
=== SIP: Media negotiation problem ===
=== SIP: Removed b=TIAS attribute from media description ===


{|
{|
Line 6,634: Line 6,494:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71009 71009]
|[http://mantis.innovaphone.com/view.php?id=78379 78379]
|}
|}
Internal re-negotiation during early media on incoming SIP call.<br/>Provide received offer to app again.<!---->
Causes trouble on some SIP providers (neotel.at)<!---->
      
      
=== SIP: Offer CLEARMODE only if bearer capabilities are "Unrestricted Digital Information" ===
=== DHCP server leases with hostnames containing non-ascii latin1 characters (for example Umlauts) could not be displayed ===


{|
{|
Line 6,645: Line 6,505:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71162 71162]
|[http://mantis.innovaphone.com/view.php?id=78382 78382]
|}
|}
On "Unrestricted Digital Information" only CLEARMODE is offered (no audio codecs).<br/>On other bearer capabilities no CLEARMOE is offered (only audio codecs).<!---->
In pre-V9 firmware hostnames were stored latin1-encoded. Names contaning non-ascii latin1 charaters must be converted to UTF8 before display.<!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
=== phone: dialog and presence subscriptions sometimes got lost after PBX restart when phone config was stored on PBX ===
=== MyPBX: Changing password did not work any more ===


{|
{|
Line 6,656: Line 6,518:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71198 71198]
|[http://mantis.innovaphone.com/view.php?id=78391 78391]
|}
|}
This happened specially when both "Store Phone Config" and "Discard Config on Phone" was checked in the user object because of a unsubscribe/subscribe race condition.\t<br/><!---->
Collateral damage of fix #75453: PBX/IP-DECT: User password length<!---->
      
      
=== when editing a phone config template the dialing location inherited from a predecessor template was stored in the edited templat ===
=== phone: ip222, ip232: USB headset sometimes mute after disconnect ===


{|
{|
Line 6,667: Line 6,529:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71246 71246]
|[http://mantis.innovaphone.com/view.php?id=78414 78414]
|}
|}
after a template has been edited unchanged information units inherited from predecessor templates must be removed from the edited template. this did not work for the dialing location and thus a later change in a predecessor template had no effect. <!---->
depending on isochronous transfer state at disconnect time. seems to happen more often with relatively cheap headsets.<!---->
      
      
=== SIP: Endpoints behind NAT could not register at public PBX ===
=== SIP: Be save against sudden death of SIP caller ===


{|
{|
Line 6,678: Line 6,540:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71266 71266]
|[http://mantis.innovaphone.com/view.php?id=78460 78460]
|}
|}
PBX must send response to the (public) IP address where request was received from.<br/>Not to the (private) IP address in Via header or Contact-URI.<!---->
Lifetime of an INVITE trasnaction is not limited by any timeout<br/>after provisional response has been send/received.<br/>Sudden death of a caller make calls hang forever.<br/>Now overall lifetime of an INVITE server transaction is limited to 3 minutes.<br/>After expiration fimnal reject response is sent and call is released.<!---->
      
      
=== ipva, update to _platform_tracing.xsl ===
=== IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax failure after transfer ===


{|
{|
Line 6,689: Line 6,551:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71274 71274]
|[http://mantis.innovaphone.com/view.php?id=78487 78487]
|}
|}
ip6 missing<!---->
channel was reconfigured on remote switch to T38, without a new activate fax relay command<!---->
''Status:''
ac_dsp3.cpp
      
      
=== SIP: No media after accepting a waiting call ===
=== PBX Boolean: Avoid unnecessary load by boolean monitoring ===


{|
{|
Line 6,700: Line 6,564:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71288 71288]
|[http://mantis.innovaphone.com/view.php?id=78504 78504]
|}
|}
Call waiting on a phone.<br/>Going onhock while another call is waiting starts ringer.<br/>After going offhook again the waiting call is accepted, but no media in both directions.<!---->
A call was sent to the monitoring endpoint every 10s<!---->
      
      
=== CX0 Wave-Encoding Not Working If Fact-Chunk Present In Header ===
=== IP241,IP222,IP232: No RTP send on outgoing call in some cases ===


{|
{|
Line 6,711: Line 6,575:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71290 71290]
|[http://mantis.innovaphone.com/view.php?id=78550 78550]
|}
|}
CX0 Wave-Encoding Not Working If Fact-Chunk Present In Header<!---->
No RTP send on outgoing call in some cases<!---->
      
      
=== DHCP Server Identifier could not be cleared via WEB interface ===
=== IP6000: Traps in DSP driver under high load ===


{|
{|
Line 6,722: Line 6,586:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71305 71305]
|[http://mantis.innovaphone.com/view.php?id=78591 78591]
|}
|}
When the field 'Server Identifier' was cleared and OK was pressed the just cleared value reappeared but after a reset 'Server Identifier' was clear. <!---->
under high load timing may change. Checks in driver relaxed to take this into account.<!---->
      
      
=== IP241: Show ISDN display information in full length ===
=== PBX-SOAP: Limit on size for Admin call too small ===


{|
{|
Line 6,733: Line 6,597:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71330 71330]
|[http://mantis.innovaphone.com/view.php?id=78592 78592]
|}
|}
Use scrolling if required<!---->
Only 4k was allowed<!---->
      
      
=== NAT: Mapping to different internal UDP port did not work ===
=== SoftwarePhone: Default ToS value for RTP ===


{|
{|
Line 6,744: Line 6,608:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71339 71339]
|[http://mantis.innovaphone.com/view.php?id=78593 78593]
|}
|}
Configured destination port got lost after configuration<!---->
Now the default ToS value for RTP packets is 0xB8.<!---->
      
      
=== phone: send config to PBX only when the config was edited on phone ===
=== PBX Filter: Maximum length increased from 13 to 17 digits ===


{|
{|
Line 6,755: Line 6,619:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71387 71387]
|[http://mantis.innovaphone.com/view.php?id=78594 78594]
|}
|}
A config from an older PBX may contain duplicate elements which are stripped by the phone. I such a stripped config is sent back to the PBX the PBX will return the old config again.<!---->
13 digits could be too little if an international number including a prefix of an external line was to be checked<!---->
      
      
=== SIP: Send first NOTIFY(dialog) after sending 200/OK for SUBSCRIBE(dialog) ===
=== IP241,IP222,IP232: Wrong call direction indication on PARTNER fkey ===


{|
{|
Line 6,766: Line 6,630:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71413 71413]
|[http://mantis.innovaphone.com/view.php?id=78642 78642]
|}
|}
NOTIFY(dialog) was sent before 200/OK for SUBSCRIBE(dialog)<br/><br/>\tRequest: SUBSCRIBE sip:150@192.168.200.14<br/>\tStatus: 401 Unauthorized<br/>\tRequest: SUBSCRIBE sip:150@192.168.200.14<br/>\tStatus: 100 Trying<br/>\tRequest: NOTIFY sip:158@192.168.200.214:5060<br/>\tStatus: 200 OK<br/>\tStatus: 481 Subscription Does Not Exist<br/><br/><!---->
When partner calls someone else it my displayed as if partner is called.<br/>(Only if option "Show connected Party in busy State" is enabled)<br/><!---->
      
      
=== SIP: Interop with Nortel CS1000 SIPLine GW ===
=== IP222 IP232 IP241: Sporadic noisy sidetone ===


{|
{|
Line 6,777: Line 6,641:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71426 71426]
|[http://mantis.innovaphone.com/view.php?id=78649 78649]
|}
|}
Nortel sends 183/Progress with 'sendrecv' answer<br/>followed by UPDATE with 'inactive' offer<br/>followed by UPDATE with 'sendrecv' offer.<br/><br/>Innovaphone SIP stack remains in 'inactive' state.<!---->
Analog and digital sidetonegain were enabled,now only the analog sidetonegain is used.<!---->
''Status:''
ac_codec3.cpp
      
      
=== H.323 Signaling Timeout event was generated for 'normal' RAS re-transmissions ===
=== Gateway: Handling of blind transfer gets stuck in routing ===


{|
{|
Line 6,788: Line 6,654:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71434 71434]
|[http://mantis.innovaphone.com/view.php?id=78709 78709]
|}
|}
A single retransmission is normal under heavy load, so this is no reason for an event. Signaling Timeout events are now generated only if they cause a state change.<!---->
Routing of transfer call stops with: reason='incomplete'<br/>But during transfer no dialing digits will follow.<!---->
   
= V9 Hotfix 10 (90600.12) =
Changes included in Version 9 hotfix10
[http://mantis.innovaphone.com/view.php?id=78895 Definition]
 
== New Features ==
 
 
      
      
=== H.323 re-negotiation: Don't reuse media proposals if a select was already sent ===
=== Linux: Enable/Disable configuration ===


{|
{|
Line 6,799: Line 6,673:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71435 71435]
|[http://mantis.innovaphone.com/view.php?id=78836 78836]
|}
|}
This fixes compatibility issues with SIP, especially when SRTP is used.<!---->
With the new feature #78786 the configuration is only saved in the downloaded configuration file with password. Now the information is also included in the configuration file with standard password and in the file downloaded by the update server.<!---->
   
== Bug Fixes ==
 
 
      
      
=== IP241: Show both parties of each call on screen "Pickup List" ===
=== PBX: Trap if enabling Unknown Registrations ===


{|
{|
Line 6,810: Line 6,688:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71455 71455]
|[http://mantis.innovaphone.com/view.php?id=78904 78904]
|}
|}
Currently only calling party infomation is displayed.<!---->
Collateral Damage from<br/><br/>fix: #77261: Phone->PBX LDAP Search returns normalised number to be dialled by phone<!---->
   
= V9 Hotfix 11 (90600.14) =
Changes included in Version 9 hotfix11
[http://mantis.innovaphone.com/view.php?id=79001 Definition]
 
== New Features ==
 
 
      
      
=== phone: when a menu item was selected by the adjacent key sometimes the function assigned to this key was also executed ===
=== HTTP-Client: MD5-sess authentication ===


{|
{|
Line 6,821: Line 6,707:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71469 71469]
|[http://mantis.innovaphone.com/view.php?id=77773 77773]
|}
|}
The function assigned to the menu item (for example pickup) was executed first and then the key-function (for example dial). <!---->
HTTP Digest Authentication with alogrithm=MD5-sess.<br/>Choose the first supported "WWW-Authenticate" line from 401 response headers.<br/><br/>Needed for new versions of IIS.<br/><br/><!---->
''Status:''
http://wiki.innovaphone.com/index.php?title=Support:DVL-Feature_Requests#HTTP_Client
      
      
=== PBX: Wrong web page when submitting an object and an error happens ===
=== Phone: Possibility to reject incoming SIP calls with customized reason phrase ===


{|
{|
Line 6,832: Line 6,720:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71470 71470]
|[http://mantis.innovaphone.com/view.php?id=77928 77928]
|}
|}
If for example a dupicate number is detected, the same web page should be displayed including the error message for the duplicate number. But not the same page was displayed but a page which could contain information not related to the object.<!---->
E.g. "SIP/2.0 480 Do not disturb" instead of  "SIP/2.0 480 Temporarily unavailable"<!---->
      
      
=== v8 to v9 upgrade problem with gateway registration names containing non-ASCII characters ===
=== Active Directory Replication: Editfield for Poll Timer added ===


{|
{|
Line 6,843: Line 6,731:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71474 71474]
|[http://mantis.innovaphone.com/view.php?id=78631 78631]
|}
|}
In general this was a problem with config line arguments seperated by ':'. This happened with the <number>:<name> argument within gateway definitions. The ':' was url-encoded and <name> interpreted as <number><!---->
If change notifications cannot be received from an AD, a poll timer can be specified. A re-replication is going to take place after the poll timer expired.<!---->
      
      
=== PBX: CFU was executed on PRESENCE_PUBLISH/SUBSCRIBE calls ===
=== X.509: Support for PKCS#12 files ===


{|
{|
Line 6,854: Line 6,742:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71479 71479]
|[http://mantis.innovaphone.com/view.php?id=78820 78820]
|}
|}
This was unexpected behaviour. You want to see the presence status of the configured user and not the presence status of the destination to which this user has configured a call forwarding<!---->
Support for certificate import using password encrypted PKCS#12 files.<br/><br/>Currently the following encryption types are supported:<br/><br/>pbeWithSHAAnd128BitRC4 (1.2.840.113549.1.12.1.1)<br/>pbeWithSHAAnd40BitRC4 (1.2.840.113549.1.12.1.2)<br/>pbeWithSHAAnd3-KeyTripleDES-CBC (1.2.840.113549.1.12.1.3)<br/>pbeWithSHAAnd128BitRC2-CBC (1.2.840.113549.1.12.1.5)<br/>pbewithSHAAnd40BitRC2-CBC (1.2.840.113549.1.12.1.6)<br/><!---->
      
      
=== SIP: Interoperability with MX-ONE ===
=== myPBX: Pass language code to reporting ===


{|
{|
Line 6,865: Line 6,753:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71480 71480]
|[http://mantis.innovaphone.com/view.php?id=78874 78874]
|}
|}
A semi-attended transfer fails if MX-ONE sends INVITE(Replaces)<br/>instead of 200/OK when connecting a call.<!---->
For localization of the call list.<!---->
      
      
=== Phone: Presence-Fkey did not always show presence set by myPBX ===
=== Phone: Show number of voice messages on label of MWI fkey ===


{|
{|
Line 6,876: Line 6,764:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71487 71487]
|[http://mantis.innovaphone.com/view.php?id=78894 78894]
|}
|}
Problem:<br/>1) Set presence A with IP phone (fkey shows A)<br/>2) Set presence B with myPBX (fkey shows B)<br/>3) Delete presence with IP phone (fkey shows no presence)<br/><br/>Now Fkey shows presence B.<br/><!---->
Use "%u Message(s)" as label text for active state of MWI fkey.<!---->
      
      
=== PBX: Tooltip on "PBX/Config/Log Calls" checkmark wrong ===
=== IP-DECT: Cisco SIP conferencing ===


{|
{|
Line 6,887: Line 6,775:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71507 71507]
|[http://mantis.innovaphone.com/view.php?id=78988 78988]
|}
|}
It read "If not checked PBX calls are logged", should be "If checked PBX calls are logged"<!---->
Cisco proprietary softkey events implemeneted to start a 3pty conference<!---->
      
      
=== IP241: Added flag to hide background-image configuration from phone menu ===
=== IP-DECT: Feature Code for calling predefined service URIs ===


{|
{|
Line 6,898: Line 6,786:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71513 71513]
|[http://mantis.innovaphone.com/view.php?id=79028 79028]
|}
|}
See http://wiki.innovaphone.com/index.php?title=Reference9:Concept_Fine_grained_function_hiding#More_Information<br/>for more information<!---->
User for Cisco compatibility<!---->
      
      
=== Media: Discard RTP packets from wrong source ===
=== Cisco Softkey Features, Status updates and Park ===


{|
{|
Line 6,909: Line 6,797:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71515 71515]
|[http://mantis.innovaphone.com/view.php?id=79030 79030]
|}
|}
Packets arriving at RTP port must be discarded if the source if not the expected one.<br/>To be save against DOS attack and for interop with Lync.<br/>In some scenarios Lync starts sending RTP packets while having the call set to 'inactive'.<!---->
suported_mask needs more flags<br/>Softkey Feature Park<!---->
      
      
=== PBX: Blind transfer with consultation to BC-Conference failed ===
=== IP-DECT: Cisco Park ===


{|
{|
Line 6,920: Line 6,808:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71540 71540]
|[http://mantis.innovaphone.com/view.php?id=79032 79032]
|}
|}
The call was disconnected<!---->
Send Park softkey event<!---->
      
      
=== H.323: No Alarm/Event should be generated by shutting down registration due to reset ===
=== IP-DECT: Forward Cisco Status Line Updates related to a call ===


{|
{|
Line 6,931: Line 6,819:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71545 71545]
|[http://mantis.innovaphone.com/view.php?id=79036 79036]
|}
|}
no event is sent to the application about this kind of unregistration<!---->
.<!---->
      
      
=== phone-orchid: micro-speaker loop of embedded test did not work ===
=== IP-DECT: SRTP for an OEM device ===


{|
{|
Line 6,942: Line 6,830:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71566 71566]
|[http://mantis.innovaphone.com/view.php?id=79061 79061]
|}
|}
Der Test war im Treiber nicht freigeschaltet<!---->
Now SRTP is configurable for an OEM device.<!---->
      
      
=== myPBX: Send existing invitation links to users that join the chat session later ===
=== IP-DECT: Forward Cisco Status Line Updates not related to a call ===


{|
{|
Line 6,953: Line 6,841:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71608 71608]
|[http://mantis.innovaphone.com/view.php?id=79104 79104]
|}
|}
Users A and B have a chat session.<br/><br/>1. A starts application sharing or audio conference<br/>-> A and B receive an invitation link<br/><br/>2. A adds a third user C to the chat and C accepts<br/>-> C should also receive the invitation link <br/><br/>This worked only for audio conference but not for application sharing.<br/><!---->
.<!---->
      
      
=== IP-DECT: Ring back tone after transfer ===
=== Merge of v10 changes: Dynamic Timer Management/oem serial no ===


{|
{|
Line 6,964: Line 6,852:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71610 71610]
|[http://mantis.innovaphone.com/view.php?id=79127 79127]
|}
|}
If in-band ring back tone is sent and the call is transferred to a new destination with no in-band ring back tone, a local ring back tone must be played to the DECT handset. This is fixed now.<!---->
Better power management on some platforms<br/>Display of oem serial no<br/>stanard behaviour of delete on NULL<!---->
      
      
=== Ring Back tone missing after transfer when in-band tone was provided before but not after transfer ===
=== Merge of v10 changes: Memory Management ===


{|
{|
Line 6,975: Line 6,863:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71612 71612]
|[http://mantis.innovaphone.com/view.php?id=79129 79129]
|}
|}
The initial local ring back tone must be restarted when no data is received after transfer.<!---->
for OEM needs<!---->
      
      
=== PBX: Send Name Identification with CLIR calls if "No CLIR on internal Calls" ===
=== Merge of v10 changes: Firmware File check when updating improved ===


{|
{|
Line 6,986: Line 6,874:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71643 71643]
|[http://mantis.innovaphone.com/view.php?id=79130 79130]
|}
|}
The feature "No CLIR on internal Calls" did not work completetly. The number was sent, but the Name Id was still suppressed<!---->
for OEM needs<!---->
      
      
=== phone_orchid: Ring Back tone missing  when Silence Compression is enabled ===
=== Merge of v10 changes: More ciphers ===


{|
{|
Line 6,997: Line 6,885:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71670 71670]
|[http://mantis.innovaphone.com/view.php?id=79134 79134]
|}
|}
When Silence Compression was enabled the tone generation was not triggered<!---->
for OEM SRTP and other Certificate formats<!---->
      
      
=== IP232: Fix for touch handling ===
=== Merge of v10 changes: Internal interfaces enhanced ===


{|
{|
Line 7,008: Line 6,896:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71674 71674]
|[http://mantis.innovaphone.com/view.php?id=79135 79135]
|}
|}
Do not open touch keyboard on controls with CTRL_READONLY.<br/>Do not open touch keyboard on controls without CTRL_ACTIVATE.<br/>Using KEY_SHIFT has modified key to uppercase permanently.<br/>Cursor positioning on text controls did not work.<br/>Multi-line editor control was not displayed after hiding touch keyboard.<br/>Hide overlay keyboard after next touched key.<br/>Move and resize editor control when activating touch keyboard.<!---->
for OEM support<!---->
      
      
=== SIP: Trap on timer expiration during call release ===
=== Merge of v10 changes: ZipZip Tone definition updates ===


{|
{|
Line 7,019: Line 6,907:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71699 71699]
|[http://mantis.innovaphone.com/view.php?id=79136 79136]
|}
|}
Media negotiation watchdog timer expired after final SIG_REL went to app.<br/>But before app deleted the call object.<!---->
should overlay normal voice<!---->
      
      
=== phone: display info provided by SETUP or CONNECT was ignored ===
=== Merge of v10 changes: Library update ===


{|
{|
Line 7,030: Line 6,918:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71727 71727]
|[http://mantis.innovaphone.com/view.php?id=79140 79140]
|}
|}
only the display info provided by an INFO event was handled  <!---->
.<!---->
      
      
=== phone_orchid: continuous dialing tone stops after 10 seconds and is not restarted anymore ===
=== Merge of v10 changes: permit DHCP mode change between client/disabled without reset ===


{|
{|
Line 7,041: Line 6,929:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71753 71753]
|[http://mantis.innovaphone.com/view.php?id=79152 79152]
|}
|}
the continuous dialing tone as used in most countries must be played as long as no digit is entered.<!---->
for OEM support<!---->
      
      
=== IP230/IP240: Sporadic traps during manufacturing programming ===
=== ASN.1 BER: Support for indefinite-length encoding ===


{|
{|
Line 7,052: Line 6,940:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71778 71778]
|[http://mantis.innovaphone.com/view.php?id=79260 79260]
|}
|}
Inefficient loop during mac address programming<!---->
needed for PKCS#12 import<br/>only decoding<br/><!---->
''Status:''
flash_firmware.cpp<br/>
      
      
=== IPVA, Reset didn't work on VMware Player 4.0 ===
=== ASN.1 BER: Support for constructed octet strings ===


{|
{|
Line 7,065: Line 6,951:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71818 71818]
|[http://mantis.innovaphone.com/view.php?id=79261 79261]
|}
|}
IPVA, Reset didn't work on VMware Player 4.0<!---->
needed for PKCS#12<br/>only decoding, only one level of nesting<!---->
      
      
=== IP241: Slow screen update when changing fkey type on fkey configuration screen ===
=== IP810: Config Wizard ===


{|
{|
Line 7,076: Line 6,962:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71831 71831]
|[http://mantis.innovaphone.com/view.php?id=79284 79284]
|}
|}
Removing and adding config controls is very slow.<!---->
Mechanism needed to add platform specific files (IP810 uses same firmware as IP6010, IP3010, ...)<!---->
      
      
=== ip6010 - calling &amp;amp; dtmf tone timing did not  work ===
=== Merge of v10 changes: Support for SIP specific phone features ===


{|
{|
Line 7,087: Line 6,973:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71966 71966]
|[http://mantis.innovaphone.com/view.php?id=79306 79306]
|}
|}
the tone time was calculated much too short<!---->
for OEM support<!---->
''Status:''
checked in to 9.00
      
      
=== IP232: Redesign of touch keyboard ===
=== phone: permit to select the directories to be searched on indirect dialing ===


{|
{|
Line 7,098: Line 6,986:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71968 71968]
|[http://mantis.innovaphone.com/view.php?id=79307 79307]
|}
|}
Redesign of touch keyboard<!---->
if there is for example a huge external directory used for inbound name resolution and dialing is restricted to internal partners it may be hard to find internal numbers via combined directory search.<br/><br/>  config add PHONE DIR-UI /dial-mask <selection><br/><br/>selection:<br/>  0x02 - local directory<br/>  0x04 - PBX directory<br/>  0x08 - external directory<br/><br/>  0x06 = local + PBX<br/>  0x0E = local + PBX + external<br/><!---->
      
      
=== IPVA, Trap After Failed FW Upload ===
=== IP-DECT: Signal waiting calls to handset ===


{|
{|
Line 7,109: Line 6,997:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71978 71978]
|[http://mantis.innovaphone.com/view.php?id=79349 79349]
|}
|}
If a firmware upload failed with "wrong checksum" the box trapped occasionally, leaving the boot disk in an inconsistent state.<!---->
Was only acoustic information.<br/><br/><!--<br/>Changed files:<br/>fty.h<br/>fty.cpp<br/>dectradio_dectcall.cpp<br/>--><!---->
      
      
= V9 Hotfix 6 (90600.06) =
=== IP-DECT: LDAP phonebook (IP1202) ===
Changes included in Version 9 hotfix6
[http://mantis.innovaphone.com/view.php?id=71745 Definition]
 
== New Features ==
 
 
   
=== IP232: Presence fkey shows current activity, but not current note ===


{|
{|
Line 7,128: Line 7,008:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71611 71611]
|[http://mantis.innovaphone.com/view.php?id=79483 79483]
|}
|}
Show current note in second line of fkey label.<!---->
The new feature LDAP phone book for the IP1202 is added now.<br/><!--<br/>Changed files:<br/>config_options.h<br/>config_options.cpp<br/>ascom/ascom_1202.mak<br/>ascom/httpfiles/_inn_dect.xml<br/>dect_comcerto/platform/config.h<br/><br/>New files:<br/>ascom/phonebook_ldap/phonebook_ldap.mak<br/>ascom/phonebook_ldap/phonebook_ldap.h<br/>ascom/phonebook_ldap/phonebook_ldap.cpp<br/>ascom/phonebook_ldap/phonebook_ldap.xsl<br/>ascom/phonebook_ldap/phonebook_ldap.xml<br/>ascom/phonebook_ldap/phonebook_ldap_hdr.xml<br/>ascom/phonebook_ldap/phonebook_ldap_request.h<br/>ascom/phonebook_ldap/phonebook_ldap_request.cpp<br/>ascom/phonebook_ldap/phonebook_ldap_search.h<br/>ascom/phonebook_ldap/phonebook_ldap_search.cpp<br/>ascom/phonebook_ldap/phonebook_ldap_entry.h<br/>ascom/phonebook_ldap/phonebook_ldap_entry.cpp<br/>ascom/phonebook_ldap/config_ldap.h<br/>ascom/phonebook_ldap/config_ldap.cpp<br/>ascom/phonebook_ldap/config_dl.h<br/>ascom/phonebook_ldap/config_dl.cpp<br/>--><!---->
      
      
=== myPBX: Cause codes for instant messaging ===
=== IP-DECT: Three party conference with innovaphone PBX ===


{|
{|
Line 7,139: Line 7,019:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71878 71878]
|[http://mantis.innovaphone.com/view.php?id=79503 79503]
|}
|}
Display causes why a chat session has been terminated.<br/><br/>For example: Unassigned number, rejected, no response...<!---->
Now it is possible to make a three party conference with DECT handsets with an innovaphone PBX (an innovaphone device with the CONF interface). The conferencing unit must be configured in the DECT master. The conference call is established with the feature code 'R' + '3'.<br/>This fix also includes a rework of the DECT radio module. It can handle more than one waiting or hold call now.<br/><br/><!--<br/>Changed files:<br/>common/style/work.css (innovaphone)<br/>common/interface/channel.h<br/>common/interface/channel.cpp<br/>common/interface/signal.h<br/>common/interface/signal.cpp<br/>dect/dectdisplay_if.h<br/>dect/master/dectmaster.h<br/>dect/master/dectmaster.cpp<br/>dect/master/dectmaster.xsl<br/>dect/localusers/dectlocalusers.h<br/>dect/localusers/dectlocalusers.cpp<br/>dect/radio/dectradio.mak<br/>dect/radio/dectradio.h<br/>dect/radio/dectradio.cpp<br/>dect/radio/dectradio_calls.xsl<br/>ascom/display/dectdisplay.h<br/>ascom/display/dectdisplay.cpp<br/>ascom/httpfiles/asc_dectmaster.xsl<br/>ascom/httpfiles/_inn_asc_dectusers.xsl<br/>ascom/httpfiles/_inn_asc_dectmaster.xsl<br/>kirk/display/display.h<br/>kirk/display/display.cpp<br/><br/>New files:<br/>dect/radio/dectradio_call.h<br/>dect/radio/dectradio_call.cpp<br/>dect/radio/dectradio_fty_call.h<br/>dect/radio/dectradio_fty_call.cpp<br/>dect/radio/dectradio_handover_in.h<br/>dect/radio/dectradio_handover_in.cpp<br/>dect/radio/dectradio_handover_out.h<br/>dect/radio/dectradio_handover_out.cpp<br/>dect/radio/dectradio_reg.h<br/>dect/radio/dectradio_reg.cpp<br/>dect/radio/dectradio_uninit_reg.h<br/>dect/radio/dectradio_uninit_reg.cpp<br/>dect/radio/dectradio_dectcall.h<br/>dect/radio/dectradio_dectcall.cpp<br/>dect/radio/dectradio_mastercall.h<br/>dect/radio/dectradio_mastercall.cpp<br/>dect/radio/dectradio_confcall.h<br/>dect/radio/dectradio_confcall.cpp<br/>--><!---->
      
      
=== myPBX: Make call state icons clickable ===
=== Gateway: 'cn' attribute for test interfaces ===


{|
{|
Line 7,150: Line 7,030:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71886 71886]
|[http://mantis.innovaphone.com/view.php?id=79506 79506]
|}
|}
Clicking a call state icon should have the same effect as clicking the text.<!---->
The 'cn' attribute is included in the test interfaces SIG0/1 for internal tests now.<br/><br/><!--<br/>Changed file:<br/>common/service/dummysig/dummysig.cpp<br/>--><!---->
   
== Bug Fixes ==
 
 
      
      
=== myPBX: Notification on disconnected chat sessions ===
=== IP222/IP232/IP241: Ethernet link configuration ===


{|
{|
Line 7,161: Line 7,045:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71918 71918]
|[http://mantis.innovaphone.com/view.php?id=78067 78067]
|}
|}
Display a text that the chat has been terminated.<!---->
In some case ethernet link configuration and display didnt work.<br/><br/>Now the autoneg status is shown if available, and the phy status if no autoneg didnt complete.<br/><br/>An option to use autoneg with fixed speeds and a link trace option (/ltrace) are added.<!---->
''Status:''
orchid_drv.cpp<br/>orchid_drv.h<br/>orchid_drv.xsl
      
      
=== Debug information on assertion ===
=== Media Relay: Don't send dummy RTP data on incoming calls ===


{|
{|
Line 7,172: Line 7,058:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71961 71961]
|[http://mantis.innovaphone.com/view.php?id=78252 78252]
|}
|}
More debug information on default event handler.<!---->
Dummy RTP data is sent just in case a NAT router is within the media path to set a UDP mapping in case both legs of the call contain a NAT router.<br/><br/>An example for such a situation is a call coming from a SIP provider thru a NAT router to the PBX, which forwards the call back out to the SIP provider. The NAT router won't get RTP data from inside to set the mappings.<br/><br/>The dummy RTP was sent to all legs of the call, but it is better to send it to outgoing call legs only, because endpoints calling in may turn off a local ringback tone when receiving dummy RTP<!---->
      
      
=== myPBX: Display reason why a browser is not supported ===
=== H.323/SIP: Avoid delayed SDP within outgoing calls as far as possible ===


{|
{|
Line 7,183: Line 7,069:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71972 71972]
|[http://mantis.innovaphone.com/view.php?id=78466 78466]
|}
|}
Currently it only says "Browser version not supported".<br/><br/>It should also tell what feature isn't supported:<br/>- XmlHttpRequest<br/>- WebStorage<br/>- PostMessage<br/><!---->
If media renegotiation is needed, to one side of the call an request for a media proposal (in SIP terms, this is an INVITE without SDP) is sent. The media proposal (in SIP terms SDP offer) is then forwarded to the other side.<br/><br/>The request for an offer should if possible not sent with the initial call, because there is equipment which does not handle 'delayed SDP'<!---->
      
      
=== myPBX: Forward invitation links to incoming chat calls that are added to a chat conference ===
=== Media Relay: Don't terminate T.38 protocol in media relay, forward transparently ===


{|
{|
Line 7,194: Line 7,080:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72280 72280]
|[http://mantis.innovaphone.com/view.php?id=78610 78610]
|}
|}
The same way like the invitation links (for application sharing and audio conferences) are forwarded to additional outgoing chat calls.<!---->
This should add robustness and reduce CPU load<!---->
      
      
=== myPBX: Display invitation links in the chat window of the organizer ===
=== Flash Directory: Display Error Messages ===


{|
{|
Line 7,205: Line 7,091:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72282 72282]
|[http://mantis.innovaphone.com/view.php?id=78656 78656]
|}
|}
The organizer of an application sharing session or a audio conference wants to see all corresponding links in the chat window.<!---->
Flash Directory: Display Error Messages<br/>Shedding light on replication problems.<br/><br/>currently only in 10.00<!---->
      
      
=== SoftwarePhone: Password encryption tool ===
=== RTP: No check for changed media destination because of received multicast ===


{|
{|
Line 7,216: Line 7,102:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72431 72431]
|[http://mantis.innovaphone.com/view.php?id=78698 78698]
|}
|}
A password encryption tool was added.<!---->
When receiving RTP normally a check is done if the source of the RTP is the same as we are sending to. If this is not the case, we assume the destination of the RTP is behind a NAT router and we change the destination address to the source address of the received RTP.<br/><br/>This does not make sense for received multicast.<!---->
      
      
=== SoftwarePhone: Crash dump ===
=== IP22 IP24 IP28 IP305: Faxempfang gelegentlich gestört ===


{|
{|
Line 7,227: Line 7,113:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72433 72433]
|[http://mantis.innovaphone.com/view.php?id=78803 78803]
|}
|}
Now, the SoftwarePhone writes a crash dump file.<!---->
DSP erhält zum falschen Zeitpunkt ein close-rtp.<!---->
''Status:''
ac_dsp3.cpp
      
      
=== myPBX: Display shortened links in chat messages ===
=== IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax receive max packet size increased to 1024 ===


{|
{|
Line 7,238: Line 7,126:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72445 72445]
|[http://mantis.innovaphone.com/view.php?id=78811 78811]
|}
|}
Shorten the displayed part of the URL in order to avoid text overflow.<br/>Cut overflowing texts at the right boundary of the chat box.<!---->
For internet fax devices<!---->
''Status:''
ac_fax3.cpp
      
      
=== PBX: New presence access flag 'online' ===
=== phone: ip222, ip232: USB controller sometimes hangs ===


{|
{|
Line 7,249: Line 7,139:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72538 72538]
|[http://mantis.innovaphone.com/view.php?id=78824 78824]
|}
|}
With this flag a user allows access to his online status<!---->
happened with certain headsets after fast connect/disconnect sequences<!---->
''Status:''
checked in to 10.00, 9.00<br/>merged to 90600
      
      
=== myPBX: Configure visibility of online state and presence separately ===
=== SIP: SDP answer sometimes contains 2 media descriptions for audio ===


{|
{|
Line 7,260: Line 7,152:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72632 72632]
|[http://mantis.innovaphone.com/view.php?id=78833 78833]
|}
|}
Phone status and IM status are now called online status.<br/>Activity and note are now called presence.<br/><br/>The visibility of both is now configured separately.<!---->
... instead of one for audio and one for video.<br/><br/>Offer:<br/><br/>    v=0<br/>    o=- 1 2 IN IP4 130.30.1.111<br/>    s=CounterPath Bria<br/>    c=IN IP4 130.30.1.111<br/>    t=0 0<br/>    m=audio 1902 RTP/AVP 0 8 18<br/>    a=alt:1 1 : 1A98X9/s 31UwK5IA 130.30.1. 111 1902 <br/>    a=fmtp:18 annexb=yes<br/>    a=rtpmap:18 G729/8000<br/>    a=sendrecv<br/>    a=x-rtp-session-id:ACE43888C33A4AC8895C7F8F69380914<br/>    m=video 28212 RTP/AVP 115 34 123 124 <br/>    a=alt:1 1 : yjz/mbyQ aWGuD8xJ 130.30.1. 111 28212 <br/>    a=fmtp:115 QCIF=1;CIF=1;I=1;J=1;T=1<br/>    a=fmtp:34 QCIF=1;CIF=1<br/>    a=fmtp:123 profile-level-id=42801e; packetization-mode=0; max-mbps=48600<br/>    a=fmtp:124 profile-level-id=42801e; packetization-mode=1; max-mbps=48600<br/>    a=rtpmap:115 H263-1998/90000<br/>    a=rtpmap:34 H263/90000<br/>    a=rtpmap:123 H264/90000<br/>    a=rtpmap:124 H264/90000<br/>    a=sendrecv<br/>    a=x-rtp-session-id:694677A9F65D419FB7B9E451FFCA02CC<br/><br/><br/>Answer:<br/><br/>    v=0<br/>    o=- 3141 1 IN IP4 10.230.5.1<br/>   s=-<br/>   c=IN IP4 10.230.5.1<br/>    t=0 0<br/>    m=audio 16410 RTP/AVP 0 13<br/>    b=TIAS:64000<br/>    a=ptime:20<br/>    a=silenceSupp:off - - - -<br/>    a=sendrecv<br/>    m=audio 16410 RTP/AVP 0 13<br/>    b=TIAS:64000<br/>    a=ptime:20<br/>    a=silenceSupp:off - - - -<br/>    a=sendrecv<!---->
      
      
=== PBX Mobility: Better support for Opticaller ===
=== Conference: Permanent noise in conference channel ===


{|
{|
Line 7,271: Line 7,163:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72701 72701]
|[http://mantis.innovaphone.com/view.php?id=78838 78838]
|}
|}
HTTP request to initiate call for mobile phone<!---->
Permanent noise in conference channel.<br/>Increasing with every new participant.<!---->
      
      
=== myPBX: New translations ===
=== PBX: When doing a show users, unnecessary data was sent to the browser ===


{|
{|
Line 7,282: Line 7,174:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72768 72768]
|[http://mantis.innovaphone.com/view.php?id=78878 78878]
|}
|}
For the other v9hotfix6 items regarding myPBX.<!---->
The data could be reduced and thus speeding up the display of big users lists.<!---->
      
      
=== Support for more automated Fax Tests ===
=== PBX Admin UI: Editing profiles was not possible if the user had too many contacts ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72826 72826]
|[http://mantis.innovaphone.com/view.php?id=78883 78883]
|}
|}
Switching from Audio to Fax to Audio to Fax. Special case which could happen if switch to Fax happened before connect.<!---->
The form is now submitted using POST instead of GET.<!---->
      
      
=== myPBX: Highlight incoming alerting calls ===
=== SIP: Wrong number of waiting messages (MWI) ===


{|
{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73093 73093]
|[http://mantis.innovaphone.com/view.php?id=78890 78890]
|}
|}
Highlight the alerting icon using an orange border.<!---->
MWI: Number of voice messages not decoded from incoming NOTIFY(application/simple-message-summary).<br/>Was either 1 or 0.<!---->
      
      
=== SIP: Interworking with KIRK Wireless Server 300 ===
=== SIP: Send even anonymous Diversion header ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73110 73110]
|[http://mantis.innovaphone.com/view.php?id=78954 78954]
|}
|}
If "KIRK Wireless Server 300 PCS10__ r3327"<br/>calls into PBX and is connected with Voicemail,<br/>Voicemail may send re-INVITE with SRTP key.<br/>Instead of accepting or ignoring the SRTP option,<br/>KIRK Wireless Server 300 rejects the whole SDP offer.<br/><br/>Now we retry the re-INVITE w/o offering SRTP key.<!---->
If diverting party's identity is unknown/hidden.<br/>Diversion: ;reason=user-busy<!---->
      
      
=== PBX: Forward original received ISDN display element to picking up or forwarded call ===
=== SIP: Failed to decode presence XML from CUCM ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73278 73278]
|[http://mantis.innovaphone.com/view.php?id=78967 78967]
|}
|}
In the display element from ISDN there could be vital information from equipment like crypto gateways. This should be available also if the call was picked or forwarded.<!---->
Support for namespace "urn:ietf:params:xml:ns:pidf:status:rpid" added.<!---->
      
      
=== SoftwarePhone: Support for Jabra SPEAK 410 USB with product id 0x0410 ===
=== IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax failure after transfer #2 ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73424 73424]
|[http://mantis.innovaphone.com/view.php?id=78969 78969]
|}
|}
Now, the SoftwarePhone supports Jabra SPEAK 410 USB with product id 0x0410.<!---->
tone flag was not disabled<!---->
''Status:''
ac_dsp3.cpp
      
      
=== SIP: Overriding cause code mappings did not work ===
=== H.323: Media problem with SIP call-completion on IP-DECT ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73477 73477]
|[http://mantis.innovaphone.com/view.php?id=78970 78970]
|}
|}
Mapping of SIP response codes into Q.931 cause codes could not be overwritten.<!---->
There was no media on the successful completion call (SDP recived in Progress was not forwarded)<!---->
      
      
=== Linux: Empty file check ===
=== SIP: Send 'Connected Number' in P-Asserted-Identity header of 200/OK ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73554 73554]
|[http://mantis.innovaphone.com/view.php?id=78974 78974]
|}
|}
Now, an empty file is not accepted as a successful file upload.<!---->
Send 'Connected Number' in P-Asserted-Identity header of 200/OK<br/>if different from original called number.<!---->
      
      
== Bug Fixes ==
=== SIP: Re-transmission of ACK with SDP answer ===
 
 
   
=== myPBX: Make more obvious that closing the chat window terminates the chat ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=71893 71893]
|[http://mantis.innovaphone.com/view.php?id=78975 78975]
|}
|}
Display warning message when the user tries to close the chat window.<br/>Show disconnect icon instead of close icon in the window.<!---->
Handling of re-transmissions of 200/OK with SDP offer.<br/>ACK must contain SDP answer.<!---->
      
      
=== IPVA, Optimize Flash-Related I/O Operations ===
=== IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: DSP packet debug didnt show some packets, version endian was wrong ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71954 71954]
|[http://mantis.innovaphone.com/view.php?id=78985 78985]
|}
|}
Specifically to reduce boot-time. Utilizing 64KB I/O accesses where possible.<!---->
cleanup<!---->
''Status:''
ac_491.cpp<br/>ac_494.cpp
      
      
=== H.323 RAS: Registration with authentication to account without authentication failed in a strange way ===
=== IP-DECT: Radio/Master calls switched to slowstart ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71958 71958]
|[http://mantis.innovaphone.com/view.php?id=78987 78987]
|}
|}
Information was missing that no authentication was supported, so it was continued to retry instead of giving up right away<!---->
If first media answer was received with a PROGRESS message, the call leg from the radio to the master was switched to progress (no EFC features are defined for the Progress message).<br/><br/>This could cause media problems later in the call with hold/retrieve/transfer.<!---->
      
      
=== TCP: Roundtrip measurement wrong in case of packet loss ===
=== SIP: REFER does not work as expected on Gateway interfaces without registration ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=71985 71985]
|[http://mantis.innovaphone.com/view.php?id=79008 79008]
|}
|}
In case of packet loss, way to high round trip values were measured. If the packet-loss was to high, this could result in a constantly increasing re-transmission timeout value.<!---->
Bad side effect of a previous fix in HF8:<br/>http://mantis.innovaphone.com/view.php?id=76309<br/><br/>Introducing new SIP interop tweak "Take Refer-To URI as Remote Target URI"<br/>to be found on the Gateway interface config (GWx).<br/>See http://mantis.innovaphone.com/file_download.php?file_id=66299&amp;type=bug<br/><br/><!--<br/>voip.h<br/>gk.cpp/h<br/>sip.cpp<br/>relay_edit_voip.xsl<br/>--><!---->
      
      
=== SIP: Workaround for delays in media negotiation ===
=== IP-DECT: Better handling when receiving a call, while a previous call is being released ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72189 72189]
|[http://mantis.innovaphone.com/view.php?id=79022 79022]
|}
|}
Workaround for delays in media negotiation caused by delayed Admission on H.323 calls.<!---->
Call to service number, which cause a callback can create problems.<br/><br/><!--<br/>Changed files:<br/>dectradio_call.h<br/>dectradio_call.cpp<br/>--><!---->
      
      
=== SIP: Trap on IP-DECT when re-configuring PBX link ===
=== Edss1 Interworking: Sending divertingLegInformation2 as PublicPartyNumber(was UnknownPartyNumber) ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72190 72190]
|[http://mantis.innovaphone.com/view.php?id=79025 79025]
|}
|}
85:2195:425:7 - REG_PRI.4 default(8102be48): serial_timeout<br/>85:2195:425:7 - Assertion failed line 748 in common/os/os.cpp, object deleted<br/><!---->
Such that the type of number resulting from an interface cgpn-map can also be  asn.1 encoded.<!---->
''Status:''
Merged to 09-80500
      
      
=== IP232: Input of capital Latin Extended-A ===
=== Gateway: Mapping of Message Center Number of MWI ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72226 72226]
|[http://mantis.innovaphone.com/view.php?id=79039 79039]
|}
|}
Input of capital Latin Extended-A<!---->
Mapping of Message Center Number of MWI according to CGPN mapping.<!---->
      
      
=== myPBX: Queue chat messages for im calls that are not yet alerting  ===
=== Qsig Interworking: Apply interface CGPN map to DGPN ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72237 72237]
|[http://mantis.innovaphone.com/view.php?id=79060 79060]
|}
|}
The following fix did not work across PBXes:<br/>#71608: myPBX: Send existing invitation links to users that join the chat session later <br/><br/>The problem was that messages were only queued in alerting state. When the call is made across PBXes this state isn't reached, yet.<!---->
Qsig Interworking: Apply interface CGPN map to DGPN<!---->
      
      
=== Scheduling improved to avoid processes not being scheduled during long flashman operations ===
=== PBX: A CFB at a PBX object was executed on call to busy user ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72243 72243]
|[http://mantis.innovaphone.com/view.php?id=79199 79199]
|}
|}
In version 7 it could happen, that IP and other processes were not scheduled any more during periods of long flashman operations (e.g. bootcode update or reorganizing flash).<br/><br/>In version 8 and higher there was already a fix for this problem, but this included special handling of the flashman priority level, which was not a good solution even if it worked.<!---->
It should only be executed, if the calls exceeded the Busy On number.<!---->
      
      
=== IP232: Touch keyboard key highlighting ===
=== PBX: Mapping of Message Center Number of MWI ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72268 72268]
|[http://mantis.innovaphone.com/view.php?id=79232 79232]
|}
|}
Added highlighting for key like SHIFT, BACKSPACE, etc.<!---->
Mapping of Message Center Number of MWI.<!---->
      
      
=== PBX: H.323 Names in some places (e.g. Waiting Queue Maps) containing non-ASCII charcters could break the user interface ===
=== PBX-SOAP: Conference Id was missing in call-info for parked calls ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72277 72277]
|[http://mantis.innovaphone.com/view.php?id=79243 79243]
|}
|}
This happened esspecially for Names as destinations for Waiting Queue maps, but it could happen at other places as well.<!---->
This caused problems with the innovaphone operator<!---->
      
      
=== SIP: Cleanup failed (resources leaking) ===
=== PBX-SOAP: UserClear cause should not be used for remote party ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72284 72284]
|[http://mantis.innovaphone.com/view.php?id=79249 79249]
|}
|}
Call and channel objects were not freed sometimes<br/>when INVITE was followed by CANCEL very fast.<!---->
This can result in strange call clearing without tones<!---->
      
      
=== IP241: Disable Link LED not working ===
=== PBX: Include error in resulting XML if submit-object with insufficient rights was attempted ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72338 72338]
|[http://mantis.innovaphone.com/view.php?id=79289 79289]
|}
|}
Timing problem with access to the paged MDIO registers<!---->
Problem with innovaphone operator.<!---->
''Status:''
orchid_drv.cpp
      
      
=== Trap: When accessing web interface ===
=== phone: ip222,ip232,ip241: wrong AM/PM time display on status line ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72362 72362]
|[http://mantis.innovaphone.com/view.php?id=79302 79302]
|}
|}
Seen once.<br/>Not reproducable yet.<!---->
15:19 PM was displayed instead of 03:19 PM<!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
=== Ip6010 DSP Disconnect timeout after fax session ===
=== Voicemail: Default for mwi.basicService now Speech(1) (was allServices(0)) ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72403 72403]
|[http://mantis.innovaphone.com/view.php?id=79303 79303]
|}
|}
Debugs added<br/>* enable with http://addr/debug.xml  DSP trace and DSP control message trace to printout  all packets to the DSP with a descriptive string. That allows to analyse the message flow to the DSP after a trap.<br/>* for further testing old fax disconnect procedure can be enabled with http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl  with "t38 skip fax close".<!---->
Voicemail: Default for mwi.basicService now Speech(1) (was allServices(0))<br/><br/>May from now on be overriden by variable "$_pbxmwiservice".<br/><!---->
''Status:''
ac_dsp3.cpp<br/>ac_dsp3.h<br/>ac_491.h<br/>dsp.xsl
      
      
=== IP2x2: Fixed fkey handling of builtin test ===
=== PBX: Local objects could not be called from Nodes with escapes as expected ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72442 72442]
|[http://mantis.innovaphone.com/view.php?id=79317 79317]
|}
|}
Fkey test did not work for touchkeys, key symbols were misplaced on screen<!---->
Was not possible to call at all, or the number was wrong<!---->
      
      
=== myPBX: Improve presentation of chat conferences ===
=== PBX-Trunk: Handling for Incomplete, Invalid, Busy, No Anser destinations for Media calls only ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72444 72444]
|[http://mantis.innovaphone.com/view.php?id=79319 79319]
|}
|}
Chat conferences should not be displayed as multiple calls but as a single item in the main window.<!---->
For example this should not be done for presence/dialog subscriptions<!---->
      
      
=== IP232: Parameters for touch sensor fixed for latest sensor ===
=== Refactored some ASN.1 BER Handling ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72455 72455]
|[http://mantis.innovaphone.com/view.php?id=79357 79357]
|}
|}
latest touch was treated like the very old version that needed other parameters.<br/>Old touch is not supported any more ( only 3 were build)<!---->
Became necessary in the run of: #79260: ASN.1 BER: Support for indefinite-length encoding<!---->
''Status:''
edt_touch.cpp
      
      
=== IP241,IP222,IP232: Make password configuration more convenient ===
=== PBX: Conference &amp;amp; VM ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72456 72456]
|[http://mantis.innovaphone.com/view.php?id=79359 79359]
|}
|}
When entering a passwort on the phone's ui content is displayed as "****".<br/>Show last enered character in plain text.<!---->
The last connected user is not disconnected although it is configured. It occurs if a VM PBX object forwards the call to the BC Conference object (like the innovaphone conferencing script). It is fixed now.<!---->
      
      
=== Ip6010 DSP Allow coder change from T38 to voice and back to T38 ===
=== IP-DECT: Subscription could get lost randomly with logout/login cycle ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72457 72457]
|[http://mantis.innovaphone.com/view.php?id=79397 79397]
|}
|}
..<!---->
Problem with user database handling<br/><!--<br/>dectusers.cpp<br/>dectusers.h<br/>--><!---->
''Status:''
ac_dsp3.cpp<br/>
      
      
=== IP222: Take center key of 4-way-navigation control an OK key ===
=== H.323: Media problem if PBX rtp-proxy is activated for CFNR to external destination ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72477 72477]
|[http://mantis.innovaphone.com/view.php?id=79430 79430]
|}
|}
Center key should enter selected menu item.<br/>Not leaving the current menu screen.<br/>ESC key can be used to leave current menu screen.<br/>Phone app will ask whether to save changes.<!---->
no voice<br/><!--<br/>h323ch.cpp<br/>h323sig.cpp<br/>--><!---->
      
      
=== myPBX: Name and Number Display not correct on IM sessions across PBXs ===
=== HTTP: Could not have sockets with same ports but different remote addresses ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72491 72491]
|[http://mantis.innovaphone.com/view.php?id=79458 79458]
|}
|}
for local sessions (same PBX) the number and the Long Name was displayed for the remote party and this should be the same for remote session.<br/><br/>Sometimes the number was missing, sometimes the Name was displayed instead of Long Name.<br/><br/>The number was not adjusted correctly if different Nodes were involved<!---->
This was a general problem with the new TCP stack, used together with IP6. Problem only happened for HTTP because only for HTTP this new stack is currently used.<br/><br/><!--<br/>ip_user.cpp<br/>tcp.cpp<br/>--><!---->
      
      
=== SoftwarePhone: Trap and media channel after conference ===
=== ISDN: Fix for call completion interworking ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72511 72511]
|[http://mantis.innovaphone.com/view.php?id=79498 79498]
|}
|}
If a call is disconnected during conference mode, the SoftwarePhone traps or there is no voice channel to the remaining party. This is fixed now.<!---->
CCBS/CCNR does not work in some cases<br/><br/><!--<br/>q950.cpp<br/>--><br/><br/><!---->
      
      
=== HTTP: Accept authentication with both Latin1 and UTF-8 coding ===
=== SIP: From-URI may got lost after call transfer ===


{|
{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=72512 72512]
|[http://mantis.innovaphone.com/view.php?id=79536 79536]
|}
|}
The server does not know what encoding is used in Basic- or Digest-Authentication. Therefore both encodings shall be tried.<!---->
UPDATE may be send out missing From-URI.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== ip28 codec hang when TEL port configured to '600Ohms' ===
=== AD Replication stuck after connection loss ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72521 72521]
|[http://mantis.innovaphone.com/view.php?id=79541 79541]
|}
|}
this option experienced problems in the past and should therefore not be selected. If selected the firmware will switch to default CTR21.<!---->
Didn't reconnect. Side effect of after-hf10 fix<br/><!--<br/>adrep.cpp<br/>--><!---->
      
      
=== PBX: Cause for release/reject of a IM session not signaled ===
=== PBX Waiting/Broadcast: Incomplete CDRs if CFNR configured on object ===


{|
{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=72533 72533]
|[http://mantis.innovaphone.com/view.php?id=79549 79549]
|}
|}
Better feedback to the User<!---->
If the CFNR was executed, the CDR for the call to the Waiting Queue/Broadcast object ended without release.<br/><!--<br/>pbx.cpp<br/>pbx_wait.cpp<br/>pbx_bc.cpp<br/>--><!---->
      
      
=== myPBX: New icons for unknown phone and im status ===
=== PBX: Web User interface problem with IE and non-ascii PBX names ===


{|
{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=72536 72536]
|[http://mantis.innovaphone.com/view.php?id=79568 79568]
|}
|}
The difference between open, closed and unknown was not understood well.<!---->
For example groups at a user assigned to a PBX with non-ascii characters could not be edited.<br/>The problem is a bug in IE XSL translation which does special handling of href attributes. Same thing with onclick attribute works.<br/><!--<br/>pbx_objs_left.xsl<br/>pbx_objs_right.xsl<br/>--><!---->
      
      
=== SoftwarePhone: Dialing outbound calls temporarily failed ===
=== ip3010 TEL port not working in NT mode ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72628 72628]
|[http://mantis.innovaphone.com/view.php?id=79570 79570]
|}
|}
When there was a held call and a consultation call and the consultation call was released by the remote peer the SoftwarePhone did not accept further outbound calls until the held call had been released. This is fixed now.<!---->
TEL port of ip3010 gateways configured in NT mode do not get Physical Link up indication. This problem applies to V9hotfix7 up to V9hotfix10.<!---->
''Status:''
ip6010.cpp
      
      
=== IP232: Fix for touch keyboard handling ===
=== SIP: Max forward value of 32 could be too small for some provider ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72651 72651]
|[http://mantis.innovaphone.com/view.php?id=79578 79578]
|}
|}
When editing call diversions, one of the on/off controls may render across touch keyboard.<br/>Content of multi-line-edit-control was mis-placed when touch-keyboard was activated.<br/>Display of first matching directory entriy on indirect dialing screen.<!---->
For a starting value of max-forwards a value of 32 was used, because this is the maximum value in H.323. This was too small for some sip providers. Starting value now increased to 64 and on H.323 the half value is transmitted.<br/><!--<br/>h323sig.cpp<br/>signal.cpp<br/>signal.h<br/>--><!---->
      
      
=== ip241 - monitor mode (handset + speaker) did not work in V9hotfix5 ===
=== SIP: Re-negotiation for T38 did not work in media-relay scenarios ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72702 72702]
|[http://mantis.innovaphone.com/view.php?id=79583 79583]
|}
|}
monitor mode (handset + speaker) was displayed but the speaker was mute<!---->
Bad SDP answer was generated.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
''Status:''
Introduced by<br/> #77277: SIP: SDP answer must have the same number of media descriptions as received offer
      
      
=== update - scfg command could hang when the HTTP session was broken or prematurely closed by the server ===
= V9 Hotfix 12 (9061009) =
Changes included in Version 9 hotfix12
[http://mantis.innovaphone.com/view.php?id=79736 Definition]
 
== New Features ==
 
 
   
=== myPBX: Syslog for sessions and application sharing ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72708 72708]
|[http://mantis.innovaphone.com/view.php?id=77289 77289]
|}
|}
in consequence update script processing was stopped until reboot<!---->
Syslog entries should be useful for debugging session timeouts and problems with the external application sharing solution.<br/><!--<br/>pbx_client.cpp<br/>pbx_client.h<br/>log.xsl<br/>--><!---->
      
      
=== Web-UI: Wider input fields for domain names ===
=== SIP: Pass display names of &lt;dialog-info&gt; to phoneapp ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72723 72723]
|[http://mantis.innovaphone.com/view.php?id=79543 79543]
|}
|}
When configuring<br/> - Remote Domain<br/> - Local Domain<br/> - Proxy<br/> - STUN Server<br/>on a Gateway interface there's need for more space.<!---->
Pass display names of <dialog-info> to phoneapp.<br/><br/><!--<br/>fty.cpp/h<br/>sip.cpp<br/>phonesig.cpp<br/>--><!---->
      
      
=== Trap: When Dectmaster registers user at PBX using SIP protocol ===
=== Gateway: Overlap dialing timeout configurable ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72729 72729]
|[http://mantis.innovaphone.com/view.php?id=79639 79639]
|}
|}
When Dectmaster registers user at PBX using SIP protocol<!---->
The default of 4s is not good for all applications. Now configurable between 500ms and 6000ms.<br/><!--<br/>relay_edit_route.xsl<br/>gk.cpp<br/>gk.h<br/>gk_if.h<br/>relay.cpp<br/>relay.h<br/>--><!---->
      
      
=== SIP: Add payload type for RTP-DTMF in case of media-relay ===
=== IP-DECT: OEM Configuration of Cisco Features changed ===


{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72732 72732]
|[http://mantis.innovaphone.com/view.php?id=79684 79684]
|}
|}
Add payload type for RTP-DTMF (telephone-event) to SDP offer in case of media-relay.<br/>Should help on Mobility scenarios.<!---->
should depend on installed license<br/><!--<br/>dectmaster.h<br/>dectmaster_if.h<br/>dectfty.cpp<br/>dectfty.h<br/>dtmffty.cpp<br/>--><!---->
      
      
=== PBX: Called Name displayed when calling an object with forking was wrong ===
=== myPBX launcher: Automatically move main window into visible area of screen ===


{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72735 72735]
|[http://mantis.innovaphone.com/view.php?id=79697 79697]
|}
|}
The name of the forking destination was displayed instead of the name of the called object<!---->
The window might be outside the visible area if the user changed the screen resolution or disconnectes one of the screens.<br/><br/>If the main window is not visible on any screen, it is now moved to the center of the primary screen, when the user clicks the tray icon.<br/><!---->
      
      
=== IP4001/IP6000 -UART driver did not work ===
=== PBX-SOAP: FindUser optional argument 'nohide' ===


{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72743 72743]
|[http://mantis.innovaphone.com/view.php?id=79734 79734]
|}
|}
the hardware requires the registers to be written 32-bit wise which was not respected by a fix one year before. <!---->
Needed if the FindUser is used to look for users which have the 'Hide from LDAP property' set.<!---->
''Status:''
<!--<br/>pbx_xml.cpp<br/>pbx900.wsdl<br/>pbx10_00.wsdl<br/>-->
      
      
=== PBX: No Audio if call thru Waiting Queue DTMF destination, was transfered to BC-Conf ===
=== PBX Mobility: Pick mobile call on fixed phone ===


{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72746 72746]
|[http://mantis.innovaphone.com/view.php?id=79794 79794]
|}
|}
Problem caused by call state management error in PBX for calls connected without alert if alert was received later<!---->
by dialing number of Mobility object on fixed phone<br/><!--<br/>pbx_mobility.cpp<br/>pbx_mobility.h<br/>pbx.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Wrong call state displayed ===
=== Gateway: Allow setting of system time from ISDN time ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72748 72748]
|[http://mantis.innovaphone.com/view.php?id=79889 79889]
|}
|}
"Destination reached" displayed instead of "subscriber busy" if consultation call was rejected.<br/><!---->
In ISDN connect messages a public network is sending a local date and time. By a configuration option this can now be used to update the system time<br/><!--<br/>relay_edit_phys,xsl<br/>gk.cpp<br/>gk.h<br/>gk_if.h<br/>relay.cpp<br/>relay_api.h<br/>--><!---->
      
      
=== PBX Waiting: Ringback missing when using DTMF to dial from one WQ to another which is alerting ===
=== IP-DECT: New Master/Radio behavior for license incompatibility ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72766 72766]
|[http://mantis.innovaphone.com/view.php?id=79914 79914]
|}
|}
This was a collateral damage for a fix for Waiting Queue announcements from a Boolean Object<!---->
The behavior of the DECT Master is changed, if an unlicensed Radio tries to register in. This is only used for an OEM license model.<br/><br/><!--<br/>Changed files:<br/>dectmaster.h<br/>dectmaster.cpp<br/>dectradio.h<br/>dectradio.cpp<br/>asc_dectradio.xsl<br/>asc_dectmaster_radios.xsl<br/>--><!---->
      
      
=== IP222,IP232: Long key press on numeric block while being offhook does not open directory search ===
=== PBX: Better handling of presence information without activity ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72831 72831]
|[http://mantis.innovaphone.com/view.php?id=80006 80006]
|}
|}
Long key press during offhook is expected to open directory search.<!---->
This type of presence information could contain a note which refers to the future<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== IP232: New touch control in menu header for screen exit ===
=== RPCAP trace: Indicate transmit/received packets by setting the remote mac address to 00-90-33-00-00-00 ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72833 72833]
|[http://mantis.innovaphone.com/view.php?id=80152 80152]
|}
|}
Re-design<!---->
When reading a trace it is currently not obvious if a packet is sent or received, we need to find out the devices IP address, e.g. by reading the config. If the devices on MAC adress is used a source only if a packet is sent and as destination only if a packet is received this process is simplified.<br/><br/>This is now done for UDP/TCP traces!<!---->
      
      
=== IP241,IP222,IP232: Support for unicode 0x308 (diaeresis/umlaut) ===
=== SIP: Support for Alert-info:&lt;Bellcore-dr1&gt; ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72859 72859]
|[http://mantis.innovaphone.com/view.php?id=80174 80174]
|}
|}
When ä,ö,ü are encoded as a,o,u followed by diaeresis from unicode block 'Combining Diacritical Marks' only a,o,u where displayed.<br/>Now ä,ö,ü are displayed.<!---->
Proprietary tagging of internal calls<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Support for hebrew and arabic presence notes ===
=== phone: ip222,ip232: more USB headsets supported ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72905 72905]
|[http://mantis.innovaphone.com/view.php?id=80224 80224]
|}
|}
Hebrew and arabic text passages are rendered frmo right to left.<!---->
for a complete list see http://wiki.innovaphone.com/index.php?title=Reference9:Concept_USB_Headset<br/><br/><!--<br/>keys_ip222.cpp<br/>--><!---->
''Status:''
checked in to 10.00, 9.00
      
      
=== PBX: Conference no media ===
=== Phones: Switch for phoneapp to disable auto-answer ===


{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=72915 72915]
|[http://mantis.innovaphone.com/view.php?id=80233 80233]
|}
|}
There is sometimes no media for a conference member. This is fixed, now.<!---->
Disable/enable auto-answer support on phoneapp level.<br/><br/><!--<br/>phonesig_if.h<br/>phonesig.cpp<br/>--><!---->
      
      
=== SIP: Memory leak during transfer ===
=== PBX: DECT attributes for DECT security ===


{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73003 73003]
|[http://mantis.innovaphone.com/view.php?id=80300 80300]
|}
|}
Occured on internal testing only (002-conf-with-bcast.xml)<!---->
In preparation for the new DECT feature DECT security there will be new attributes for the endpoint data which must be taken over. With this fix the innovaphone PBX supports the new attributes if the user is edited.<br/><br/><!--<br/>Changed files:<br/>pbx_admin.cpp<br/>pbx_edit_object.xsl<br/>--><br/><!---->
      
      
=== IP241,IP222,IP232: Update for CFU indication on idle screen ===
=== phone: ip222, ip232: Jabra  USB Headset feature "Reject incoming call" supported now ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73013 73013]
|[http://mantis.innovaphone.com/view.php?id=80401 80401]
|}
|}
Not displaying "cfu:241" on header of idle screen<br/>but "Diverted to 241"<br/>or "Umgeleitet zu 241"<br/>or "Renvoyé sur 241"<br/>etc.<!---->
A double tap on talk button rejects a ringing call with 'busy'<br/><br/><!--<br/>keys_ip222.cpp<br/>--><!---->
      
      
=== myPBX: Simplify adding multiple users to a chat session ===
=== IP-DECT: Signal waiting calls to handset ===


{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73016 73016]
|[http://mantis.innovaphone.com/view.php?id=80408 80408]
|}
|}
Do not unselect the chat session, after a person has been added. Replace "start chat" buttons by "add to chat" buttons when a chat session is selected.<!---->
New event to the DECT system to add a waiting call to the call list.<br/>This is step two of the implementation and related to the case #79349.<br/><br/><!--<br/>Changed files:<br/>fty.h<br/>fty.cpp<br/>dectmaster.cpp<br/>dectradio_call.cpp<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp<br/>h450.h<br/>h450.cpp<br/>h450asn1.h<br/>h450asn1.cpp<br/>--><br/><!---->
      
      
=== RTP-DTMF: Digit may get lost during media re-negotiation ===
=== IP-DECT: Anonymous endpoint information ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73037 73037]
|[http://mantis.innovaphone.com/view.php?id=80421 80421]
|}
|}
Receiver starts handling on END event.<br/>Sender may stop sending before END event was sent.<!---->
Information about anonymous endpoint is added to the event to the DECT system. In preparation for the new feature "easy subscription".<br/><br/><!--<br/>Changed files:<br/>signal.h<br/>signal.cpp<br/>dectradio.cpp<br/>--><!---->
      
      
=== myPBX: Do not show bubble for outgoing chat calls and messages ===
=== IP-DECT: DECT security ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73039 73039]
|[http://mantis.innovaphone.com/view.php?id=80424 80424]
|}
|}
Notifications are only needed for incoming calls and messages.<!---->
Some changes in preparation for the new feature "DECT security". This feature will be available later.<br/><br/><!--<br/>Changed files:<br/>signal.h<br/>signal.cpp<br/>dummysig.cpp<br/>dectusers_if.h<br/>dectusers.h<br/>dectusers.cpp<br/>dectmaster.h<br/>dectmaster.cpp<br/>dectradio.h<br/>dectradio.cpp<br/>dectlocalusers.h<br/>dectlocalusers.cpp<br/>dectmobmaster.h<br/>dectmobmaster.cpp<br/>asc_dectusers.xsl<br/>asc_dectradio.xsl<br/>asc_dect_edit_user.xsl<br/>--><!---->
      
      
=== myPBX: Skript errors in chat window ===
=== Phone: Accept MWI from Exchange Server ===


{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73040 73040]
|[http://mantis.innovaphone.com/view.php?id=80446 80446]
|}
|}
Closing the chat window caused different script errors in Internet Explorers and Firefox.<!---->
MS Exchange Server sends unsolicited NOTIFY(message-summary) to served user<br/>with served user's number as destination and origin.<br/>But phones expect to receive MWI message center number as origin.<br/>MWI fkey would not light up.<br/><br/><!--<br/>app_reg.cpp<br/>--> <!---->
      
      
=== LDAP replication from v7 could stop ===
=== SIP: New config option /take-zero-addr-for-hold ===


{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73052 73052]
|[http://mantis.innovaphone.com/view.php?id=80516 80516]
|}
|}
LDAP replication from v7 could stop<!---->
From now on "c=IN IP4 0.0.0.0" is no longer accepted as hold signaling.<br/>Config option /take-zero-addr-for-hold is introduced to get back old handling.<br/><br/><!--<br/>sip.cpp<br/>sip.h<br/>--><!---->
      
      
=== SIP: Don't write SRTP key into T.38 part of media description ===
=== IP-DECT: User log in, endpoint data added ===


{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73112 73112]
|[http://mantis.innovaphone.com/view.php?id=80620 80620]
|}
|}
v=0<br/>o=- 1295 1 IN IP4 10.120.55.3<br/>s=-<br/>c=IN IP4 10.120.55.6<br/>t=0 0<br/>m=audio 20026 RTP/SAVP 8 101 13<br/>a=rtpmap:101 telephone-event/8000<br/>a=fmtp:101 0-15<br/>a=ptime:20<br/>a=silenceSupp:off - - - -<br/>a=inactive<br/>a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9xHxSg836505XOXwdIHfQ8Cm2ZYezNPpjvHNPCvb<br/>m=image 0 udptl t38<br/>a=T38FaxVersion:0<br/>a=T38MaxBitRate:9600<br/>a=T38FaxFillBitRemoval:0<br/>a=T38FaxTranscodingMMR:0<br/>a=T38FaxTranscodingJBIG:0<br/>a=T38FaxRateManagement:transferredTCF<br/>a=T38FaxUdpEC:t38UDPRedundancy<br/>a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9xHxSg836505XOXwdIHfQ8Cm2ZYezNPpjvHNPCvb<br/><!---->
During the user log in with a change of the IP-DECT Master the endpoint data (product id/software version) is sent to this new one now.<br/><br/><!--<br/>Changed files:<br/>dectlocalusers.cpp<br/>dectradio.cpp<br/>--><!---->
      
      
=== PBX External UC: Support of multisite setup ===
=== SIP: Pass display information to application ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73115 73115]
|[http://mantis.innovaphone.com/view.php?id=80632 80632]
|}
|}
This requires all calls to the External UC to be routed to one interface. Before this fix the calls were routed within the local PBX only.<br/><br/>Now a Name (not Long Name as before) has to be configured for the External UC interface.<!---->
Pass display information received in Call-Info header in 200/OK for BYE to app.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== IP241,IP222,IP232: Show number of missed calls in status bar ===
=== IP-DECT: Idle display update ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73140 73140]
|[http://mantis.innovaphone.com/view.php?id=80654 80654]
|}
|}
New extended symbol is more eye-catching.<!---->
Handling of idle display update message in the call release message is added. Used with an OEM PBX.<br/><br/><!--<br/>Changed file:<br/>dtmffty.cpp<br/>--><!---->
      
      
=== PBX: Called party number wrong at original called phone with forking ===
=== myPBX launcher: Configurable hotkey action ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73145 73145]
|[http://mantis.innovaphone.com/view.php?id=80684 80684]
|}
|}
If the forking destination and the original called phone are assigned to different nodes<!---->
Options are:<br/>- Copy selected phone number to myPBX<br/>- Show myPBX<br/><!--<br/>Workaround for #76582<br/>ctiwin.cs<br/>Config.resx<br/>Config.Designer.cs<br/>Config.cs<br/>--><br/><!---->
      
      
=== Media Negotiation: v9 XPARENT not compatible to v8 or earlier XPARENT ===
=== myPBX launcher: Autocomplete configured URLs ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73153 73153]
|[http://mantis.innovaphone.com/view.php?id=80689 80689]
|}
|}
In v9 a dynamic payload type is used for XPARENT to be compatible to SIP, whereas in v8 an earlier payload type 0 was used. Within the media negotiation this should be detected and switched back to payload type 0.<!---->
If the user configures just an IP address, it shall be replaced by the full myPBX URL.<br/><br/>For example 192.168.0.10 will be replaced by http://192.168.0.10/PBX0/MY/client.htm.<br/><!--<br/>Config.cs<br/>--><!---->
   
== Bug Fixes ==
 
 
      
      
=== Phones: Update of phone text data ===
=== PBX: CFNR at Gateway object with incomplete destination ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73211 73211]
|[http://mantis.innovaphone.com/view.php?id=79605 79605]
|}
|}
Some translations missing (Polski,Eesti).<br/>Support for language typical letters (Polski).<br/>Some abbreviation removed (for color phones).<!---->
A CFNR at a gateway object is executed if there is no registration. Any additional digits dialed should be added to the CFNR destination. This did not work if the original CFNR destination was incomplete and only completed with additional digits dialed.<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== PBX: Conference GUI update with IE ===
=== PBX Waiting: Filter did not work anymore for CFNR ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73256 73256]
|[http://mantis.innovaphone.com/view.php?id=79610 79610]
|}
|}
The PBX broadcast conference object window is not closed if IE is used. This fixed now.<!---->
Collateral damage from<br/><br/>fix: #75465: PBX: CFNR Loop check detected loops that weren't<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== SIP: support for "Content-Type: multipart/alternative" ===
=== LDAP Server caused crash on port-scanner attack ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73260 73260]
|[http://mantis.innovaphone.com/view.php?id=79634 79634]
|}
|}
Lync interoperability<!---->
LDAP Server caused crash on port-scanner attack<br/><!--<br/>ldapsrv.cpp<br/>ldapsrv.h<br/>--><!---->
      
      
=== IP2x2: Keyboard handling on call reroute ===
=== myPBX: Quotes in connected names caused java script errors ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73268 73268]
|[http://mantis.innovaphone.com/view.php?id=79636 79636]
|}
|}
Redial key launches "Transfer" screen.<br/>After entering the destination number, Redial key should cause execution of call reroute.<!---->
When the connected name of a call contained the ' character there was a script error.<br/><!--<br/>client.js<br/>client_dtmf.js<br/>client_util.js<br/>--><!---->
      
      
=== PBX Waiting: When using a waiting queue for outgoing dialing, the announcement was cut off at the beginning ===
=== PBX: Status displayed on boolean function key could be wrong ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73274 73274]
|[http://mantis.innovaphone.com/view.php?id=79643 79643]
|}
|}
This was because the announcement was started already when the call was sent out and not when the call was connected<!---->
In case of severe network problems, it could happen that the status displayed on a Boolean function key was wrong and was only corrected when the boolean status changed.<br/><!--<br/>pbx_bool.cpp<br/>pbx_gi.cpp<br/>--><!---->
      
      
=== myPBX: Allow spaces in last LDAP attribute ===
=== DHCP Server Identifier was cleared after editing the DHCP-Server page ===


{|
{|
Line 8,055: Line 7,959:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73276 73276]
|[http://mantis.innovaphone.com/view.php?id=79651 79651]
|}
|}
Currently "j doe innovaphone" finds all internal users starting with j. It should only find internal users that start with "j doe innovaphone".<br/><!---->
The value of "IP4/ETHx/DHCP/Server Identifier" was cleared when the OK or Renew button was pressed on the "IP4/ETHx/DHCP-Server" page. This bug was introduced with V9hotfix5.<br/><br/><!-- <br/>dhcp_server.xsl<br/>--><br/><br/><!---->
      
      
=== IP232: Cursor positioning when touching ip address control ===
=== SIP: Wrong Contact-URI in outgoing NOTIFY ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73407 73407]
|[http://mantis.innovaphone.com/view.php?id=79700 79700]
|}
|}
Make sure input cursor is at right side of touched octet.<!---->
Contact-URI should match the Request-URI of the SUBSCRIBE.<br/>Also the Message-Account URI in "simple-message-summary" was wrong as result of the wrong Contact-URI.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== SIP: Configuration of SIP response code mappings did not work on IPxx10 and IPVA ===
=== Config: Could not dynamically set or reset /trace on the LICENSE module ===


{|
{|
Line 8,077: Line 7,981:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73453 73453]
|[http://mantis.innovaphone.com/view.php?id=79709 79709]
|}
|}
http://wiki.innovaphone.com/index.php?title=Howto:Tweak_SIP_Response_Code_to_Q.931_Cause_Code_Mapping<!---->
Missing handling of MODULE_UPDATE.<br/><br/><!--<br/>license.cpp<br/>--><!---->
      
      
=== PBX: Reporting licenses counting could be wrong if config templates were used ===
=== PBX: Allow call completion only if recall is not prohibited by CFU ===


{|
{|
Line 8,088: Line 7,992:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73466 73466]
|[http://mantis.innovaphone.com/view.php?id=79739 79739]
|}
|}
It could happen that a reporting license was used up by a config template object, if reporting was checked and other config templates where referenced<!---->
A CFU loop results in a rejection with busy. A subsequent call completion attempt was allowed and a recall possible was signaled right away. This was very confusing.<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== myPBX: Allow overriding implicit allows ===
=== SIP: Handling of Call Pickup ===


{|
{|
Line 8,099: Line 8,003:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73471 73471]
|[http://mantis.innovaphone.com/view.php?id=79741 79741]
|}
|}
It was not possible to define an allow with the same name. This is needed to override allows that are inherited from templates or group memberships.<!---->
Failed to handle INVITE with Replaces as Call Pickup.<br/>Must redirect Pickup call to alerting party.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== IP241,IP222,IP232: Trap when scrolling down long call list ===
=== myPBX: Problem with checking browser capabilities in Firefox ===


{|
{|
Line 8,110: Line 8,014:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73494 73494]
|[http://mantis.innovaphone.com/view.php?id=79757 79757]
|}
|}
Out of memory trap when scrolling down long call list.<!---->
When checking the supported browser features, Firefox thows an uncaught exception if cookies are deactivatd by the user. Therefore myPBX is stuck in the "Loading" screen instead of displaying a configuration hint.<br/><!--<br/>client.js<br/>client_util.js<br/>--><!---->
      
      
=== Trap: When using Diagnostics/Ping ===
=== IP241,IP222,IP232: Show lengthy number information on Partner fkey ===


{|
{|
Line 8,121: Line 8,025:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73536 73536]
|[http://mantis.innovaphone.com/view.php?id=79779 79779]
|}
|}
When leaving Diagnostics/Ping or Diagnostics/Traceroute web screen a trap may occur.<!---->
Toggle between 'show head' and 'show tail' presentation.<br/><br/><!--<br/>forms_color.cpp/h<br/>--><!---->
      
      
=== PBX: Navigating in locations and groups if non-ascii charaters were used, failed with internet explorer ===
=== SIP: DNS resolving of STUN server failed ===


{|
{|
Line 8,132: Line 8,036:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73567 73567]
|[http://mantis.innovaphone.com/view.php?id=79788 79788]
|}
|}
Internet explorer buggy concerning url encoding of href attributes<!---->
Wrong STUN server port used (5060 instead of real port).<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== IPVA, crash-dump without backtrace, if  tracing was switched off internally ===
=== Fax: Channel memory leak ===


{|
{|
Line 8,143: Line 8,047:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73574 73574]
|[http://mantis.innovaphone.com/view.php?id=79815 79815]
|}
|}
IPVA, crash-dump without backtrace, if  tracing was switched off internally<!---->
Memory leak is fixed in fax channel now.<br/><br/><!--<br/>Changed file:<br/>t30.cpp<br/>--><!---->
      
      
=== ISDN: Do not provide tones and no signaling of inband info for unrestricted digital information calls ===
=== PBX: Export to CSV, utf-8 byte order mark was missing ===


{|
{|
Line 8,154: Line 8,058:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73603 73603]
|[http://mantis.innovaphone.com/view.php?id=79844 79844]
|}
|}
There is equipment, which is doing unrestricted digital information ISDN calls, which gets confused if there is an ALERT message indicating inband tones (ringback).<!---->
Without the utf-8 BOM the file was not correctly interpreted as utf-8 by Microsoft Excel and other applications.<br/><!--<br/>pbx_admin.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Symbol "new messages" and symbol "headset" do overlap in status bar ===
=== PBX: For registrations containing name and number, use name to identify the device ===


{|
{|
Line 8,165: Line 8,069:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73608 73608]
|[http://mantis.innovaphone.com/view.php?id=79846 79846]
|}
|}
Moved "headset" symbol to right side of status bar.<!---->
It did depend on the sequence of the name and the number. If the number was first, the first device was selected and the name was ignored. Now the name is used to select the device regardless of sequence.<br/><br/>This is a problem with endpoints which always send name and number for registration.<br/><br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== IP-DECT: DECT endpoint update for multicast identities ===
=== SIP: CLEARMODE does not work since Hotfix10 ===


{|
{|
Line 8,176: Line 8,080:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73613 73613]
|[http://mantis.innovaphone.com/view.php?id=79858 79858]
|}
|}
The update information to DECT is changed if the user logs out a handset. This is used to update the multicast identities in some OEM handsets.<!---->
Invalid SDP answer is send:<br/><br/>    v=0<br/>    o=- 4 3 IN IP4 10.28.108.8<br/>    s=-<br/>    c=IN IP4 0.0.0.0<br/>    t=0 0<br/>    m=audio 0 RTP/AVP 125 101<br/>    a=rtpmap:125 CLEARMODE/8000<br/>    a=rtpmap:101 telephone-event/8000<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== Ip6010 DSP Allow calls to start with T.38 ===
=== memory violation in http client when processing an URI &gt; 8000 byte ===


{|
{|
Line 8,187: Line 8,091:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73626 73626]
|[http://mantis.innovaphone.com/view.php?id=79860 79860]
|}
|}
Needed for some T38 fax applications<!---->
happens when the digest authentication header is constructed <br/><br/><!--<br/>httpclient_i.cpp<br/>--><br/><br/><br/>  <!---->
''Status:''
''Status:''
ac_dsp3.cpp
checked in to 10.00, 9.00
      
      
=== IP241,IP222,IP232: Missing "Recall possible" display text on call completion ===
=== PBX Broadcast: Forking/Mobility configured at Broadcast object did not work ===


{|
{|
Line 8,200: Line 8,104:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73628 73628]
|[http://mantis.innovaphone.com/view.php?id=79863 79863]
|}
|}
Missing display text on 'recall possible' notification.<!---->
Mobility can be used now for forking with delay<br/><!--<br/>pbx_bc.cpp<br/>pbx_bc.h<br/>pbx.cpp<br/>--><!---->
      
      
=== SIP: Media negotiation problem on outbound call from BC Conference object ===
=== Potential trap in packet libraray ===


{|
{|
Line 8,211: Line 8,115:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73652 73652]
|[http://mantis.innovaphone.com/view.php?id=79866 79866]
|}
|}
INVITE was sent without SDP offer.<br/>test/9.00/pbx/conf/SIP/002-conf-with-bcast.xml<!---->
Ther was a not obvious size limit for packet::put_head and packet::put_tail, which caused a trap if the size was exceeded. This could happen if a CDR exceeded a certain size.<br/><!--<br/>packet.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Extra digits dialed when using RIGHT key on screnn 'indrect dialling' ===
=== SIP: Domain Name System (DNS) names compared case sensitive ===


{|
{|
Line 8,222: Line 8,126:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73694 73694]
|[http://mantis.innovaphone.com/view.php?id=79887 79887]
|}
|}
'123123' instead of '123' was dialed.<!---->
Domain Name System (DNS) names are "case insensitive"<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== ISDN Trunk: Transfer to ISDN Trunk with TONE interface failed ===
=== SIP: Interworking with "AUDC-IPPhone" ===


{|
{|
Line 8,233: Line 8,137:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73695 73695]
|[http://mantis.innovaphone.com/view.php?id=79892 79892]
|}
|}
There was not media after the transfer<!---->
RFC-3551 4.5.2<br/>  Even though the actual sampling rate for G.722 audio is 16,000 Hz,<br/>  the RTP clock rate for the G722 payload format is 8,000 Hz because<br/>  that value was erroneously assigned in RFC 1890 and must remain<br/>  unchanged for backward compatibility.  The octet rate or sample-pair<br/>  rate is 8,000 Hz.<br/><br/>Codec description must be: a=rtpmap:9 g722/8000<br/>but "AUDC-IPPhone" sends: a=rtpmap:9 G722/16000<br/>in SDP offer.<br/><br/><!--<br/>sdpmsg.cpp<br/>--><!---->
      
      
=== H.323: Renegotiation to SRTP after dialtone failed ===
=== SIP: Problems with DNS resolving of proxy adresses ===


{|
{|
Line 8,244: Line 8,148:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73698 73698]
|[http://mantis.innovaphone.com/view.php?id=79907 79907]
|}
|}
Happens with Escape Dialtones configured in PBX and calls with SRTP enabled<!---->
If resolving of the primary proxy failes, the backup proxy is never resolved.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== Option added to wait  for the DSP to disconnect fax, needed only for some fax devices. ===
=== IP-DECT: Reset required if Radio password changed ===


{|
{|
Line 8,255: Line 8,159:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73748 73748]
|[http://mantis.innovaphone.com/view.php?id=79929 79929]
|}
|}
Use the prodedure of <br/>Mantis 67821: ip22/24/28/302/305: DSP affected by certain fax tones<br/>only if the option /t38-wait-fax-close set.<br/><br/>This avoids traps caused by disconnect timeouts if the DSP response is not received in time.<br/><!---->
Now reset required is shown if the Radio password for the Master registration is changed.<br/><br/><!--<br/>Changed file:<br/>dectradio.cpp<br/>--><!---->
''Status:''
ac_dsp3.cpp<br/>ac_dsp3.h<br/>dsp.xsl
      
      
=== SIP: Using wrong remote port when registering ===
=== Phone: PBX (operator) initiated outbound call was connected but mute when another call was ringing on phone already ===


{|
{|
Line 8,268: Line 8,170:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73784 73784]
|[http://mantis.innovaphone.com/view.php?id=79943 79943]
|}
|}
Only affects IP-DECT when handset is switched OFF and ON and if the SIP runs on non-standard port.<!---->
the phone was not switched from ring mode to handsfree/headset mode <br/><br/><!--<br/>phonesig.cpp<br/>--><!---->
      
      
=== IP232: Memory leak in display rendering ===
=== SIP: Interworking with HD audio client ===


{|
{|
Line 8,279: Line 8,181:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73864 73864]
|[http://mantis.innovaphone.com/view.php?id=79953 79953]
|}
|}
Memory leak in display rendering<!---->
Support for unknown codecs.<br/>Passing transparentley.<br/><br/><!--<br/>channel.cpp/h<br/>sdpmsg.cpp/h<br/>sip.cpp<br/>--><!---->
      
      
= V9 Hotfix 7 (90600.08) =
=== Phone: Out-Of-Memory-Trap ===
Changes included in Version 9 hotfix7
[http://mantis.innovaphone.com/view.php?id=75932 Definition]
 
== New Features ==
 
 
   
=== IP241,IP222,IP232: Some diagnostics to check phone's ui memory consumption ===


{|
{|
Line 8,298: Line 8,192:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74049 74049]
|[http://mantis.innovaphone.com/view.php?id=79980 79980]
|}
|}
Where have all the memory gone.<!---->
If LOG server is configured but not reachable<br/>the device will buffer arising LOG entries until a limit of 300kB.<br/>This limit was to high for old black/white telephones.<br/>Now not more then 1% of DRAM size is used for LOG buffer.<br/><br/><!--<br/>logging.cpp<br/>--><!---->
      
      
=== SIP: Support for "Content-ID" in "multipart/alternative" bodies ===
=== PBX: When editing a Node object it was changed to a PBX object ===


{|
{|
Line 8,309: Line 8,203:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74161 74161]
|[http://mantis.innovaphone.com/view.php?id=79982 79982]
|}
|}
Support for "Content-ID" in "multipart/alternative" bodies<!---->
This was a collateral damage of fix: #78878: PBX: When doing a show users, unnecessary data was sent to the browser<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Added drop shadow to lower end of menu screens ===
=== ISDN: Problem configuring negative volume levels ===


{|
{|
Line 8,320: Line 8,214:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74278 74278]
|[http://mantis.innovaphone.com/view.php?id=79992 79992]
|}
|}
To make end of list of menu entries more discernible.<!---->
Collateral Damage from<br/>79028: IP-DECT: Feature Code for calling predefined service URIs<br/><!--<br/>config_options.cpp<br/>dect_comcerto/platform/config.h<br/>--><!---->
      
      
=== myPBX: Support static URLs for application sharing ===
=== Dect: AD replication makes IPEI disappearing on all users ===


{|
{|
Line 8,331: Line 8,225:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74389 74389]
|[http://mantis.innovaphone.com/view.php?id=80015 80015]
|}
|}
In some application sharing solutions a fixed link can be used to create and join meetings (GoMeetNow, BeamYourScreen).<br/><br/>There are two URLs configured, one for the presenter and one for attendees. When the user clicks the aplication sharing button the links are sent using chat messages.<br/><!---->
Side effect of a previous fix #72672 aiming at ad-replicated and deleted objects coming back to life.<br/><!--<br/>flashdir.cpp<br/>--><!---->
      
      
=== PBX: Tracing flag turns on tracing in all dyn PBX's as well ===
=== myPBX: Setting CFx with boolean did not work in IE8 ===


{|
{|
Line 8,342: Line 8,236:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74390 74390]
|[http://mantis.innovaphone.com/view.php?id=80018 80018]
|}
|}
Helpfull to debug dyn PBX setups<!---->
Using Internet Explorer 8 the selected Boolean object was not saved.<br/>Internet Explorer 8 requires option tags to have a value attribute.<br/><!--<br/>client_diversions.js<br/>--><!---->
      
      
=== debug method raw_ethernet ===
=== IP-DECT: Handover with IP1202 ===


{|
{|
Line 8,353: Line 8,247:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74897 74897]
|[http://mantis.innovaphone.com/view.php?id=80022 80022]
|}
|}
A new debug method to trace raw ethernet data.<!---->
Only one handover is possible since the last fix with the IP1202. It is fixed now.<br/><br/><!--<br/>Changed files:<br/>dectradio.cpp<br/>dectradio_call.cpp<br/>dectradio_dectcall.cpp<br/>--><!---->
      
      
=== Support for new hardware ===
=== Name and Number configured for a registration must be completely replaced by registration result ===


{|
{|
Line 8,364: Line 8,258:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74990 74990]
|[http://mantis.innovaphone.com/view.php?id=80039 80039]
|}
|}
.<!---->
If for example Number and Name is configured but the registration result provides the Number only, the configured Name must not be used in further signaling operations (diversion queries ...) <br/><br/><!--<br/>phonesig.cpp<br/>--><!---->
''Status:''
checked in to 10.00, 9.00
      
      
=== possible use of push-pull drivers for new hardware ===
=== PBX: Possible trap on calls from misconfigured nodes/PBXs (node parent loop) ===


{|
{|
Line 8,375: Line 8,271:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75051 75051]
|[http://mantis.innovaphone.com/view.php?id=80093 80093]
|}
|}
not visible to customers<!---->
If a node or PBX is configured with a parent node configured to itself in the most simple case, a call from an endpoint configured for this node to a destination which cannot be found in this node, will cause a trap.<br/><br/>This is a collateral damage of fix: #79317: PBX: Local objects could not be called from Nodes with escapes as expected<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>--><!---->
      
      
=== PBX/IP-DECT: User password length ===
=== IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305 IP222 IP232 IP241: receveid RTP packets limited to 480bytes ===


{|
{|
Line 8,386: Line 8,282:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75453 75453]
|[http://mantis.innovaphone.com/view.php?id=80111 80111]
|}
|}
The maximum user password length in the PBX and IP-DECT is increased to 23.<!---->
increased to 640 Bytes to allow G711 80ms.<br/><br/><!-- ac_phone3.h  ac_dsp3.h --><br/><!---->
      
      
=== H.323: Support for switch back to local media on endpoint ===
=== DHCP Survivability mode doesn't work after a DHCP restart ===


{|
{|
Line 8,397: Line 8,293:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75522 75522]
|[http://mantis.innovaphone.com/view.php?id=80112 80112]
|}
|}
needed to resume normal call after a conference using external conference unit<!---->
``Survivability´´ mode is used by WLAN phones. In this mode the settings of the 'saved lease' (IP addr, mask, ...) are used until a fresh lease is received. The 'saved lease' is the last lease received from a server, it is kept over a reboot.<br/><br/>When WLAN coverage is lost for a while and then regained a DHCP restart is requested to get a fresh lease from a server in a possibly different network. If this happened while using the 'saved lease' the phone lost it's (saved) IP address.  <!---->
''Status:''
checked in to 10.00, 9.00
      
      
=== IP232: Touch gesture to open directory-search/indirect-dialing screen ===
=== AC-DSP3: Switch trace off if the DSP Host interface shows an error ===


{|
{|
Line 8,408: Line 8,306:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75549 75549]
|[http://mantis.innovaphone.com/view.php?id=80130 80130]
|}
|}
Vertical upwards on idle screen opens directory-search/indirect-dialing screen.<br/>Vertical downwards on idle screen opens main-menu screen.<!---->
<!-- ac_dsp3.cpp --><!---->
      
      
=== PBX-SOAP: Wsdl Versioning mechanism fixed ===
=== SIP: Keep backup registration while calls are active ===


{|
{|
Line 8,419: Line 8,317:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75552 75552]
|[http://mantis.innovaphone.com/view.php?id=80137 80137]
|}
|}
Better mechanism, to avoid constantly adding arguments to the Version function with new Versions.<!---->
Terminating backup registration will active calls (through backup system) get disconnected.<!---->
      
      
=== different port naming convention for new hardware ===
=== SIP: Content of Allow-Events header must be treated case-insensitive ===


{|
{|
Line 8,430: Line 8,328:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75575 75575]
|[http://mantis.innovaphone.com/view.php?id=80173 80173]
|}
|}
BRI1..5 instead of TEL1..4+PPP<!---->
Event names are case insensitive.<br/><br/><!--<br/>sipmsg.cpp<br/>--><!---->
      
      
=== PBX-SOAP: UserHold without MOH to local User ===
=== IP-DECT/Analog Features: Call Park should be done on the last active call ===


{|
{|
Line 8,441: Line 8,339:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75577 75577]
|[http://mantis.innovaphone.com/view.php?id=80209 80209]
|}
|}
UserHold was sending MOH to the local and the remote User. With the argument remote=true, the MOH is sent to the remote user only<!---->
Szenario is an active call, then a waiting call comes in, which is accepted, then call park is executed. This call park should be done on the accepted waiting call and not the original.<br/><!--<br/>dtmffty.cpp<br/>--><!---->
      
      
=== myPBX launcher: Use default configuration from local machine registry hive ===
=== IP-DECT/Analog Features: Call completion ===


{|
{|
Line 8,452: Line 8,350:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75634 75634]
|[http://mantis.innovaphone.com/view.php?id=80212 80212]
|}
|}
If no user configuration is given, the following values are copied from the local machine hive.<br/><br/>"Software\\innovaphone\\myPBX\\URL"<br/>"Software\\innovaphone\\myPBX\\secondaryURL"<br/><br/>Also the autostart checkmark is disabled, if autostart is activated, globally. Autostart is controlled with the following registry key.<br/><br/>"Software\\Microsoft\\Windows\\CurrentVersion\\Run\\innovaphone myPBX"<!---->
The type of the call is changed back to normal state if the call completion is executed, and facility conversion is added for the call completion state. <br/>This fixes the reusing of features for a call completion callback call, used if IP-DECT/analog features are enabled. This also fixes missed remote hold and retrieve events to the gatekeeper.<br/><br/><!--<br/>Changed file:<br/>dtmffty.cpp<br/>--><!---->
      
      
=== SIP: support for proprietary message header ===
=== SIP: Handling of 488 for encrypted media ===


{|
{|
Line 8,463: Line 8,361:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75864 75864]
|[http://mantis.innovaphone.com/view.php?id=80246 80246]
|}
|}
Support for proprietary message header (CAL)<!---->
If SRTP offer is rejected by remote endpoint with "SIP/2.0 488 Not Acceptable Here"<br/>we should re-try offer without encryption keys.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== H.323: Call to SIP Trunk with MediaRelay and exclusive coder renegotiation from TONE to early media did not work ===
=== Phones: Show 'tel' presence on configuration screen ===


{|
{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=75959 75959]
|[http://mantis.innovaphone.com/view.php?id=80274 80274]
|}
|}
No ringback and no audio after connect<!---->
Show latest presence on 'Presence' fkey,<br/>but show 'tel' presence on phone config menu.<br/><br/><!--<br/>app_ctl.h<br/>app_gui.cpp<br/>app_reg.cpp<br/>phonesig_if.h<br/>phonesig.cpp<br/>--><!---->
      
      
=== SIP: Added Call-Info header to re-INVITE for hold/retrieve ===
=== PBX-Waiting: Calls to a Waiting queue object in altering state could not be cleared with SOAP ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76011 76011]
|[http://mantis.innovaphone.com/view.php?id=80291 80291]
|}
|}
Call-Info: <urn:X-cisco-remotecc:hold><br/>Call-Info: <urn:X-cisco-remotecc:resume><!---->
A SOAP UserClear issued for the call monitored at the Waiting Queue object didn't do anything<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
      
      
=== SIP: Support for active call pickup acc. to RFC-3891 ===
=== Phones: Sometime fkey could not be deleted ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76035 76035]
|[http://mantis.innovaphone.com/view.php?id=80389 80389]
|}
|}
Sending INVITE with Replaces header to do pickup at SIP PBX's<!---->
Sometime fkey could not be deleted when using the fast edit mode (long fkey press).<br/><br/><!--<br/>phone_user_func.cpp<br/>phone_config.cpp<br/>--><!---->
      
      
=== phone ip222, ip232 : USB headset support (beta) ===
=== IP-DECT: Handset display ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=76463 76463]
|[http://mantis.innovaphone.com/view.php?id=80402 80402]
|}
|}
general support for USB headsets added, some headsets are tested and working, others will follow soon <!---->
Names with special characters are not correctly shown in the handsets. This is fixed in the IP1200 now.<br/><br/><!--<br/>Changed file:<br/>dectdisplay.cpp<br/>--><br/><!---->
      
      
=== phone - Headset Function Key can be configured now as enable/disable or as call control key ===
=== IP210 IP230 IP240: Handset receiver volume increased, especially at lower frequencies ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76468 76468]
|[http://mantis.innovaphone.com/view.php?id=80403 80403]
|}
|}
a headset key mode can be configured to use the key either to enable/disable the headset (Mode: Enable) or to start/accept/clear calls via headset (Mode: Control)<!---->
<br/>For better performance with jinlida receiver.<br/><br/><!-- inca_afe.cpp --><!---->
      
      
== Bug Fixes ==
=== Gateway: Pass through ctSetup facility ===
 
 
   
=== Ip6010 DSP Allow coder change from T38 to voice and back to T38 with local DSP ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=73193 73193]
|[http://mantis.innovaphone.com/view.php?id=80437 80437]
|}
|}
..<!---->
Lync sends diverting party information inside Referred-By header.<br/>Referred-By is interworked to ctSetup facility.<br/>ctSetup facility needs to be passed through by Gateway application.<br/><br/><!--<br/>relay.cpp/h<br/>--><!---->
''Status:''
ac_dsp3.cpp<br/>
      
      
=== ISDN: Enable fax detection only after connect ===
=== Linux: IP address with external DHCP server ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73213 73213]
|[http://mantis.innovaphone.com/view.php?id=80471 80471]
|}
|}
On ISDN networks it can happen that the Connect message is delayed. This way fax tones are forwarded to the caller before the caller has received this Connect.<br/><br/>This way a renegotiation on voip to fax could happen before the connect, which is not supported by sip.<!---->
Assigning a IP address to Linux by a external DHCP server is not working if the network interface which is used is configured with a fix IP address (DHCP disabled). This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dhcp.cpp<br/>--><!---->
      
      
=== Phones: Update of phone text data ===
=== IP232: Invalid text on second line of fkey label ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73775 73775]
|[http://mantis.innovaphone.com/view.php?id=80484 80484]
|}
|}
External input<!---->
Invalid text on second line of fkey label when changing registrations.<br/><br/><!--<br/>forms_color.cpp<br/>--><!---->
      
      
=== SIP: Handling of collision of transfer and release ===
=== PBX: Routing problem from sub-slave to master to object in sub-slave node, but registered to master ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=73936 73936]
|[http://mantis.innovaphone.com/view.php?id=80591 80591]
|}
|}
If one end releases a call while the other initiates an attended transfer, a "ghost call" may remain.<br/>Resource leak.<br/><!---->
Collateral damage from #77874: PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Limitation of background image size ===
=== SIP: Calls may remain in clearing state ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=73957 73957]
|[http://mantis.innovaphone.com/view.php?id=80623 80623]
|}
|}
Limit is 512 kBytes now.<br/>To keep phone from out of memory trap.<!---->
SIP calls may remains undeleted.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== H.323: Renegotiation after PBX Node dialtone failed sometimes ===
=== SIP: Code optimization ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=74020 74020]
|[http://mantis.innovaphone.com/view.php?id=80635 80635]
|}
|}
This depended on timing. This happened if the call was sent after the dialtone to a master over a slow link.<!---->
Reduce number object constructions/destructions during message encoding.<br/><br/><!--<br/>siptrans.cpp<br/>--><!---->
      
      
=== IP232: Open phone menu by touch gesture ===
=== PBX: Update of conference id did not always work after multiple transfer accross PBXs ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74046 74046]
|[http://mantis.innovaphone.com/view.php?id=80656 80656]
|}
|}
Touching header bar and moving vertically down opens the phone menu.<br/>No need to use center key of 4-way control.<!---->
A scenario which did not work was A calls B, B does consultation to C, B Transfers, C does consultation to D, C transfers with B on different PBX then A.<br/>After this the conference id on the call on A should be identical to the conference id on D. This was not the case.<br/><br/>SOAP/TAPI applications which are keeping track of transfered calls could have a problem with this.<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== Dect User Gui: AD-replicated objects weren't listed under certain conditions ===
=== AD Replication: LDAP filter encoding failed, when Poll Timer was configured ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74051 74051]
|[http://mantis.innovaphone.com/view.php?id=80658 80658]
|}
|}
Deleted object thate were re-replicated werent't visible in the Dect user GUI. A mandatory Ldap attribute was missing under such circumstances.<br/><!---->
AD Replication: LDAP filter encoding failed, when Poll Timer was configured<br/><!--<br/>adrep.cpp<br/>--><br/><!---->
      
      
=== IP222: Handset Microphone level too low ===
=== PBX: More consistent use of conferenceId after transfer ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=74073 74073]
|[http://mantis.innovaphone.com/view.php?id=80660 80660]
|}
|}
Handset micro increased by 6db<br/>Handsfree micro unchanged<br/><!---->
The conference ID is used (SOAP/TAPU, CDRs) to associate different call legs to the same call. After a transfer two calls, which have been seperate are connected, so one of the call legs has to change its conference ID, so that the resulting call has a single conference ID again.<br/><br/>There was a complicated logic implemented in the PBX to decide which conference ID should be used, this is now changed to a simple logic: The conference ID of the call on which the transfer is performed, is used.<br/><br/>Example:<br/><br/>A calls B, B does a consuktation to C, and B transfers A to C - This means the transfer is performed on call leg A, so the conference ID of the original call A-B is used for A-C<br/><!--<br/>pbx.cpp<br/>--><!---->
''Status:''
ac_codec3.cpp<br/>
      
      
=== IP222,IP232: Language configuration not saved ===
=== Gateway: Routing of incoming SIP calls may not work ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74087 74087]
|[http://mantis.innovaphone.com/view.php?id=80709 80709]
|}
|}
Language can be selected on phone menu, but change is neither saved nor applied.<!---->
... because the To-URI is not reduced to a called number (CDPN).<br/>Gateway routing is based on CDPN and does not work for URIs.<br/><br/>Before this fix:<br/>.LOG CALL 2 Alloc<br/>.LOG CALL 2 A:Call    ->                        / GW8::->*::<br/>.LOG CALL 2 B:Call    :081604998@212.13.249.90->:018108680@192.168.5.230 / GW8::->GW1::<br/><br/><br/>After this fix:<br/>.LOG CALL 1 Alloc<br/>.LOG CALL 1 A:Call    ->                        / GW8::->*::<br/>.LOG CALL 1 B:Call    081604998->018108680      / GW8:081604998:->GW1:018108680:<br/><br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== SIP: Don't take NOTIFY(application/simple-message-summary) with "Do-Not-Disturb: no" as FTY_MWI_DEACTIVATE ===
=== SIP: Memory leak when closing signaling interface ===


{|
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|[http://mantis.innovaphone.com/view.php?id=74162 74162]
|[http://mantis.innovaphone.com/view.php?id=80752 80752]
|}
|}
Interop with Nortel CS1k PBX<!---->
Memory leak when closing signaling interface while DNS request is pending on a call.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== Phones: Fine grained function hiding did not disable Fkey configuration ===
=== Phone: Message function key cannot be configured ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74174 74174]
|[http://mantis.innovaphone.com/view.php?id=80766 80766]
|}
|}
Value 0x00000800 (PHONE_HIDE_US_FUNCTION_KEYS) hides function keys entry from user-list,<br/>but fkeys can still be (re)configured by long fkey press.<br/><br/>For more information see:<br/>http://wiki.innovaphone.com/index.php?title=Reference9:Phone/Protect<br/><!---->
Configuration is not saved<br/><br/><!--<br/>phone_edit.cpp<br/>fkey_edit_message.xsl<br/>--><!---->
      
      
=== myPBX: Allow asterisk as a wildcard in LDAP search strings ===
=== SIP: Diversion information was wrong on incoming SIP calls ===


{|
{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=74200 74200]
|[http://mantis.innovaphone.com/view.php?id=80812 80812]
|}
|}
As it is done on our phones.<!---->
Parsing of History-Info header was wrong, but only if multiple History-Info headers were present.<br/><br/><!--<br/>sipmsg.cpp/h<br/>--><!---->
      
      
=== IP232: Pixel noise with some color values ===
=== PBX: Routing problem with nodes/escapes/slaves with calls to extern numbers in other PBX defined nodes ===


{|
{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=74221 74221]
|[http://mantis.innovaphone.com/view.php?id=80853 80853]
|}
|}
- changed to lower pixel clock<!---->
Could happen that the number dialed when sending the call to the extern interface was not adjusted correctly<br/><!--<br/>pbx.cpp<br/>--><!---->
''Status:''
orchid_lcd.cpp
      
      
=== PBX: Conference to VM ===
=== Build Number format changed ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74222 74222]
|[http://mantis.innovaphone.com/view.php?id=80925 80925]
|}
|}
Now a call to the VM as a conference member works again.<!---->
The build number of the hotfixes changes from the 90600.xx format to the 9.061xxx format. This is due to organizational changes without any other significance.<br/><!--<br/>box.cpp<br/>--><!---->
      
      
=== RTP-DTMF: Digit may get lost during media re-negotiation ===
=== IP-DECT: Memory leak for SIP calls ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=74232 74232]
|[http://mantis.innovaphone.com/view.php?id=80937 80937]
|}
|}
Receiver starts handling on END event.<br/>Sender may stop sending before END event was sent.<!---->
buffer for received name-id was not freed. Could happen with other facilities as well.<br/><!--<br/>dtmffty.cpp<br/>--><!---->
      
      
=== No media for calls from SIP provider forwarded back to SIP provider in case of NAT and MediaRelay ===
=== SIP: Transcation handling was wrong ===


{|
{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=74290 74290]
|[http://mantis.innovaphone.com/view.php?id=80989 80989]
|}
|}
NAT traversal depends on a packet being sent from inside the NAT to outside, to fix the RTP destination of the outside endpoint. This does not happen if both endpoints are outside.<br/><br/>Dummy packets are sent from the Media Relay function in this case to achieve this.<!---->
Double delete on a INVITE client transaction object.<br/><br/>0:0027:988:5 - SIP_TAC_INVITE:serial::delete(805b0c80) caller=800440ec<br/><br/><!--<br/>siptrans.cpp<br/>--><!---->
      
      
=== ISDN: Send HLC with mobility calls ===
=== PBX: Adjusting called number, when sending call to extern ===


{|
{|
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|[http://mantis.innovaphone.com/view.php?id=74296 74296]
|[http://mantis.innovaphone.com/view.php?id=81005 81005]
|}
|}
Some ISDN networks refuse the forwarding of a call to a mobile network if no HLC (High Layer Compatibility) Information Element indicating Telephony is included in the call.<!---->
depending on the node of the extern object and the called node, the called party number has to be adjusted (escapes added, prefixes added/removed). This did not work unders some conditions.<br/><!--<br/>pbx.cpp<br/>--><!---->
   
= V9 Hotfix 13 (9061024) =
Changes included in Version 9 hotfix13
[http://mantis.innovaphone.com/view.php?id=81128 Definition]
 
== New Features ==
 
 
      
      
=== IP241,IP222,IP232: Improved phone screen rendering ===
=== myPBX: Interface for tracing ===


{|
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|[http://mantis.innovaphone.com/view.php?id=80973 80973]
|}
|}
Reduce map copy operations.<!---->
The web application can now write messages into the trace file of the myPBX launcher.<br/><!--<br/>client.js<br/>client_util.js<br/>--><!---->
      
      
=== SIP: Accept INVITE's from alternative proxy ===
=== Oem Code: Accelerating Boot Snmp Traps ===


{|
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|[http://mantis.innovaphone.com/view.php?id=81132 81132]
|}
|}
If alternative proxy is configured, calls should accepted from alternative proxy,<br/>even if primary proxy is alive and kicking.<!---->
Oem Code: Accelerating Boot Snmp Traps<!---->
      
      
=== Flash Directory: Substring match could miss results ===
=== PBX: Handling of call limits at PBX objects improved ===


{|
{|
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|[http://mantis.innovaphone.com/view.php?id=74320 74320]
|[http://mantis.innovaphone.com/view.php?id=81232 81232]
|}
|}
Searching by substring worked case sensitive in some cases, instead of functioning case insensitive.<!---->
A call from a PBX, which is sent back to the same PBX is not counted anymore. This can happen because of node-extern.<br/>Incoming calls at master, which are above the limit are rejected now. They can be rerouted on the slave with "Route Master calls if no Master to"<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_api.h<br/>--><!---->
      
      
=== PBX: CF at Gateway Type objects - additional dialed digits should be added to the destination ===
=== SIP: New config option for endpoints not refreshing their registration during call ===


{|
{|
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|[http://mantis.innovaphone.com/view.php?id=81243 81243]
|}
|}
This way a CFNR at a trunk object can be used to reroute the call to another trunk.<!---->
Interoperability:<br/>New config option /keep-active-endpoints for endpoints not refreshing their registration during call.<br/><br/>AVM FRITZ!Box Fon WLAN 7270 v2 (UI) 54.05.21 (Apr  2 2012)<br/><br/><!--<br/>sip.cpp/h<br/>siptrans.cpp<br/>--><!---->
      
      
=== Gateway: Deleting of routes could result in duplication of routes ===
=== UI hint where to update Linux AP ===


{|
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|[http://mantis.innovaphone.com/view.php?id=74356 74356]
|[http://mantis.innovaphone.com/view.php?id=81295 81295]
|}
|}
This happened if an interface registration was disabled, for which automatic routes have been generated and then a route was deleted. The last route was duplicated.<!---->
Changed a hint on the upload/update tab and added a link to the update/upload page on the firmware side.<br/><br/><!--<br/>lang*.inc<br/>application.php<br/>linux.xsl<br/>--><!---->
      
      
=== IP241/222/232: Monitormode (Lauthören): Level too low ===
=== SIP: New config option /no-certificate-check ===


{|
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|[http://mantis.innovaphone.com/view.php?id=74362 74362]
|[http://mantis.innovaphone.com/view.php?id=81601 81601]
|}
|}
Use independent analog codec channels for speaker and headset receiver.<br/>Speaker volume in monitor mode is configured as in handsfree mode.<br/><br/>To change to speaker level in monitor mode the gain of the speaker can be configured with<br/> config change AC-DSP0 RINGER /DualOutputModeGain level<br/> config activate<br/><br/>level is from 0..63  0<br/>0 -> -32dm <br/>32 -> 0db<br/>63 -> 31db<br/><br/><!---->
New config option to disable validation of remote certificate name<br/>when opening outbound TLS connection.<br/><br/><!--<br/>sip.cpp/h<br/>siptrans.cpp<br/>--><!---->
''Status:''
ac_codec3.cpp<br/>ac_codec3.h
      
      
=== IP241,IP222,IP232: Could not activate first builtin background image from phone menu ===
=== IP-DECT: Own priority for idle display set by messages ===


{|
{|
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|[http://mantis.innovaphone.com/view.php?id=81616 81616]
|}
|}
Could not activate first builtin background image from phone menu if external background image is currently active.<!---->
The idle display set by messages has got an own priority now.<br/><br/><!--<br/>Changed files:<br/>dtmffty.h<br/>dtmffty.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Show multiple diversions on incoming calls ===
=== SoftwarePhone: Support for Jabra PRO 930 ===


{|
{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=74371 74371]
|[http://mantis.innovaphone.com/view.php?id=81618 81618]
|}
|}
If a call was diverted more than once, the phone shows first diverting party (original called number) and last diverting party.<br/>(Not only the last diverting party)<!---->
Support for Jabra PRO 930 with product id 0x1016 added.<br/><br/><!--<br/>Changed file:<br/>hid.tbl<br/>--><!---->
      
      
=== myPBX launcher: Unhandled exception when accessing browser object ===
=== SIP: New config option "No Inband Disconnect" on GW interfaces ===


{|
{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=74380 74380]
|[http://mantis.innovaphone.com/view.php?id=81803 81803]
|}
|}
The myPBX launcher could crash when accessing the browser document because of an unhandled exception.<!---->
Introduced VOIP_OPTION_NO_INBAND_DISC.<br/>Option was missing on Gateway interfaces.<br/><br/><!--<br/>voip.h<br/>sip.cpp<br/>gk.cpp/h<br/>relay_edit_voip.xsl<br/>--><!---->
      
      
=== IP232: Pickup fkey displays no information in active state ===
=== SIP: New config option /product-id-format ===


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{|
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|[http://mantis.innovaphone.com/view.php?id=74392 74392]
|[http://mantis.innovaphone.com/view.php?id=81880 81880]
|}
|}
Pickup fkey displays no information in active state<!---->
Change the User-Agent string from<br/>  User-Agent: (innovaphone IP232/10.00 dvl [90910/90879/501])<br/>into<br/>  User-Agent: innovaphoneIP232x90910x501<br/>with /product-id-format 1<br/><br/><!--<br/>sip.cpp/h<br/>siptrans.cpp<br/>sipmsg.cpp<br/>--><!---->
      
      
=== Phones: Only one diverting party displayed on incoming calls ===
=== phone: ip222,ip232: more USB headsets supported ===


{|
{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=74406 74406]
|[http://mantis.innovaphone.com/view.php?id=81972 81972]
|}
|}
Only original called party displayed, not the last diverting party.<br/>Phones with non-color display only (IP240,IP230,IP200,IP11,IP150)<br/><br/><!---->
Sennheiser and some more Jabra Headsets added. Look for "(since V9hotfix13)" in<br/>http://wiki.innovaphone.com/index.php?title=Reference9:Concept_USB_Headset<br/><br/><!-- keys_ip222.cpp --><!---->
      
      
=== NAT: Permanent UDP forwarding did not work ===
=== Debug information on assertion ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74549 74549]
|[http://mantis.innovaphone.com/view.php?id=81973 81973]
|}
|}
Permanent UDP forwarding was unusable after first forwarded datagram.<!---->
More debug information on default event handler.<!---->
      
      
=== PBX Waiting: If used as outgoing dialing device, coder list sent with the call was not correct ===
=== phone: ip222,ip232: support for advanced USB headset functions (redial, reject call, accept waiting call) ===


{|
{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74593 74593]
|[http://mantis.innovaphone.com/view.php?id=81975 81975]
|}
|}
Should be based on the coders paramter in the configured URL<!---->
A lot of USB headsets generate special indications to request redialing of last number dialled, to reject a ringing call, to accept a waiting call and to put the active call on hold or to switch between an active and an held call.<br/><br/><!--<br/>keys_gen.cpp<br/>keys_ip222.cpp<br/>forms_color.cpp<br/>forms_impl.cpp<br/>forms_keyboard.cpp<br/>keyboard.h<br/>app_call.cpp<br/>app_reg.cpp<br/>app_ctl.cpp<br/>app_ctl.h<br/>--><br/><br/><!---->
   
== Bug Fixes ==
 
 
      
      
=== Phone_orchid: Residual echo in handset ( sporadic) ===
=== IP222  equalizer update ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74671 74671]
|[http://mantis.innovaphone.com/view.php?id=80004 80004]
|}
|}
Echo canceller NLP sensitivity Mode changed to 2 <!---->
- handsfree speaker equalizer enabled<br/>- handset mic and receiver equalizer smoothed<br/>- ADC gain reduced, input gain increased ( after ec ) to avoid clipping<br/>- halfduplex mode disabled<br/><br/><!-- ac_codec3.cpp platform/filter/* --><!---->
''Status:''
ac_dsp3.cpp
      
      
=== IP222,IP232: Function key "Hotdesk" did not work ===
=== IP222 IP232 IP241: repeated ethernet link status 1000M wrong ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74723 74723]
|[http://mantis.innovaphone.com/view.php?id=80029 80029]
|}
|}
Function key "Hotdesk" did not work<!---->
.<br/><br/><br/><!-- orchid_drv.cpp --><!---->
      
      
=== IP232: Fix for long touch on fkey ===
=== IP222 IP232 : Default LCD backlight standby brightness reduced ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74743 74743]
|[http://mantis.innovaphone.com/view.php?id=80031 80031]
|}
|}
Opens fkey config, but mist not toggle control on touch-off.<!---->
<br/><!-- config.h --><!---->
      
      
=== Web-UI: Configuration of "Sync Server" does not return with HTML page ===
=== IP241 handsfree equalizer update, IP222 handsfree mic equalizer update ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74778 74778]
|[http://mantis.innovaphone.com/view.php?id=80157 80157]
|}
|}
Configuration of "Sync Server" does not return with HTML page<!---->
- IP241 handsfree speaker equalizer enabled<br/>- IP241 handsfree micro equalizer enabled<br/>- IP222 handsfree micro equalizer enabled <br/>- All three filter calculated with measured frequency response <br/>- IP241 ADC gain reduced, input gain increased ( after ec ) to avoid clipping<br/><br/><br/><!-- ac_codec3.cpp platform/filter/* --><!---->
      
      
=== IP222,IP232: changes not saved when leaving menu screen with LEFT key ===
=== phone: ip222, ip232: Pressing Talk button on USB Headset when there is an active and a held call does not transfer ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74792 74792]
|[http://mantis.innovaphone.com/view.php?id=80413 80413]
|}
|}
Even after confirming to save<!---->
Instead of transfer the active call is disconnected and the held call is retrieved<!---->
''Status:''
checked in to 10.00
      
      
=== IPxx10: Flashdir Segments Default to 129(was 51) ===
=== IP241,IP222,IP232: Show name of diverting party in incoming calls ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74826 74826]
|[http://mantis.innovaphone.com/view.php?id=80839 80839]
|}
|}
IPxx10: Flashdir Segments Default to 129(was 51)<!---->
Show name of diverting party in incoming calls (not only number).<br/><br/><!--<br/>forms_color.cpp/h<br/>call-ctrl-ring.png<br/>call_diverted.png<br/>call_transfered.png<br/>backgrounds_320x240.mak<br/>backgrounds_480x272.mak<br/>icons.mak<br/>--><!---->
      
      
=== Interop: Don't add an error log "SRTCP authentication failed" on non-RTCP packets ===
=== PBX-SOAP: UserRedirect - original called number got lost ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74903 74903]
|[http://mantis.innovaphone.com/view.php?id=80854 80854]
|}
|}
Microsoft Lync send invalid RTP and RTCP packets at the beginning of each call.<br/>In case of encrypted media, decryption of these packets fails.<!---->
For example if an application used an Waiting Queue object to monitor for incoming calls and redirected these calls to agents. The agent receiving the call could not see if the call was diverted to the waiting queue already.<br/><!--<br/>pbx_xml.cpp<br/>--><!---->
      
      
=== phone_orchid: ethernet statics counters sometimes displayed as negative numbers ===
=== SIP: Send 200/OK for MESSAGE(text/plain) when accepted by application ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74913 74913]
|[http://mantis.innovaphone.com/view.php?id=81017 81017]
|}
|}
the unsigned counters were printed as signed integers<!---->
Application gives "NormalCallClearing" to SIP stack.<br/>SIP stack better sends "200 OK" instead of "603 Decline".<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Volume control not displayed when already at maximum ===
=== SIP: Un-escape content of XML elements ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74927 74927]
|[http://mantis.innovaphone.com/view.php?id=81019 81019]
|}
|}
Volume control is not displayed when trying to increase while being at maximum.<!---->
XML element content requires some resevered characters to be escaped (<>).<br/>These escape sequences (&amp;gt; or &amp;lt;) must be un-escaped onthe receiving end.<br/><br/><!--<br/>xml.cpp/h<br/>sip.cpp<br/>soap.cpp<br/>fault_handler.cpp<br/>webex.cpp<br/>pbx_bool.cpp<br/>pbx_dtmf.cpp<br/>pbx_mwi.cpp<br/>pbx_vm.cpp<br/>pbx_wait.cpp<br/>--><!---->
      
      
=== IP6000 crypto driver: Trap when buffers are depleted ===
=== IP241,IP222,IP232: Two status symbols may overlay each other ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74935 74935]
|[http://mantis.innovaphone.com/view.php?id=81027 81027]
|}
|}
Avoid the trap and log an Event when the buffers are depleted.<!---->
Symbol for "Call diversion" and symbol for "PIN-locked" overlay each other.<br/>Hide "Call diversion" as long as phone is PIN-locked.<br/><br/><!--<br/>forms_color.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Mute status indication does not work ===
=== SIP: Handling of multiple 401/407 responses ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74964 74964]
|[http://mantis.innovaphone.com/view.php?id=81045 81045]
|}
|}
"MICROPHONE OFF" is displayed on status bar now.<!---->
Implemented handling of up to 3 401/407 responses per transaction.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== SIP: Rare problem with blind transfer ===
=== SIP: Trap while releasing call ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=74991 74991]
|[http://mantis.innovaphone.com/view.php?id=81089 81089]
|}
|}
callIdentity must be zero when giving ctInitiate to PBX on blind transfer.<!---->
Timer fires during call release and causes trap.<br/><br/><!--<br/>sip.cpp<br/>sipstate.cpp<br/>--><!---->
      
      
=== TLS: Flow control for incoming data ===
=== H.323: Memory leak when sending special OEM H.235 key elements ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75004 75004]
|[http://mantis.innovaphone.com/view.php?id=81160 81160]
|}
|}
The TLS socket has to wait for the application to process incoming data before sending the next RECV.<!---->
Only happens if Avaya SRTP is enabled in OEM products<br/><!--<br/>h323ch.cpp<br/>--><!---->
      
      
=== H.323: Renegotiating to Fax was rejected, if PCM switch and local networks were used ===
=== H.323: Potential Trap in case of high load and media renegotiation/call clearing collision ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75006 75006]
|[http://mantis.innovaphone.com/view.php?id=81173 81173]
|}
|}
If the side which initiated a switch to T.38 has configured PCM and the media address was classified as local due to local network configuration, the T.38 was rejected.<!---->
There is a very unlikely situation when media-renegotiation is started and then the call is cleared, which could cause a message related to the media-renegotiation to be sent to a already deleted call object. High load could make this situation more likely.<br/><!--<br/>h323sig.cpp<br/>--><!---->
      
      
=== SIP: Interworking of divertingLegInformation1 improved ===
=== IP241 handset equalizer ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75025 75025]
|[http://mantis.innovaphone.com/view.php?id=81174 81174]
|}
|}
Honour subscriptionOption<!---->
Enable IP241 handset micro and speaker equalizer.<br/>Same values as on IP222.<br/>Reduces noise in G711 mode. <!---->
      
      
=== IP222,IP232: Changes not saved in some cases ===
=== IP1060/3010/6010/22/24/28/302/305: sequence number to DSP changed from byte to word, caused jiter buffer warnings ===


{|
{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75088 75088]
|[http://mantis.innovaphone.com/view.php?id=81189 81189]
|}
|}
Some config screens did not write changed settings directly after "Save Changes" dialog.<br/>If menu was left with DISC key (instead of ESC key) the changes have been discarded.<br/><br/>Also affects other phones: IP241,IP240,IP230,IP110<br/>Changes are saved immediately when leaving the current screen.<br/>Not when leaving "User Settings" or "Phone Setting" screen.<!---->
<!-- ac_dsp3.h--><!---->
      
      
=== PBX Waiting: Diverting leg1 info not correct when diverting to a Waiting Queue ===
=== IP222/232/241/1060/3010/6010/22/24/28/302/305: min jitter buffer changed to 10ms ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75091 75091]
|[http://mantis.innovaphone.com/view.php?id=81197 81197]
|}
|}
If Waiting queue not defined in root node and escapes are used<!---->
was 35ms, now 10ms as on ip6000/800<br/><br/><!-- ac_dsp3.cpp ac_phone3.cpp --><!---->
      
      
=== PBX: Adjusting received leg1 info wrong ===
=== IP-DECT: Duplicate call setup ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75097 75097]
|[http://mantis.innovaphone.com/view.php?id=81213 81213]
|}
|}
In context of nodes with escapes<!---->
The IP-DECT Master sends in some circumstances a call twice to the same radio in the same time. This affects only the IP1202 and OEM devices, not the IP1200, and is fixed now.<br/><br/><!--<br/>Changed file:<br/>dectmaster.cpp<br/>--><!---->
      
      
=== VM: &lt;pbx-upd-obj type="cfu"..&gt; without effect when invoked multiple times ===
=== IP-DECT: OEM configuration option ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75121 75121]
|[http://mantis.innovaphone.com/view.php?id=81214 81214]
|}
|}
Statement <pbx-upd-obj type="cfu"..> failed to work properly after being used for diversion manipulation multiple times within a single script session.<!---->
The configuration format of an OEM configuration option was changed for the backward compatibility.<br/><br/><!--<br/>Changed file:<br/>dectmaster.cpp<br/>--><!---->
      
      
=== IP28 Watchdog leads to endless interrupt and trap ===
=== phone_orchid:  micro mute when a waiting call was connected after an outbound call setup had  been cancelled ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75186 75186]
|[http://mantis.innovaphone.com/view.php?id=81229 81229]
|}
|}
.<!---->
- A goes offhook<br/>- B calls A, a waiting call from B is indicated on A<br/>- A goes onhook, phone rings<br/>- A goes offhook again and is connected to B<br/>- A hears B, B doesn't hear A <br/><br/><!-- ac_codedc3.cpp --><!---->
      
      
=== IP800/6000: Problem mit FAX und CNG detect: Option to disable CNG detect added ===
=== SIP: Pass display information to application ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75278 75278]
|[http://mantis.innovaphone.com/view.php?id=81230 81230]
|}
|}
<br/>config change AC-DSP0 /t38-cng-detect-disable 1 <br/>config write<br/>config activate<br/><br/><!---->
Pass display information to application when registration comes up.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
''Status:''
ac_dsp3.cpp/h<br/>ac_fax3.cpp/h
      
      
=== SIP: Decrement Max-Forwards and interwork to H.323 ===
=== IP0010 IP1060 IP3010 IP6010: During long reset the ready LED was blinking orange and the Ethernet LEDs were swaped ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75288 75288]
|[http://mantis.innovaphone.com/view.php?id=81236 81236]
|}
|}
Instead of sending always "Max-Forwards: 70"<!---->
Now the ready LED shows the green blinking during long reset.<br/>The Ethernet LEDs are initialized directly after reset to overwrite the default setting that swaps link and speed.<br/><br/><!-- <br/>ip6010.cpp platform_mv78x00.c ether_mv78x00.c boot_xxx.y<br/><br/>Note: the IPxx10 uses always the 3 pin LED<br/>The 2 pin LED of the IP6000 is not used any more. <br/>The 3 pin LED needs both GPIOs to be zero in off state, with the 2 pin LED both GPIOs can be zero or one in off state.<br/>--><br/><!---->
      
      
=== H.323: Renegotiation to T.38 did not work for slowstart call thru media-relay to efc endpoint ===
=== X.509: Creating certificate containing IPv4 address did not work ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75305 75305]
|[http://mantis.innovaphone.com/view.php?id=81251 81251]
|}
|}
A common scenario when this happend is an XCAPI (slowstart) calling thru a PBX with media-relay configured to an ISDN interface<!---->
When creating a certificate or a request with an IPv4 address, the IPv4 address was mapped to an IPv6 address. This was not the expected behaviour.<br/><br/>172.16.10.32 -> ::ffff:172.16.10.32<br/><br/>Now both IPv4 and IPv6 addresses can be used.<br/><!--<br/>x509.cpp<br/>--><!---->
      
      
=== Fix for MIPS counter ===
=== ip222,ip232: waiting calls mute after having been accepted/connected on USB headsets, mainly seen with Jabra LINK 14201-30 ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75310 75310]
|[http://mantis.innovaphone.com/view.php?id=81312 81312]
|}
|}
MIPS counter was incorrect<!---->
In some cases only the accepted call was mute and the next call was OK again but the Jabra LINK 14201-30 lost the USB connection in most cases.<br/>Delaying the HID-commands sent to the headset solves this problem.<br/><br/><!--<br/>keys_ip222.cpp<br/>-->  <br/> <!---->
      
      
=== PBX: Allow 'Max Calls' configuration of 0 ===
=== PBX: Objects list filter for numbers did not work correctly anymore ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75362 75362]
|[http://mantis.innovaphone.com/view.php?id=81376 81376]
|}
|}
So that no call at all is sent to the object but a CFB (if configured) is executed<!---->
only top level nodes were displayed<br/>Collateral damage of fix<br/><br/>79982: PBX: When editing a Node object it was changed to a PBX object<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== SIP: Send "305 Use Proxy" if INVITE is received from unexpected source ===
=== SIP: Pass display information to application ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75380 75380]
|[http://mantis.innovaphone.com/view.php?id=81379 81379]
|}
|}
Applies to registered interfaces only (e.g. phones).<!---->
Pass display information received in REGISTER response to app.<br/><br/><!--<br/>sip.cpp/h<br/>--> <!---->
      
      
=== TLS: Possible trap when restoring sessions ===
=== IP241,IP222,IP232: Show H323-ID instead of Display Name if Display Name is not available ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75394 75394]
|[http://mantis.innovaphone.com/view.php?id=81419 81419]
|}
|}
Only in v9, depending on the certificate.<!---->
Like on old b/w phones.<br/><br/><!--<br/>forms_color.cpp/h<br/>--><!---->
      
      
=== PBX Exec Object: Leg2 information send to secretary not adjusted correctly for nodes ===
=== IP-DECT: Memory leak with special Cisco features ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75431 75431]
|[http://mantis.innovaphone.com/view.php?id=81463 81463]
|}
|}
The leg2 information is used to display at the secretary the number of the exec which was called. This number was not correct if nodes with escapes were used<!---->
Only with special OEM features.<br/><!--<br/>dtmffty.cpp<br/>--><!---->
      
      
=== Gateway Interface Maps: Should be applied to leg1 info also ===
=== SIP: Problems with DNS resolving of proxy adresses ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75437 75437]
|[http://mantis.innovaphone.com/view.php?id=81522 81522]
|}
|}
The same rules used for a Calling Party Number are applied to leg1 info<!---->
Not always re-tried when failed at startup.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== SoftwarePhone: Timer tick resolution ===
=== IP241,IP222,IP232: Rendering errors on Fkey configuration screen ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75447 75447]
|[http://mantis.innovaphone.com/view.php?id=81556 81556]
|}
|}
The timer tick resolution is increased. This fixes the call RTCP calculation transmitted to the PBX.<!---->
When scrolling or leaving Fkey configuration screen.<br/>List (Toggle) controls were not rendered correctly.<br/><br/><!--<br/>form_color.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Indirect dialing of names not possible ===
=== IP-DECT: Mobility Master registration ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75451 75451]
|[http://mantis.innovaphone.com/view.php?id=81571 81571]
|}
|}
Could not dial a name from 'indirect dialing' screen<br/>by moving the focus to input field and going offhook.<br/>(Only fkey worked)<!---->
The Mobility Master does not accept clients with the OEM name IP1202. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dectmobmaster.cpp<br/>--><!---->
      
      
=== PBX: CFNR Loop check detected loops that weren't ===
=== HTTP: Possible trap on many simultaneous sessions ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75465 75465]
|[http://mantis.innovaphone.com/view.php?id=81597 81597]
|}
|}
Esspecially a CFNR at a Waiting Queue was not executed if the destination of the CFNR has transfered the call to the Waiting Queue<!---->
Sorting the TCP sessions did not work correctly, which caused an assertion because it could happen that it was not possible to remove a session<br/><!--<br/>ip_user.cpp<br/>--><!---->
      
      
=== Gateway: Event "No Media Data Received" was created for each SWITCH-PCM call ===
=== IP-DECT: System GUI, disabled local coder options ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75483 75483]
|[http://mantis.innovaphone.com/view.php?id=81603 81603]
|}
|}
There should be no event in this case, this is normal.<!---->
If the PARI function (only IP1202) of the IP-DECT Master is disable, configuration changes on the System GUI do not effect anything. The settings for the local coder are disabled on this GUI page now.<br/><br/><!--<br/>Change file:<br/>_inn_asc_dectusers.xsl<br/>--><!---->
      
      
=== PBX: Conference id prefix/suffix configuration ===
=== IP-DECT: System settings not to dynamically connected radios ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75507 75507]
|[http://mantis.innovaphone.com/view.php?id=81612 81612]
|}
|}
The broadcast conference PBX configuration is changed: now, the third party conference unit option is saved and must be enabled to use the configured id prefix and suffix. Otherwise they are ignored now and default values for the innovaphone conference interface of the current device firmware version are used. This fixes the configuration if the firmware is updated from V8 to V9 and the innovaphone conference interface is used.<br/>Disabling the "Create Dynamic Conference Id" option in firmware V9 hotfix 5 and 6 is also fixed now.<!---->
System settings should not be sent if the radio-master connection is a dynamic one (IP1202 only). This is fixed now.<br/><br/><!--<br/>Changed files:<br/>dectusers_if.h<br/>dectusers.h<br/>dectusers.cpp<br/>dectmaster.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Call duration display wraps after 100 minutes ===
=== Support for old versions of Jabra BIZ 2400 headset with 48 kHz output sampling frequency ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75508 75508]
|[http://mantis.innovaphone.com/view.php?id=81613 81613]
|}
|}
Now after 60 minutes the display changes from mode [mm:ss] to [hh:mm].<!---->
the newer versions come with the usual 16 kHz sampling frequency but have the same signature (vendor/product) as the older ones.<!---->
      
      
=== TLS: Duplicate alert message on malformed ClientHelloV2 ===
=== SoftwarePhone: Trap with trace and mute ringer event ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75509 75509]
|[http://mantis.innovaphone.com/view.php?id=81619 81619]
|}
|}
Only one alert should be sent per session.<!---->
A trap occurs if the trace is enabled and the ringer mute option is toggled.<br/><br/><!--<br/>Changed file:<br/>phone_V5/phone.h<br/>--><!---->
      
      
=== TLS: Improved negotiation of protocol version ===
=== IP222 IP232 IP241: Force same speed of the switch ports for 1000M/100M scenarios ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75510 75510]
|[http://mantis.innovaphone.com/view.php?id=81693 81693]
|}
|}
TLS server unnecessarily rejected ClientHello messages with TLS 1.1 and higher. Instead of rejecting it should tell the client that it wants to use TLS 1.0.<!---->
In scenarios that operate one Ethernet port with 1000M and the other with 100M the switch througput was low.<br/> Now the 1000M port is reconfigured to 100M, and the throughput is high.<br/><br/><!-- orchid_drv.* --><!---->
      
      
=== TLS: Skip empty records ===
=== SIP: Support for multiple audio media descriptions ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75511 75511]
|[http://mantis.innovaphone.com/view.php?id=81712 81712]
|}
|}
TLS record layer should ignore records with zero length without doing anything.<!---->
One without encryption and one with encryption:<br/><br/>v=0<br/>o=OpenStage-Line_0 968610650 446118927 IN IP4 10.30.1.144<br/>s=SIP Call<br/>c=IN IP4 10.30.1.144<br/>t=0 0<br/>m=audio 5012 RTP/AVP 9 8 0 18 101<br/>a=rtpmap:9 G722/8000<br/>a=rtpmap:8 PCMA/8000<br/>a=rtpmap:0 PCMU/8000<br/>a=rtpmap:18 G729/8000<br/>a=rtpmap:101 telephone-event/8000<br/>a=silenceSupp:off - - - -<br/>a=fmtp:18 annexb=no<br/>a=fmtp:101 0-15<br/>a=sendrecv<br/>m=audio 5010 RTP/SAVP 9 8 0 18 101<br/>a=rtpmap:9 G722/8000<br/>a=rtpmap:8 PCMA/8000<br/>a=rtpmap:0 PCMU/8000<br/>a=rtpmap:18 G729/8000<br/>a=rtpmap:101 telephone-event/8000<br/>a=silenceSupp:off - - - -<br/>a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ujVU8G6kgknZnPflRwx8tadNskkp9glas/DFCbC3<br/>a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:+uoFBNVVhDY5OOGvMOdAlvpxvc98hX/VeProhlwH<br/>a=fmtp:18 annexb=no<br/>a=fmtp:101 0-15<br/>a=sendrecv<br/><br/><br/><!--<br/>sip.cpp/h<br/>sdpmsg.cpp/h<br/>--><!---->
      
      
=== myPBX: Script error after waking the PC up from hibernation ===
=== IP232: Show more of collapsed fkeys ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75532 75532]
|[http://mantis.innovaphone.com/view.php?id=81807 81807]
|}
|}
Sometimes window.open throws an unhandled exception in this case.<!---->
140px when expanded.<br/>100px when collapsed (80px before).<br/><br/><!--<br/>forms_color_mod.cpp<br/>--><!---->
      
      
=== SIP: Check proxy availability did not work in any scenario ===
=== SIP: Double Replaces header in INVITE after REFER ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75542 75542]
|[http://mantis.innovaphone.com/view.php?id=81847 81847]
|}
|}
OPTIONS can be used to poll remote proxy's availablity<br/>to avoid TCP timeout when INVITE is to be sent.<br/>Signaling interface is marked as down and not used anymore.<!---->
Reject by "Swyx IpPbxSrv/8.1.0.246" with "400 Bad Request(Multiple values in single-value header Replaces)"<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== IP232: Hiding touch keyboard by touching a control ===
=== PBX Mobility: No response for call to busy mobile phone, if no fixed phone ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75576 75576]
|[http://mantis.innovaphone.com/view.php?id=81850 81850]
|}
|}
Touching a text control activates the touch keyboard.<br/>Touching the text control again now deactivates the touch keyboard.<!---->
If a user is called with mobility configured and no fixed phone and the mobile phone was busy, then the call did not complete and was hanging as if the number was not complete. The call should be answer with busy instead.<!---->
      
      
=== IP241,IP222,IP232: CFU information in header bar ===
=== Chrome compatibility issue on PRI statistics page ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75654 75654]
|[http://mantis.innovaphone.com/view.php?id=81859 81859]
|}
|}
CFU information in header bar is now displayed even if there's not is enough space between name and number.<br/>Either name or number is omitted is required.<!---->
If clear was clicked an error message was displayed<!---->
      
      
=== IP241,IP222,IP232: Replace triangle by arrow to display diversion/transfer information ===
=== PBX Admin UI: Not possible to change DECT parameters for users with cf/grp admin rights ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75689 75689]
|[http://mantis.innovaphone.com/view.php?id=81871 81871]
|}
|}
Replace quite heavy 'BLACK RIGHT-POINTING POINTER'<br/>by much lighter 'RIGHTWARDS ARROW'<br/>to display diversion/transfer information<br/>on call control, fkeys and call lists.<!---->
The user interface prohibited the editing of DECT parameters based on the rights of the object to be edited not based on the login<!---->
      
      
=== H.323: A forwarded HopCount&gt;32 could result in a very small HopCount ===
=== phone: ip222,ip232: when a call is released from remote while another call is waiting the waiting call cannot be accepted. ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75765 75765]
|[http://mantis.innovaphone.com/view.php?id=81907 81907]
|}
|}
There are only 5 bits for transmitting a HopCount in H.323. A HopCount from SIP is typically 70 and this value was not reduced to 32 but only the 5 lower bits were transmitted, which resulted in a HopCount of 6<!---->
When the call is released from remote a buys tone is generated for two seconds. Therafter phone rings to indicate that the waiting call can be accepted now.<br/>When trying to accept this call by pressing the headset talk button the call was disconnected instead.<br/><br/><!-- keys_ip222.cpp --><!---->
      
      
=== IP: Minor memory management change ===
=== SIP: Memory leak in SIP stack ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75770 75770]
|[http://mantis.innovaphone.com/view.php?id=82058 82058]
|}
|}
no use of malloc in ipproc<!---->
Memory leak in SIP stack.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Two waiting calls are now displayed (instead of one) ===
= V9 Hotfix 14 (9061046) =
Changes included in Version 9 hotfix14
[http://mantis.innovaphone.com/view.php?id=82178 Definition]
 
== New Features ==
 


{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=75789 75789]
|}
Waiting calls are displayed below the two main calls.<br/>Only two lines of information and grey background.<!---->
      
      
=== SIP: Handling of some more proprietary Alert-Info and Call-Info in INVITE ===
=== phone: support PBX-directory access via TLS ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75824 75824]
|[http://mantis.innovaphone.com/view.php?id=78275 78275]
|}
|}
Alert-Info: <Bellcore-dr3>;info=alert-recall<br/>and<br/>Alert-Info: <http://not_used.com>;info=alert-autoanswer<br/>and<br/>Call-Info: <sip:127.0.0.1>;answer-after=0<!---->
A "Use TLS" checkmark has been added to the PBX directory config of the phone. If set, port 636 is used instead of port 389.<br/><br/><!-- phone_config.cpp, phone_config.h, phone_edit.cpp, reg_edit_dir.xsl, phone_dir.cpp --><!---->
      
      
=== PBX Waiting: Leg2 information sent with calls to operators not adjusted correctly for nodes ===
=== PBX-Mobility: Data Call Thru ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75830 75830]
|[http://mantis.innovaphone.com/view.php?id=81513 81513]
|}
|}
If a Waiting Queue was configured in a Node not the root node, the leg2 info was not adjusted corrcectly. The leg2 information is used to signal to the operator which Waiting Queue is forwarding the call<!---->
To speed up dialing with the mobility client, the called number can be posted to the PBX via HTTP before the call<br/><!--<br/>pbx_admin.cpp<br/>pbx_api.h<br/>pbx.h<br/>pbx.cpp<br/>pbx_mobility.h<br/>pbx_mobility.cpp<br/>--><!---->
      
      
=== SIP: Decoding of RFC-4412 definitions ===
=== IP-DECT: DECT security (2) ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75848 75848]
|[http://mantis.innovaphone.com/view.php?id=82191 82191]
|}
|}
The 'Resource-Priority' Header Field<br/>The 'Accept-Resource-Priority' Header Field<br/>The 'resource-priority' Option Tag<br/>417 Unknown Resource-Priority response<!---->
Some changes in preparation for the new feature "DECT security". This feature will be available later (IP1202).<br/><br/><!--<br/>New files:<br/>ascom/mobmaster/registration_context.h<br/>ascom/mobmaster/registration_context.cpp<br/>ascom/mobmaster/cryptomaster.h<br/>ascom/mobmaster/cryptomaster.cpp<br/>ascom/mobmaster/cryptomaster_registration.h<br/>ascom/mobmaster/cryptomaster_registration.cpp<br/>ascom/mobmaster/crypto_key_table.h<br/>ascom/mobmaster/crypto_key_table.cpp<br/>ascom/mobmaster/crypto_key_table_entry.h<br/>ascom/mobmaster/crypto_key_table_entry.cpp<br/>ascom/httpfiles/asc_dectcryptomaster.xsl<br/>ascom/httpfiles/asc_dectcryptomaster_mobmasters.xsl<br/><br/>Changed files:<br/>common/interface/signal.cpp<br/>dect_comcerto/platform/config.h<br/>ip1201/platform/config.h<br/><br/>dect/dectusers_if.h<br/>dect/users/dectusers.cpp<br/>dect/master/dectmaster.cpp<br/>dect/localusers/dectlocalusers.cpp<br/><br/>ascom/mobmaster/mobmaster.mak<br/>ascom/mobmaster/dectmobmaster.h<br/>ascom/mobmaster/dectmobmaster.cpp<br/>ascom/httpfiles/httpfiles.mak<br/>--><!---->
      
      
=== SIP: Handling of 422 Session Interval Too Small ===
=== IP-DECT: No fall-back after unattended call transfer ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75873 75873]
|[http://mantis.innovaphone.com/view.php?id=82198 82198]
|}
|}
Get Min-SE and re-try INVITE<!---->
Now there is no fall-back after an unattended call transfer and the behavior is consistent with the other call transfer types (attended, semi-attended). To switch back to the hold call the R-key must be pressed.<br/><br/><!--<br/>Changed file:<br/>dectradio_call.cpp<br/>--><!---->
      
      
=== SIP: Send b=TIAS attribute in media description ===
=== IP-DECT: Idle display update ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75881 75881]
|[http://mantis.innovaphone.com/view.php?id=82199 82199]
|}
|}
Send b=TIAS attribute in media description<!---->
Handling of idle display update message in the call release message is added. Used with an OEM PBX.<br/><br/><br/><!--<br/>Changed files:<br/>dectmaster.cpp<br/>dectfty.cpp<br/>--><!---->
      
      
=== remove indirect calls to os_mem_alloc() via malloc() ===
=== phone: ip222,ip232: the variable KEYS0/HID-MAP permits to map new USB headsets to builtin descriptors ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75886 75886]
|[http://mantis.innovaphone.com/view.php?id=82635 82635]
|}
|}
malloc() shall not be used anymore in sources dedicated to run on innovaphone hardware<!---->
Sometimes USB headsets come with a signature different from the signature of similar headsets which are already supported.<br/>A "vars create KEYS0/HID-MAP p <map>" maps the new signature to an existing one.<br/><map> format is<br/>  manufacturer:product=manufacturer:product<br/>the second manufacturer:product tuple is the signature of an already supported headset,<br/>'manufacturer' and 'product' are plain 4 digit hex numbers without a "0x" prefix.<br/><br/><!-- keys_ip222.cpp --><!---->
      
      
=== IPv6: Memory leak if packets received, which are not handled locally ===
=== IP2x2, IP241: Coder Preferences for prefered coder G.722 suboptimal ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75920 75920]
|[http://mantis.innovaphone.com/view.php?id=82815 82815]
|}
|}
E.g. an IPv6 multicast UDP packet, for some other application was not deleted.<!---->
In case G.7222 was selected as prefered coder and the called endpoint did not support G.722, as next best coder G.729 was selected. This is typically not what is desired in such a case, G.711 is the better alternative in this case<br/><!--<br/>ac_dsp3.cpp<br/>ac_phone3.cpp<br/>--><!---->
      
      
=== SIP: Fix for video negotiation ===
=== myPBX: Show version of launcher in the list of sessions ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75954 75954]
|[http://mantis.innovaphone.com/view.php?id=82821 82821]
|}
|}
Fix for video negotiation<!---->
The version of the used launcher is now shown on the page PBX/myPBX.<br/><!--<br/>ctiinterface.cs<br/>pbx_client_sessions.xsl<br/>pbx_client.h<br/>pbx_client.cpp<br/>client.js<br/>client_util.js<br/>--><!---->
''Status:''
checked-in: 10.00<br/>checked-in: 9.00
      
      
=== phone_orchid: after leaving a conference the active call was mute sometimes ===
=== H.323: Support for Avaya SRTP with AES128/80 ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75955 75955]
|[http://mantis.innovaphone.com/view.php?id=82829 82829]
|}
|}
The conference was not stopped internally in all cases. <!---->
Needed for OEM<br/><!--<br/>h323ch.cpp<br/>h323lib.cpp<br/>h323lib.h<br/>--><!---->
      
      
=== Flashman: Show meaningful result after OEM certificate upload in production ===
=== SoftwarePhone: HID Support for new headsets ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75958 75958]
|[http://mantis.innovaphone.com/view.php?id=83184 83184]
|}
|}
"Certificate" instead of "-unknown-"<!---->
With this new version the following headsets are supported for call control:<br/>- Jabra GO 6430 (Jabra LINK 350 USB with firmware 5.4.17 or later) with product id 0xa342. Please select the first device.<br/>- Jabra SUPREME UC (Jabra LINK 360 USB) with product id 0xa346. Please select the first device.<br/>- Jabra PRO 9470 with product id 0x1042.<br/>- Sennheiser VoIP USB headset (SH 350 IP) with product id 0x0008.<br/>- Sennheiser DW Office with product id 0x740a. Please select the first device.<br/>- Sennheiser CEHS-CI 02 (USB adapter cable) with product id 0x0030. Please select the second device.<br/><br/><!--<br/>Changed files:<br/>hid.tbl<br/>sphone_keys.cpp<br/>--><!---->
      
      
=== SIP: Fix for Supported header ===
=== PBX: Description was missing for DECT System object ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76016 76016]
|[http://mantis.innovaphone.com/view.php?id=83198 83198]
|}
|}
Re-implemented to easily add more tags<!---->
A configurable description is useful for the DECT System object as for all other objects<br/><!--<br/>pbx_edit_dect.xsl<br/>--><!---->
      
      
=== IP222,IP232: Alpha mode did not work on multi-line edit controls ===
=== Permit logging to a second (shadow) log server ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76165 76165]
|[http://mantis.innovaphone.com/view.php?id=83206 83206]
|}
|}
Could no enter alpha chars using the num block<!---->
"Services/Logging/Log Server/Log Server Shadow/Address" defines the adress of a second server. <br/>"Services/Logging/Log Server/Log Server Shadow/Enable" starts/stops logging to the second server.<br/>Except the address the configuration for the second server is copied from the first server.<br/><br/><!-- logging.cpp, logging.h, logging.xsl platform/config.h --><br/><!---->
   
== Bug Fixes ==
 
 
      
      
=== SIP: Trap when parsing very large History-Info header ===
=== Incorrect disk usage calculation for more than 4GB ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76198 76198]
|[http://mantis.innovaphone.com/view.php?id=81209 81209]
|}
|}
Trap when parsing very large History-Info header<!---->
The disk usage calculation was wrong, if more than ~4 GB of the card were used.<br/><br/><!--<br/>fat32.cpp<br/>fat32.h<br/>fat32.xsl<br/>--><!---->
      
      
=== SIP: SIP message may exceed available buffer ===
=== PBX Waiting: Call state shows "Disconnecting" after switch from announcement 1 to announcement 2 ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76203 76203]
|[http://mantis.innovaphone.com/view.php?id=81370 81370]
|}
|}
Trace message like this can occur:<br/> ERROR: SIP message buffer (1025) exceeded! (9,14,801d781c)<br/> SIP: CSeq missing<br/> SIP: Message encoding failed!<br/><!---->
Announcement disconnect changed state<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
      
      
=== SIP: Contact not allowed in BYE/CANCEL/PRACK request ===
=== PBX: Switch from Music on Hold to inband ringback on Alert after unpark oder transfer ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76210 76210]
|[http://mantis.innovaphone.com/view.php?id=81407 81407]
|}
|}
Acc. to RFC-3261 there must not be Contact header line in BYE/CANCEL request<br/>Acc. to RFC-3262 there must not be Contact header line in PRACK request<!---->
To give the caller feedback that soon somebody may answer the call<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== reset could fail, if issued during firmware update ===
=== PBX: Don't forward in-band info indicator if no media channel ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76266 76266]
|[http://mantis.innovaphone.com/view.php?id=81879 81879]
|}
|}
and even worse any following reset command were not accepted as well<!---->
Not good to indicate in-band info available if no media channel can be negotiated<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Show "New message from ..." on active phone screen ===
=== SIP: Trap when configuring user presence ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76283 76283]
|[http://mantis.innovaphone.com/view.php?id=81996 81996]
|}
|}
Show "New message from ..." on phone screen for 3 seconds when active with calls (non-idle).<br/>Like on b/w phones.<!---->
Trap may occur when configuring user presence.<br/><br/><!--<br/>sip_presence.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Toggle of display-name/name-alias/number was buggy ===
=== HTTP-Server: Configuration of "Public compact flash access" did not work for all cases ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76290 76290]
|[http://mantis.innovaphone.com/view.php?id=82064 82064]
|}
|}
String termination was missing.<br/>Garbage data was displayed.<!---->
E.g. /DRIVE/CF0/Neuer Ordner/ does not work, because HTTP request contains escaped sequences.<br/><br/><!--<br/>http.cpp<br/>--><!---->
      
      
=== IP232: Open call details when touching entry in call list ===
=== phone: ip222,232.241: no notification tone on a successfull redial attempt ===


{|
{|
Line 9,590: Line 9,489:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76292 76292]
|[http://mantis.innovaphone.com/view.php?id=82166 82166]
|}
|}
Instead of initiating call to remote party in call list entry.<!---->
On a failing or unanswered call the menu key opens the "Recall" menu. If "Redial" is selected the call is automatically redialed for 20 minutes in intervals depending on the result of the previous attempt. On success the user should be notified about the connection.<br/><br/><!-- ac_codec3.cpp --><br/><!---->
      
      
=== H.323: Merges from v10 for Media Negotiation fixes and special OEM protocol features ===
=== IP-DECT: Trap during subscribing handsets ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76301 76301]
|[http://mantis.innovaphone.com/view.php?id=82190 82190]
|}
|}
- Problem with transparent forwarding of SDP<br/>- special SRTP key exchange<br/>- Video fixes<br/>- Unit Testing support<!---->
A trap occurs during subscribing handsets on the IP1200. This is a fix for the previous fix #80424 and fixed again.<br/><br/><!--<br/>Changed file:<br/>dect.cpp<br/>--><!---->
      
      
=== IP-DECT: Memory leak with rejected radio registration ===
=== IP-DECT: Wrong release code ===


{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76308 76308]
|[http://mantis.innovaphone.com/view.php?id=82195 82195]
|}
|}
Memory leak occurred with a rejected radio registration is fixed, now.<!---->
The IP-DECT sends the wrong release code "User not reachable" if the handset was not connected and the remote parts disconnects. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dectradio_dectcall.cpp<br/>--><!---->
      
      
=== PBX-SOAP: When monitoring Waiting queue, the peer number was not indicated ===
=== IP-DECT: No delay with semi-attended call transfer and SIP ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=76315 76315]
|[http://mantis.innovaphone.com/view.php?id=82197 82197]
|}
|}
The number was considered presentation restricted, because of some bit manipulation error<!---->
If the SIP protocol is used and the user do a semi-attended call transfer, the call transfer is directly confirmed again. The semi-attended call transfer is stored in the base station and executed as an attended call transfer if the target party connects.<br/><br/><!--<br/>Changed files:<br/>dectradio.h<br/>dectradio.cpp<br/>dectradio_call.h<br/>dectradio_call.cpp<br/>dectradio_dectcall.h<br/>dectradio_dectcall.cpp<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp <br/>--><!---->
      
      
=== ip22/24/28/302/305/6010/3010/1060: Fax failed if tones are sent during the fax call ===
=== IP-DECT: Display info with failed user log-in ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76342 76342]
|[http://mantis.innovaphone.com/view.php?id=82200 82200]
|}
|}
In some cases the Fax messages were missing due to tones send during the fxx call<!---->
If the user log-in fails, the cause is shown in the display.<br/><br/><!--<br/>Changed file:<br/>dectradio_call.cpp<br/>--><!---->
''Status:''
ac_dsp3.cpp
      
      
=== Gatway: Transfer handling may cause re-routing after regular hang-up ===
=== IP-DECT: Short tone info by PBX only one time ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=76357 76357]
|[http://mantis.innovaphone.com/view.php?id=82204 82204]
|}
|}
Re-routing is expected to take place on non-connected calls only.<!---->
Tone information with defined length and requested by the gatekeeper is played only one time to the handset. This is used by an OEM PBX and fixed now.<br/><br/><!--<br/>Changed file:<br/>dectradio_call.cpp<br/>--><!---->
      
      
=== PBX Number Map: Not possible to use overlap dialing to Number map with incomplete destination ===
=== IP-DECT: Hanging aborted semi-attended call transfers with SIP ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76377 76377]
|[http://mantis.innovaphone.com/view.php?id=82205 82205]
|}
|}
If a Number Object with incomplete destination was called and the number was to be completed with overlap dialing a wrong number was called.<br/><br/>This is a usefull feature to use Number Maps as quick dial to other nodes. In this case Number Maps are used with a destination of the remote node, so the number is incomplete, the number within this node has to be dialed in addition to the Number of the Number Map object.<!---->
If the SIP protocol is used, a semi-attended call transfer is done by the user and the call transfer can not be executed, the remaining call party is not disconnected. This is added now.<br/><br/><!--<br/>Changed files:<br/>dectradio.h<br/>dectradio.cpp<br/>dectradio_call.h<br/>dectradio_call.cpp<br/>dectradio_dectcall.h<br/>dectradio_dectcall.cpp<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp<br/>--><!---->
      
      
=== SIP: Support for media recording ===
=== IP-DECT: LDAP server GUI description ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76396 76396]
|[http://mantis.innovaphone.com/view.php?id=82258 82258]
|}
|}
Media recording to HTTP URL<!---->
The LDAP server GUI description is wrong and corrected now. (The LDAP server can not be a HTTP server.)<br/><br/><!--<br/>Changed files:<br/>dectmaster.xsl<br/>asc_dectmaster.xsl<br/>_inn_asc_dectmaster.xsl<br/>phonebook_ldap.xsl<br/>--><!---->
      
      
=== Gateway: Conference interface, no voice ===
=== DHCP: Name registration at WINS was not refreshed as requested by TTL in registration response ===


{|
{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76419 76419]
|[http://mantis.innovaphone.com/view.php?id=82289 82289]
|}
|}
The ADSP firmware is changed to version 122. This fixes a bug in the conference interface of IP6000/IP6010/... which results in conference calls without voice in one direction for a single member.<!---->
If the DHCP-client gets a lease containing a WINS-server address and a NETBIOS node type P or M (1 or 2) the client tries to register it's NETBIOS-name (ipxxx-xx-xx-xx) with the WINS-server. The TTL returned by the server in the registration response determines when a name refresh has to be sent. <br/><br/><!-- dhcp.cpp, dhcp.h --> <!---->
      
      
=== phone_orchid: spurious trap in long conference calls ===
=== Phones: Presence info during ringing state may show garbage data ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76445 76445]
|[http://mantis.innovaphone.com/view.php?id=82306 82306]
|}
|}
in long confernce calls the phone may trap because the DSP delivers an unexpected zero size packet.<!---->
In case a presence update arrives at the phone while phone is in ringback state.<br/>Have been observed in conjunction with call forking with mobility only.<br/><br/><!--<br/>app_ctl.h<br/>app_call.cpp<br/>app_disp.cpp<br/>--><br/><!---->
      
      
=== IP241,IP222,IP232: Call held indication ===
=== myPBX: Support contact names containing a single quote ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76454 76454]
|[http://mantis.innovaphone.com/view.php?id=82323 82323]
|}
|}
If remote side has put the call on hold<br/>the user should get a visible indication<br/>(in addition to hearing music on hold).<br/><br/>New call status "held" is displayed.<!---->
Configuring contacts with a H.323 id containing a single quote (') caused script errors in the web application.<br/><br/><!--<br/>client.js<br/>client_allows.js<br/>client_dirsearch.js<br/>client_util.js<br/>--><!---->
      
      
=== LDAP-Expert, Edit Object Dialog: Could show more than one object ===
=== PBX: RTP-DTMF was disabled by Voice Mail object during re-negotiation ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76464 76464]
|[http://mantis.innovaphone.com/view.php?id=82332 82332]
|}
|}
LDAP-Expert, Edit Object Dialog: Could show more than one object<!---->
RTP-DTMF acc. to RFC-2833 was disabled by Voice Mail object during re-negotiation.<br/><br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== Qsig: Connected Number could be encoded wrong ===
=== Gateway CDR with '0. 0' charge amount ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76472 76472]
|[http://mantis.innovaphone.com/view.php?id=82359 82359]
|}
|}
Qsig: Connected Number could be encoded wrong<!---->
Should be '0.00' instead<br/><!--<br/>fty.cpp<br/>--><!---->
      
      
=== Gateway: Better handling of call-reroute requests ===
=== phone: call completion did not recover when DND(busy) was set at the phone requesting the call completion ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76515 76515]
|[http://mantis.innovaphone.com/view.php?id=82390 82390]
|}
|}
Apply interface maps to numbers in reroute request<!---->
when DND(busy) was set on the phone reqesting the call completion and was cleared some time later a "Recall possible" was not indicated anymore although a pending call completion was indicated on the called phone. <br/><br/><!-- phonesig_if.h, phonesig.cpp, phonesig.h, app_cc.cpp --><br/><!---->
      
      
=== PBX-SOAP: Better error handling when using a Waiting Queue for outgoing calls ===
=== ipv6: memory leak when sending fragmented packets ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76544 76544]
|[http://mantis.innovaphone.com/view.php?id=82394 82394]
|}
|}
- call was hanging if an invalid number or user without registration was called<br/>- trap if outgoing call was done on operator connect<!---->
<!-- ip6_main.cpp, ip6_route.cpp --><!---->
      
      
=== Spurious leaks in leak checking ===
=== H.323:No Media for calls with reverse media to a H.323/SIP exclusive Code Media Relay interface ===


{|
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|[http://mantis.innovaphone.com/view.php?id=76589 76589]
|[http://mantis.innovaphone.com/view.php?id=82408 82408]
|}
|}
A problem which mostly shows up in automated release testing and then requires manual investigation.<!---->
The execlusive coder/media relay config is used to avoid media negotiation problems with carrier which do not support media renegotiations. In case of a call with reverse media to such an interface, this did not work. This happens for example if a CFNR is configured at a Waiting Queue which redirects a call, which received an announcement from the Queue to such interface.<br/><!--<br/>h323ch.cpp<br/>--><!---->
      
      
=== SoftwarePhone: Auto start configuration ===
=== Kerberos: Allow editing multiple fields in admin UI ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76595 76595]
|[http://mantis.innovaphone.com/view.php?id=82425 82425]
|}
|}
The duplicate back slash occurred with the install directory is fixed now. Used by the run mode configuration 'At login'.<!---->
This is a fix for the page General/Kerberos in the admin UI.<br/><br/>Editing more that one item at a time could result in strange effects. Now it is possible to edit multiple fields.<br/><br/><!--<br/>kerberos_db.cpp<br/>kerberos_db.h<br/>kerberos_ldap.cpp<br/>kerberos_ldap.h<br/>--><!---->
      
      
=== myPBX launcher: Window position was not always remembered ===
=== phone_inca: when a call completion was set up with CLIR active the called party was not displayed on a possible recall ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76626 76626]
|[http://mantis.innovaphone.com/view.php?id=82435 82435]
|}
|}
When the user just moved the window without resizing it, the window position was not remembered for the next session.<!---->
A sets CLIR, A calls B, B is busy<br/>A sets a CCBS request via Menu/Recall<br/>B goes on Hook<br/>A rings and sees 'anonymous' instad of the number of 'B', status line is empty (should show "Recall possible")<br/><br/><!-- forms_gen.cpp --><br/><!---->
      
      
= V9 Hotfix 8 (90600.10) =
=== IP-DECT: Display info with remote control call ===
Changes included in Version 9 hotfix8
[http://mantis.innovaphone.com/view.php?id=76725 Definition]
 
== New Features ==
 
 
   
=== myPBX: Log event if application sharing failed ===


{|
{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=76116 76116]
|[http://mantis.innovaphone.com/view.php?id=82444 82444]
|}
|}
Log an event if the creation of a meeting failed.<!---->
The display info shown after accepting a remote control call is fixed now.<br/><br/><!--<br/>Changed file:<br/>dectradio_call.cpp<br/>--><!---->
      
      
=== myPBX: Localised date and time format in call list ===
=== IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: enable modembypass ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76737 76737]
|[http://mantis.innovaphone.com/view.php?id=82458 82458]
|}
|}
myPBX should show a localised date format instead of YYYY-MM-DD HH:MM:SS.<!---->
Modembypass is enabled on all calls with disabled T.38 and coders G711A or G711U. <br/>Switch to modem bypass is indicated in the trace by "switch to modembypass".<br/>The feature can be disabled with http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl <br/><br/>Modembypass works best if T38 is disabled on both sides.<br/>If T38 is enabled on the called side the CED may trigger a T38 session, this changed back to voice and modem bypass is enabled (if G711 is active). The first modem tone is interupted, but we still have modembypass on both sides.<br/>If T38 is enabled on the calling side the calling side stays on regular G711.<br/><br/><br/><br/><!-- ac_dsp3.cpp ac_dsp3.h dsp.xsl --><br/><!---->
      
      
=== IP241,IP222,IP232: Support for placeholder in background image url ===
=== IP800 IP6000: reduce probability of false DTMF detection ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76756 76756]
|[http://mantis.innovaphone.com/view.php?id=82471 82471]
|}
|}
Support for $serial and $product in background image url<br/><br/>URL<br/>  http://x.x.x.x/DRIVE/CF0/backgrounds_480x272/$serial.png<br/>results in<br/>  GET /DRIVE/CF0/backgrounds_480x272/00-90-33-30-00-bc.png HTTP/1.1<br/><br/>URL<br/>  http://x.x.x.x/DRIVE/CF0/backgrounds_480x272/$product.png<br/>results in<br/>  GET /DRIVE/CF0/backgrounds_480x272/IP232.png HTTP/1.1<br/><!---->
Change DTMF detector signal to noise ratio from 12db to 18db.<br/><!-- ac_dsp2.cpp  --><br/><br/><!---->
      
      
=== SoftwarePhone: Support for Plantronics Savi 740 ===
=== Gateway: 'Enable PCM' option added at the CONF interface of the IP800 ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76975 76975]
|[http://mantis.innovaphone.com/view.php?id=82481 82481]
|}
|}
HID Keypad Mapping support for the Plantronics Savi 740 with the product id 0xac01.<!---->
The option 'Enable PCM' is available at the CONF interface of the IP800 now.<br/><br/><!--<br/>Changed file:<br/>config.h<br/>--><!---->
      
      
=== phone_orchid: headset "Plantronics - Voyager PRO UC v2"  (USB/bluetooth) supported ===
=== Gateway: PCM mode of the CONF interface fixed ===


{|
{|
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|[http://mantis.innovaphone.com/view.php?id=77048 77048]
|[http://mantis.innovaphone.com/view.php?id=82482 82482]
|}
|}
HID report definitions added<!---->
The PCM mode of the CONF interface is not activated for a call even though it is configured and possible. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>conference_sig.cpp<br/>--><!---->
      
      
=== phone ip222, ip232 : Jabra Link 14201-30 DHSG/USB adapter cable supported, ===
=== IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: T38 trace flag added ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=82484 82484]
|}
|}
The Jabra Link 14201-30 connects the DHSG interface of Jabra Dual Mode Headsets (GN9350e, PRO 9450/9470 ..) to the USB port of an ip2x2. The USB port of the headset can be connected to a PC.<!---->
Allows to trace T38 connection on the PCM port and on the DSP host interface.<br/>Use this if fax modem problems are suspected.<br/>Enable at http://addr/debug.xml at trace->T38 trace.<br/><br/><!-- ac_dsp3.cpp ac_dsp.h ac_491.cpp ac_494.cpp ip6010/ip24: trace.xsl --><!---->
''Status:''
checked in to 10.00, 9.0, 90600
      
      
=== phone ip222, ip232 : Plantronics DA45  USB headset supported ===
=== Gateway: Routing problem with blockdial route and following matching non-blockdial incomplete routes ===


{|
{|
Line 9,866: Line 9,753:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77541 77541]
|[http://mantis.innovaphone.com/view.php?id=82486 82486]
|}
|}
.<!---->
Example:<br/><br/>Blockdial Route 00-><br/>After this a non-blockdial route with 0...<br/><br/>If now a number of 001 was dialed, the first route should match and after the enbock dialout the call should be sent to the destination of the route. Instead the call was rejected with "no destination found"<!---->
''Status:''
checked in to 10.0, 9.00, 90600
      
      
=== Trap during HTTP request with missing user/pwd in basic authentication header ===
=== failure of analog ports of ip28 ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77695 77695]
|[http://mantis.innovaphone.com/view.php?id=82488 82488]
|}
|}
.<!---->
ip28 analogue ports do not react to incoming calls and hook-off. Problem could only be solved by reset.<!---->
''Status:''
http.cpp
      
      
=== phone ip222, ip232 : USB headset support : Plantronics Blackwire C620, Jabra GN2000 USB Duo ===
=== Debug "HTTP_GET LOG_HTTP.1: retry, authentication failed" removed ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77703 77703]
|[http://mantis.innovaphone.com/view.php?id=82499 82499]
|}
|}
beim Kunden getestet<!---->
<!-- httpclient_i.cpp --><!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
== Bug Fixes ==
=== Phone: Display text received with BYE ===
 
 
   
=== SIP: Fix for REFER handling ===


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|[http://mantis.innovaphone.com/view.php?id=76309 76309]
|[http://mantis.innovaphone.com/view.php?id=82525 82525]
|}
|}
Handling of blind transfer without userpart in Refer-To-URI.<!---->
Pass display text to phoneapp.<br/><br/><!--<br/>phonesig.cpp<br/>--><!---->
      
      
=== SIP: Do not offer CLEARMODE on exclusive coder config ===
=== SIP: Trap during call handling ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=76654 76654]
|[http://mantis.innovaphone.com/view.php?id=82544 82544]
|}
|}
Do not offer CLEARMODE if exclusive coder configured<!---->
Trap during call handling<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== myPBX launcher: Always save window position when closing the application ===
=== IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: update to DSP code 660.27 ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76763 76763]
|[http://mantis.innovaphone.com/view.php?id=82551 82551]
|}
|}
The window position was only saved when the application was closed from the context menu of the tray icon. I.e. it was not saved when the user logged out.<!---->
660.27 is the released version on the audiocodes web.<br/>Fax relay and modem bypass improved.<br/><br/><!-- <br/>Laut AC49x Voice over Packet Processor (VoPP) Release Notes Ver 6.6 Rev3 bei den Gateways keine wesentliche Änderung zu 660.19, also nur ein update um auf Stand zu sein.<br/><br/>ac_491.cpp ac_494.cpp ip24.mak ip6010.mak ac_firmware.h --><!---->
      
      
=== SIP: Handling of 300 responses ===
=== IP222 IP232 IP241: update to DSP code 660.27 ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76771 76771]
|[http://mantis.innovaphone.com/view.php?id=82554 82554]
|}
|}
Select Contact-URI according to q-value.<br/>Replace remote target URI and change remote transport address.<!---->
660.27 is the released version on the audiocodes web.<br/>Improved acoustic echocanceller.<br/><br/><!-- <br/><br/>ac_494e.cpp ac_codec.cpp phone_orchid.mak  --><!---->
      
      
=== IP-DECT: Searching for users containing non-ascii characters failed ===
=== phone: call forwarding options offered even when call forwarding was not  possible ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76778 76778]
|[http://mantis.innovaphone.com/view.php?id=82567 82567]
|}
|}
A page indicating an xml error was displayed instead<!---->
Call forwarding is not supported when running SIP. But when the menu key was pressed after entering a number call forwarding options were offered (happened with the primary registration only). <br/><br/><!-- phone_user.cpp --> <!---->
      
      
=== ip22/24/28/302/305/6010/3010/1060/222/232/241: Silence compression with G711 coders not working ===
=== PBX Mobility: Trap in case of Transfer of a call from a mobile endpoint to another mobile endpoint ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76826 76826]
|[http://mantis.innovaphone.com/view.php?id=82584 82584]
|}
|}
Silence compression was used only for G729/G723. Now it is used for all codecs.<!---->
The Trap happens in the following call scenarion<br/>- Mobile endpoint calls in, using mobility two-stage dialing<br/>- call is accepted at local phone<br/>- on local phone a consultation call is initiated to another user with mobility<br/>- when mobile phone rings, a transfer is initiated on local phone<br/>- the called mobile phone accepts the call<br/>- the trap happens when the called mobile phone hangs up<br/><br/>There could be other call scenarions where the trap happens as well<!---->
''Status:''
ac_dsp3.cpp
      
      
=== Random problems with T.38 (Fax) Media Relay ===
=== SIP: SRTP key exchange failed ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76870 76870]
|[http://mantis.innovaphone.com/view.php?id=82616 82616]
|}
|}
Fax data could be corrupt<!---->
Bug in base64 decoding of SRTP key.<br/><br/><!--<br/>sdp.cpp<br/>--><!---->
      
      
=== phone_inca: wrong display of a CFU to a name containing non ascii characters ===
=== PBX Trunk: Number to Name Feature did not work for calls to busy endpoints ===


{|
{|
Line 9,988: Line 9,863:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76877 76877]
|[http://mantis.innovaphone.com/view.php?id=82619 82619]
|}
|}
all names are stored UTF8-encoded internally. the UTF8 string was displayed instead of a decoded version.<!---->
Such a call was not rejected with cause 'User Busy', but was just hanging.<!---->
      
      
=== Phones: Small change on screen "Directory input" ===
=== PBX Mobility: Call to mobile phone was sent with invalid diverting information ===


{|
{|
Line 9,999: Line 9,874:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76883 76883]
|[http://mantis.innovaphone.com/view.php?id=82622 82622]
|}
|}
Put a colonn at the end of label "Subscriber"<!---->
A call to a mobile phone is sent with a diverting leg2 info, which means, the call contains the information, that it was diverted by the called user to the mobile phone. So in theory this could be displayed on a mobile phone.<br/><br/>The coding of this information was wrong and created interop problems with some networks.<br/><!--<br/>pbx_mobility.cpp<br/>fty.h<br/>--><!---->
      
      
=== phone_orchid: no ring tone when ring volume was set to MAX via WEB config ===
=== SIP: Trap on subscription handling ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76927 76927]
|[http://mantis.innovaphone.com/view.php?id=82623 82623]
|}
|}
but ring tone was OK with any other volume<!---->
Trap on RAS_REGISTRATION_VERIFY between sending SUBSCRIBE and receiving 200/OK.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== PBX: Update of remote number after call-transfer did not always work across PBXs ===
=== SIP: Removed cisco-special retrieve signaling ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76942 76942]
|[http://mantis.innovaphone.com/view.php?id=82637 82637]
|}
|}
If a call was done from a phone at a slave to a phone at the master and on the master phone a consultation call was the done to the slave again with transfer afterwards, one phone on the slave did still show as connected number the phone on the master.<br/><br/>The same wrong number was also indicated on SOAP/TAPI as well as on myPBX.<!---->
Removed cisco-special retrieve signaling.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Config file option "/backlight-delay" to adjust display dimming ===
=== phone: ip222,ip232: Plantronics APU70 - Savi 7xx - Radio Link not cleared on release from remote ===


{|
{|
Line 10,032: Line 9,907:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76948 76948]
|[http://mantis.innovaphone.com/view.php?id=82654 82654]
|}
|}
Now works as it did on black 'n' white phones:<br/> config change PHONE FORMS /backlight-delay 1800<br/><!---->
On a release from remote for a call set up by pressing the Talk button (headset or base) the Radio Link between base and headset was not cleared until the Talk button was pressed again. <br/><br/><!-- keys_ip222.cpp -->    <!---->
      
      
=== IP222/IP232/IP241: residual echo  in handset mode ===
=== SIP: Mobility did not work due to RTP-DTMF ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76958 76958]
|[http://mantis.innovaphone.com/view.php?id=82674 82674]
|}
|}
new dsp code<!---->
DTMF must be passed through signaling channel to get mobility working.<br/>Suppress RTP-DTMF capability in SDP answer also.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
''Status:''
ac_494e.cpp
      
      
=== Memory Leaking In 802.1x Module ===
=== IP3010 IP810: number of DSP channels and number of conference channels was wrong. ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76960 76960]
|[http://mantis.innovaphone.com/view.php?id=82675 82675]
|}
|}
Memory Leaking In 802.1x Module<!---->
Now the IP810 shows 20 DSP channels and 30 conference channels.<br/>Now the IP3010 shows 42 DSP channels and 60 conference channels.<br/><br/><!-- ac_491.dsp ac_491.h adsp2191_drv.cpp adsp2191_drv.h ip6010-config.h ip6000 config.h --><!---->
      
      
=== PBX-SOAP: Call initiated by SOAP for softwarephone or IP-DECT was sent as transfered call ===
=== SIP: Restart NAT discovery if failed ===


{|
{|
Line 10,067: Line 9,940:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76962 76962]
|[http://mantis.innovaphone.com/view.php?id=82676 82676]
|}
|}
The result was that call diversions or busy on ... calls settings were ignored<!---->
Restart NAT discovery if failed<br/><br/><!--<br/>sip.cpp<br/>siptrans.cpp<br/>--><!---->
      
      
=== Edss1 Interworking: divertingLegInformation2 didn't contain redirectingNumber ===
=== IP222 IP232 IP241: Reduce DTMF level ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77003 77003]
|[http://mantis.innovaphone.com/view.php?id=82685 82685]
|}
|}
Edss1 Interworking: divertingLegInformation2 didn't contain redirectingNumber<!---->
During DTMF receive and transmit levels similar as on IP240 are used<br/><br/><!-- ac_phone3.cpp <br/>Beim Inca ist das so definiert:<br/>" /playback_dtmf_tx_coeff_LEV12 28"<br/>" /playback_dtmf_tx_coeff_ATT12 c0"<br/>0x28=40 --> -10db level<br/>0xc0=208--> 22db attenuation<br/>also insgesamt ein level von -32db ( bei Vollauschlag ) oder -29dbm.<br/>Der alte Wert beim ac_phone3.cpp war -9db <br/><br/>Weitere Diskussion:<br/>Es gibt den Fall das inband DTMF zum IP Netz geschickt wird, da gabs in Fall 59846 die Änderung zum IP mit LEV=0x28 -->-10db und Attenuation 0xff-->18db, also mit -28db zu senden. <br/>Da das gut funktioniert und die beiden  Pegel nicht so unterschiedlich sind unde der ac_dsp3 nicht unterschiedliche Pegel zum IP und zum Codec kann  nehmen wir die -28db=-25dbm.<br/><br/><br/>Laut www.itu.int/rec/dologin_pub.asp?lang=e&amp;id=T-REC-Q.24-198811-I!!PDF-E&amp;type=item Tabelle A-1 sind die  -25dbm am unteren Ende, da ist vom Wert A <br/>-22..-30 je nach Administration die Rede. Wahrscheinlich ist das kein Problem,<br/>bei Audiocodes (ac_dsp2) kann man die sensitivity von -28 bis -38db einstellen (DTMF_DETECTION_ENERGY_THRESHOLD__28dBm)<br/><br/><br/>--><!---->
      
      
=== IPVA, IP800, IP6000, IP6010: Typo RT30 instead of RT40 ===
=== IP222,IP232: Going offhook in call list always dials first list entry (not touched list entry) ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77027 77027]
|[http://mantis.innovaphone.com/view.php?id=82732 82732]
|}
|}
The routing table entry RT30 was duplicate and RT40 was missing. This causes config changes to be lost if more then 29 routes are configured<!---->
Going offhook in call list always dials first list entry (not touched list entry)<br/><br/><!--<br/>forms_color.cpp<br/>--><!---->
      
      
=== Phones: Not able to disable option "Use Newline/OK Key as Headset Key" ===
=== Memory leak after Firmware or Bootcode download ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77070 77070]
|[http://mantis.innovaphone.com/view.php?id=82740 82740]
|}
|}
Not able to disable option "Use Newline/OK Key as Headset Key"<!---->
After a Firmware or bootcode download via Maintenance/Download the associated command processor instance was not released.<br/><br/><!-- command.cpp --><!---->
      
      
=== Gateway: Routes got lost when reading config file ===
=== IP-DECT: Call transfer compatibility with SIP ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77157 77157]
|[http://mantis.innovaphone.com/view.php?id=82742 82742]
|}
|}
Gateway routes got lost when reading config file.<!---->
For compatibility reasons with the SIP protocol the call transfer initiate result message should not be sent until the connect message is received. This is changed now again.<br/><br/><!--<br/>Changed files:<br/>dectradio_call.cpp<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp<br/>--><!---->
      
      
=== SIP: 0.0.0.0 in o line of SDP ===
=== SIP: Trap if multiple calls arrive at same time ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77161 77161]
|[http://mantis.innovaphone.com/view.php?id=82743 82743]
|}
|}
    v=0<br/>    o=- 1 1 IN IP4 0.0.0.0<br/>    s=-<br/>    c=IN IP4 192.168.1.200<br/>   t=0 0<br/>   m=audio 16386 RTP/AVP 18 101 13 <br/>    a=rtpmap:101 telephone-event/8000<br/>   a=fmtp:18 annexb=no<br/>    a=fmtp:101 0-15<br/>   a=ptime:20<br/>    a=silenceSupp:off - - - -<br/>    a=sendrecv<br/><!---->
Trap if multiple calls arrive at same time.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== IP222,IP232: Fix for rendering problem of screen "Directory input" ===
=== SIP: Bug in digest authorization ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77209 77209]
|[http://mantis.innovaphone.com/view.php?id=82761 82761]
|}
|}
When leaving screen was refused with "Subscriber required" popup,<br/>the screen "Directory input" was not rendered again.<!---->
Sometimes wrong method is used in digest calculation.<br/><br/><!--<br/>sip.cpp<br/>siptrans.cpp/h<br/>--><!---->
      
      
=== SIP: Bug when interworking UUI of 1-byte length ===
=== phone: ip222,ip232:  USB headset mute after activation of changes in codec parameters, headset icon cleared on status line ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77225 77225]
|[http://mantis.innovaphone.com/view.php?id=82823 82823]
|}
|}
Results into bad formatted INVITE since User-to-User header contains uninitialized data.<!---->
After for example<br/>  config add AC-DSP0 HEADSET /InputGain 32<br/>  config activate<br/>the headset icon was cleared on status line and the headset was mute although the headset Talk key was handled. <br/><br/><!-- ac_codec3.cpp -->  <br/><br/><!---->
      
      
=== SIP: SDP answer must have the same number of media descriptions as received offer ===
=== PBX Waiting: Evaluate Busy on ... Calls for calls to an operator ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77277 77277]
|[http://mantis.innovaphone.com/view.php?id=82858 82858]
|}
|}
SDP answer must have the same media descriptions as received offer<!---->
If a operator has configured Busy on 1 call, there should be no call from the Waiting Queue if the operator is already busy.<!---->
      
      
=== SoftwarePhone: Exception during start up ===
=== PBX: Avoid signaling loop after call transfer ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77288 77288]
|[http://mantis.innovaphone.com/view.php?id=82865 82865]
|}
|}
Fix for fix #75886.<!---->
A signaling loop could be created by calling from a phone registered at one PBX to a phone at another PBX, then putting the call on hold and do the same call again, accept on the other side the waiting call. If both parties do then a transfer there is the signaling loop.<br/><br/>Such loop ist now detected and the call is cleared.<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== SIP: DNS refresh is not executed according to DNS-TTL ===
=== PBX Broadcast: No diverting name sent with broadcasted call ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77309 77309]
|[http://mantis.innovaphone.com/view.php?id=82880 82880]
|}
|}
Proxy IP address needs to be refreshed by DNS request according to DNS-TTL.<!---->
The diverting and original-called name info was missing from the diverting leg2 info generated by the broadcast object<br/><!--<br/>pbx_bc.cpp<br/>pbx_bc.h<br/>--><!---->
      
      
=== Phones: Show proxy IP address in web UI ===
=== Web-UI: Misplaced reset-required indication ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77316 77316]
|[http://mantis.innovaphone.com/view.php?id=82896 82896]
|}
|}
Useful if domain name has been configured and IP address has been resolved by DNS.<!---->
Misplaced reset-required indication on ISDN interface config (TEL1,TEL2,...).<br/><br/><!--<br/>ipac_drv.xsl<br/>isdn.xsl<br/>--><!---->
      
      
=== IP222/IP232/IP241: residual echo  in handsfree mode ===
=== IP0010 IP1060 IP3010 IP6010 IP810 IP302 IP305: ISDN: Enable fax detection only after connect - fixed ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77358 77358]
|[http://mantis.innovaphone.com/view.php?id=82900 82900]
|}
|}
changed ec settings<!---->
Fax detection was not enabled in all cases.<br/><br/><br/>Happens in test/10.00/box/dsp/ip6010 with<br/> fix: #78316: SIP/H.323: Don't complete media negotiation if no media can be seen<br/><br/><!-- ac_dsp3.cpp --><!---->
''Status:''
ac_codec3.cpp
      
      
=== SIP: Media negotiation during blind transfer caused ERROR log 0x70007 ===
=== IP1060: Memory size is not correctly shown ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77385 77385]
|[http://mantis.innovaphone.com/view.php?id=82906 82906]
|}
|}
Media negotiation during blind transfer caused ERROR log 0x70007 on the outbound call to transfer destination.<!---->
The memory size is not correctly shown in the IP1060 in diagnostics counter page.<br/><br/><!--<br/>Changed file:<br/>ip6010.cpp<br/>--><!---->
      
      
=== IP222/IP232/IP241/IP6010: trace feature tdm record: wrong endian ===
=== phone: a calling party name found by inverse directory lookup  for an external call was sometimes not stored in call list ===


{|
{|
Line 10,223: Line 10,094:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77508 77508]
|[http://mantis.innovaphone.com/view.php?id=82919 82919]
|}
|}
..<!---->
When the external call setup came in with a name identification provided by the external source and there was another name found by inverse directory lookup the name from directory was displayed on the call screen but the name identificication was stored in the call list. Now the name found by inverse directory lookup will be stored.<br/><br/><!-- app_call.cpp --><!---->
''Status:''
ac_dsp3.cpp<br/>ac_phone3.cpp
      
      
=== GUI: Adjust ETH DHCP link ===
=== PBX Routing: Node extern did not work for calls from a trunk marked as local object ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77538 77538]
|[http://mantis.innovaphone.com/view.php?id=82948 82948]
|}
|}
Adjust ETH DHCP link on page General Info and Reset is fixed now.<!---->
The call was not routed back to the originating slave<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== DECT: Reset link on page 'Features' ===
=== IP222 IP232 IP241: New equalizer and volume setting ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77539 77539]
|[http://mantis.innovaphone.com/view.php?id=82952 82952]
|}
|}
The reset link on the page 'Features' is fixed now.<!---->
- use equalizer up to approx 6db<br/>- use digital volume to adjust volume<br/> <br/><!-- ac_codec3.cpp filter/* --><br/><!---->
      
      
=== Media: Send CN frame with minimal noise level ===
=== SIP: Trap on out of memory ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77625 77625]
|[http://mantis.innovaphone.com/view.php?id=83004 83004]
|}
|}
For NAT reasons innovaphone endpoints send up to 4 RTP packets to remote end<br/>in order to open mapping on NAT router.<br/>These CN packets carry one-byte payload with value of 0.<br/>Value of 127 turned out to be more clever.<!---->
SIP-Client allocations not deleted.<br/>Outbound control calls without facility interworking.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== late RTP data sent from DSP driver to media endpoint may cause a trap ===
=== PBX Routing: A CFNR at a PBX object for WAN re-routing did not work if it contained escapes ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77642 77642]
|[http://mantis.innovaphone.com/view.php?id=83017 83017]
|}
|}
this was observed in a phone_orchid test setup with media recording activated <!---->
The number configured at the PBX object is interpreted in the context of the node of this PBX object. If escapes were needed to dial the WAN trunk, it did not work.<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== IP222/IP232/IP241: instable echo canceller  in handsfree mode ===
=== External-UC: Presence info assigned to wrong PBX object ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77661 77661]
|[http://mantis.innovaphone.com/view.php?id=83020 83020]
|}
|}
changed ec settings, halfduplex on start only switch on again.<br/>IP241 gains adjusted<!---->
When presence eventlist is received from External-UC<br/>all presence info was assigned to same PBX object.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
''Status:''
ac_codec3.cpp
      
      
=== IP241,IP222,IP232: Illegal characters when editing on a text control ===
=== PBX: Editing Config Templates impossible, if by some old firmware a strange config ended up in a User object ===


{|
{|
Line 10,293: Line 10,160:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77735 77735]
|[http://mantis.innovaphone.com/view.php?id=83033 83033]
|}
|}
E.g. when configuring call forwarding destination illegal character (0x8a) may appear in result string.<!---->
The problem was a empty hardware id (hw=""). This is now ignored.<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== IP222,IP232: Redial key does not start calling when used in call list ===
=== PBX mobility: The forking destination put in for mobility at a user should be dialed from the node of the user ===


{|
{|
Line 10,304: Line 10,171:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77746 77746]
|[http://mantis.innovaphone.com/view.php?id=83040 83040]
|}
|}
Redial key does not start calling when used in missed call list<!---->
This number was dialed from the node of the Mobility object. This was confusing, because this number was configured at the user and it was also different behaviour as with forking without mobility<br/><!--<br/>pbx_mobility.cpp<br/>--><!---->
      
      
=== ip6010/3010/1060: Ethernet transmit packet length is sometimes wrong ===
=== PBX: Standyby PBX generated alarms for missing slave registrations, even if active PBX up ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77774 77774]
|[http://mantis.innovaphone.com/view.php?id=83089 83089]
|}
|}
Sometimes old content of the tx dma descriptor was used by the ethernet MAC.<br/>Now the memory write buffers are drained before enabling the tx dma.<!---->
There should be only an alarm, if the standby PBX is active<br/><!--<br/>pbx.cpp<br/>--><!---->
''Status:''
mv78x00_drv.cpp<br/>mmu.S
      
      
=== IP222,IP232: "Lock phone" does not work ===
=== SIP: One-way audio after mutual hold on dect systems ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77778 77778]
|[http://mantis.innovaphone.com/view.php?id=83126 83126]
|}
|}
Authentication dialog for "Lock phone" could not be acknowledged.<!---->
Dect ep gets FTY_HOLD_NOTIFY but no FTY_RETRIEVE_NOTIFY.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== ip6010/3010/1060: Ethernet receive packet sometimes delayed ===
=== X.509: Avoid alarms on missing system time after reboot ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=77781 77781]
|[http://mantis.innovaphone.com/view.php?id=83137 83137]
|}
|}
Sometimes the rx descriptor are processed with the next tx event.<br/>Now the rx queue is processed completely in on interrupt.<br/><!---->
Allow 60 seconds for setting the system time before an alarm is set. Also certificates are now rejected silently, in that time.<br/><!--<br/>x509.cpp<br/>x509.h<br/>--><!---->
''Status:''
mv78x00_drv.cpp<br/>mv78x00_drv.h
      
      
= V9 Hotfix 9 (90600.11, withdrawn) =
=== SIP: Problems with presence signaling on External-UC link ===
Changes included in Version 9 hotfix9
[http://mantis.innovaphone.com/view.php?id=76991 Definition]
 
== New Features ==
 
 
   
=== PBX-SOAP: UserCall new option diversion override ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=75773 75773]
|[http://mantis.innovaphone.com/view.php?id=83177 83177]
|}
|}
To initiate calls, which are not diverted by a configured CFU.<br/><br/>Use Info[] element with type="fty-no-cf"<!---->
Missing parameters on Contact-URI in 200/OK for SUBSCRIBE(presence):<br/>  maddr<br/>  transport<br/><br/>Wrong Contact-URI in presence XML in PUBLISH.<br/><br/><!--<br/>sip.cpp<br/>sip_presence.cpp<br/>--><br/><!---->
      
      
=== SIP: Support for "Allow-Events: ccnr" and "Allow-Events: ccbs" ===
=== IP-DECT: Support for more than 32 LDAP attributes ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=76658 76658]
|[http://mantis.innovaphone.com/view.php?id=83191 83191]
|}
|}
For interoperability:<br/>Support for "Allow-Events: ccnr" and "Allow-Events: ccbs"<!---->
The IP-DECT devices support up to 256 LDAP attributes now.<br/><br/><!--<br/>Changed file:<br/>dectusers.h<br/>--><!---->
      
      
=== Phone-&gt;PBX LDAP Search returns normalised number to be dialled by phone ===
=== PBX: Support for more than 64 LDAP attributes ===


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|}
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1)Phone->PBX LDAP Search returns normalised number to be dialled by phone.<br/>2)Phone receives info about escape digits when registering at its PBX.<br/><br/>With 1) the PBX includes an object's normalized number into the LDAP search result.<br/>With 1) the phone is able to dial that normalized number.<br/>With 2) the phone is able to prefix required escape digits to the received normalized number.<!---->
The PBX supports up to 256 LDAP attributes now.<br/><br/><!--<br/>Changed file:<br/>pbx.h<br/>--> <!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
=== PBX soap method SetPresence ===
=== IP-DECT: LDAP replication alarm loop ===


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PBX soap method SetPresence added to set the presence of a PBX user.<!---->
If the LDAP replication is configured, but it can not be synchronized, there is an alarm loop. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dectusers.cpp<br/>--><!---->
      
      
=== phone ip222, ip232 : USB headsets need not to be configured anymore ===
=== Gateway: On IP24, IP302, IP305 it could happen that multiple routes disappeared if a route was deleted ===


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If an USB headset with a known signature (vendor/product id) is plugged it is automatically enabled. This is indicated by the headset symbol in the status line.<br/>"Phone/Preferences/Start Outbound Call on Electronic Hook Switch (EHS) Signal" is implied in this case because some headsets will loose state if a hoook signal is ignored. <br/><!---->
Hard to predict when this happened.<br/><!--<br/>gk.cpp<br/>--><!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
=== phone: ip222, ip232: USB headset support - Plantronics C420,  GN2000 USB - MS OC Version ===
=== Gateway: No Media received event was generated for T.38 calls, which started without actual RTP Traffic ===


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|}
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.<!---->
This happend for calls to/from Fax servers, which switch to T.38 without sending any RTP packets first.<br/><!--<br/>media.cpp<br/>--><!---->
''Status:''
checked in to 9.00, 90600
      
      
=== PBX Mobility: No-Alert checkmark as workaround for provider which do not provide Alerting ===
=== SIP: Multiple subscriptions for 'message-summary' ===


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|}
|}
Some SIP provider do not provide an Alerting signal when a mobile phone is called. This could result in no ringback signal to the caller or the min/max-alert feature not working.<br/><br/>This new checkmark provides a fake Alerting in case Progress is received<!---->
After every RAS_START another subscription for 'message-summary' is established.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
= V9 Hotfix 15 (9061078) =
Changes included in Version 9 hotfix15
[http://mantis.innovaphone.com/view.php?id=83650 Definition]
 
== New Features ==
 
 
      
      
=== show linux shutdown warning on firmware reset page ===
=== New remote control codes to be used for phone tests by soap applications ===


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|}
|}
If linux is running, a warning is now shown, that linux should be shutdowned before a firmware reset is performed.<br/><!---->
The new UserRc codes are executed only when the addressed phone is either in handset, headset or handsfree mode, i.e when calling, connected or disconnected but not when alerting:<br/>16 - change to handset mode<br/>17 - change to headset mode<br/>18 - change to handsfree mode<br/>19 - monitor mode on (add speaker to handset or headset mode)<br/>20 - monitor mode off (back to plain handset or headset mode)<br/><br/><!-- fty.h phonesig_if.cpp phonesig.cpp app_call.cpp --><!---->
      
      
=== Linux: Enable/Disable support link ===
=== Alarm/Event handling: Authentication for received remote Alarms/Events ===


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|[http://mantis.innovaphone.com/view.php?id=83603 83603]
|}
|}
Now the Linux menu is always shown and a link is provided to enable or disable the Linux support (RAM reservation). The support state is also saved in the downloaded configuration file and restored with the upload.<br/>Update: The support state is only saved in the downloaded configuration with password. The state is not saved in the configuration file with standard password or if downloaded by the update server. Please use the next or a later hotfix instead, see also fix #78836.<!---->
Allow by configuration to only accept authenticated alarms or events<br/><!--<br/>logging.cpp<br/>logging.h<br/>fault_handler.cpp<br/>fault_handler.h<br/>logging.xsl<br/>--><!---->
      
      
== Bug Fixes ==
=== PBX: Preparations for objects visible only if appropriate license installed ===
 
 
   
=== H.323: Media Negotiation problem with conferences on IP-DECT ===


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|}
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A channel was not switched to the conference after a hold/retrieve cycle<!---->
for v10<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_api.h<br/>--><!---->
      
      
=== PBX Mobility: Dialed digits could get lost, when using Opticaller ===
=== PBX: New Feature to allow registration with password to devices regardless of address filter ===


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There could be a collision of a dialed digit with media renegotiation. For example if with the first digit a media was switched to inband information from a carrier.<!---->
It is now possible to configure a flag at a device to allow a registration for this device even if there is an IP Filter which does not match. This is useful if registrations from the public internet to the PBX shall be possible. Without this feature this could be opened only for the complete PBX. Now it can be restricted to a few devices.<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_admin.cpp<br/>pbx_edit_object.xsl<br/>--><!---->
      
      
=== phone: Ring Tone Titles containing apostrophes  garble phone configuration ===
=== SoftwarePhone: Signature added ===


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|}
|}
When under "Phone/Ring Tones/Add Ring Tone" a title containing apostrophes is entered the page "Phone/User-x/Preferences" cannot be edited anymore because of a XML-Error.<!---->
The SoftwarePhone installer and the install package is signed now.<br/><br/><!--<br/>Changed file:<br/>Makefile<br/>--><!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
=== SIP: Record-Route handling on outbound subscriptions ===
=== IP-DECT: Phone book 'Use TLS' option ===


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|[http://mantis.innovaphone.com/view.php?id=84001 84001]
|}
|}
Processing of Record-Route in SUBSCRIBE response.<br/>Simply missing.<!---->
'Use TLS' option added for the central phone book search. This changes the standard port from 389 to 636 if no port is configured. The central phone book search is only available with the IP1202.<br/><br/><!--<br/>Changed files:<br/>ascom/phonebook_ldap/config_ldap.h<br/>ascom/phonebook_ldap/config_ldap.cpp<br/>ascom/phonebook_ldap/phonebook_ldap_search.cpp<br/>ascom/phonebook_ldap/phonebook_ldap.xsl<br/>--><br/><!---->
      
      
=== phone: Pickup fuction key is not displayed while all alerting calls are displayed on a Partner key ===
=== simple static logging interface ===


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|}
|}
alerting calls displayed on a Partner key are not displayed on the Pickup key.<br/>if nothing has to be displayed on the pickup key the key should not disappear but display the 'idle' label<!---->
to simplify sending of log messages, alarms and errors a simple static interface to the logging module was added. log_if::log(class serial src, const class event &amp; event) passes the given event to the primary logging module (aka LOG0). This works also with 'src' = 0.<br/><br/><!-- log_if.cpp log_if.h logging.cpp http.cpp --> <!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
=== H.323: Media Negotiation problem with transfer in Gateway (not PBX) ===
=== myPBX: Pass selected device to launcher ===


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|}
|}
Under special conditions a blind transfer happend in the Gateway could result in a call without media. This only happened if the call was transfered twice and the destination of the first transfer was a physical interface.<!---->
Needed for version 10 remote video<br/><!--<br/>client.js<br/>client_util.js<br/>--><!---->
      
      
=== PBX Twinning: When calling another (twin) phone, the call was sent to the original phone also ===
=== SIP: Announcing  "a=T38MaxBitRate:14400" in T.38 offer ===


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|}
|}
It still works, but this waiting call could be confusing<!---->
Announcing  "a=T38MaxBitRate:14400" in SDP offer for T.38<br/>since all current devices support that modem speed.<!---->
      
      
=== PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX ===
=== SIP: Set numbering plan to "ISDN/telephony" in case of "Bellcore-dr2" as Alert-Info in INVITE ===


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|}
|}
A call from a object within a node with escapes on a slave PBX was not routed to the master if the destination was within the same node and not known on this slave but was sent to the node-extern destination directly<!---->
Set numbering plan to "ISDN/telephony" in case of "Bellcore-dr2" as Alert-Info in INVITE.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== PBX Waiting: Name Id missing in calls initiated with SOAP ===
=== IP-DECT: Phone book error events ===


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|}
|}
When a Waiting Queue is used by applications to initiate outgoing calls, the name of the waiting queue should be sent with these calls as calling name. This name id was missing<!---->
The phone book module sends an error event now if the LDAP directory search fails.<br/><br/><!--<br/>Changed files:<br/>platform/config.h<br/>dect/dectusers_if.h<br/>ascom/phonebook_ldap/phonebook_ldap.h<br/>ascom/phonebook_ldap/phonebook_ldap.cpp<br/>ascom/phonebook_ldap/phonebook_ldap_request.cpp<br/>ascom/phonebook_ldap/phonebook_ldap_search.h<br/>ascom/phonebook_ldap/phonebook_ldap_search.cpp<br/>--><!---->
   
== Bug Fixes ==
 
 
      
      
=== RTP: Potential random trap when closing channels ===
=== IP-DECT: Trap during subscribing handsets ===


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Happens if there is a collision with a received packet and closing of the channel. Window for this is very small, so it should happen very rarely. Probability can increase with high load.<!---->
A trap occurs during subscribing handsets on the IP1200. This is a fix for the previous fix #80424 and fixed again.<br/><br/>Trap identification: <br/>XCPT: no 2 (TLB load)  pc 942e23d8  ra 942e23cc  va 00000000<br/><br/><!--<br/>Changed file:<br/>dectusers.cpp<br/>--><!---->
      
      
=== Web-UI: Visually separated commands ===
=== PBX Waiting: Trap on leak-check if dtmf maps are configured ===


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|}
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Visually separated commands 'Clear' and 'Save' on Maintenance/diagnostics/Events.<br/>Also 'download' and 'delete' on General/License.<!---->
Only happened if debug.xml leak check was used<br/><!--<br/>pbx_wait.h<br/>--><!---->
      
      
=== IP241,IP222,IP232: Show special symbol for call completion entries in call list ===
=== IP222 IP232 IP241: Adjust gains for better echo canceller performance ===


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Show special symbol for call completion entries in call list to separate from usual missed calls.<!---->
..<br/><!-- <br/>Use original EC mode by selecting mode 2.<br/>Max output gain reduced to avoid strong echo<br/>ADC input gain max<br/>Digital input gain set to 0db.<br/><br/>ac_phone3.cpp ac_codec.cpp --><!---->
      
      
=== IP241,IP222,IP232: Status message "Recording" not displayed ===
=== myPBX: Remove unimportant notifications ===


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Status message "Recording" needs to be displayed during recording.<!---->
The following events are not so important that the user has to be notified using a windows bubble notification.<br/>- Visibility requests<br/>- Missed calls<br/><!--<br/>client.js<br/>--><br/><!---->
      
      
=== Gateway: Wrong Media info sent for calls forwarded from a VOIP interface to another VOIP interface ===
=== IP222 IP232 IP241: Enable noise reduction ===


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|}
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This could cause wrong information to be displayed on a PBX calls page. Esspecially SRTP was not indicated.<br/><br/>Better to not send any Media Info in this case.<!---->
Enable noise reduction on all handset/handsfree/DHSG headset micro.<br/>Use  DSP code 660.27.pa.03<br/><br/>Noise reduction parameter can be tuned at<br/>http://addr/AC-DSP0/mod_cmd.xml?xsl=phone-dsp.xsl<br/><br/><!-- <br/>ac494e002ce4.660.27.pa.03.h ac494eac_494e.cpp ac_phone3.cpp ac_phone3.h phone-dsp.xsl <br/>--><!---->
      
      
=== IP241,IP222,IP232: Wrong display name in call list ===
=== Voicemail: Prevent &lt;prompt&gt;, &lt;record&gt; on incoming control calls ===


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|}
|}
Name identification of remote party is to be displayed,<br/>but name identification of diverting party was displayed instead<br/>on incoming calls list.<!---->
Turned out to cause idle-reset requests never being processed.<br/><br/>An administrative solution also exists: Watch out for the URL variable "$_noctl" in this article:<br/>http://wiki.innovaphone.com/index.php?title=Howto:Configure_the_innovaphone_Voicemail#URL_Query_String_Variables<br/><br/><!--<br/>pbx_vm.cpp<br/>--><!---->
      
      
=== phone: the call which was the active call at start of a conference was not  automatically cleared  on a release from remote ===
=== HTTPCLIENT: Allow configuration of a http authenticated URL even if the server needs no authentication ===


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|}
|}
The call which was the active call at start of a conference had to be cleared manually but the call which was on hold at start of conference was cleared automatically.<br/>Now any call will be automatically cleared when relesed from remote.<!---->
If a password was configured and the server didnt need it, a HTTP put created a file of zero length.<br/><br/><!-- httpclient_i.cpp httpclient_.h --><br/><br/><!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
=== phone_orchid: remaining call mute after remote relase for the call which was the active call at start of a conference ===
=== PBX Waiting: Trap on collision of operator connect and two-stage dialing ===


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did not happen after a remote release for the call which was on hold at start of the conference.<!---->
If two-stage dialing (Maps) is used to call a Trunk or Gateway object, the call is sent after a blockdial timeout. If an operator connected the call before this timeout happened, a trap occured.<br/><br/>For this to happen DTMF maps and operators have to be configured on the same Waiting Queue object, with is kind of unusual<br/><!--<br/>pbx_wait.cpp<br/>pbx_wait.h<br/>--><!---->
''Status:''
checked in to 10.00, 9.00
      
      
=== phone: ip222,ip232: Muting USB headset micro via microphone key did not work but MICROPHONE OFF was displayed ===
=== PBX: Call to a Trunk/Gateway was not marked correctly as external, if no connected number was received ===


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|}
|}
.<!---->
Calls from a Trunk must be explicitly marked as internal with respective connected number, otherwise they should be treated as external<br/><!--<br/>pbx.cpp<br/>pbx_mobility.cpp<br/>--><!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
=== IP241,IP222,IP232: Trap in font rendering ===
=== HTTP: Chunked transfer fails if the last 2 bytes of the chunk header are in the next tcp packet ===


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|}
|}
Some characters at beginning of line may cause trap.<!---->
Seen with the application platform as broken pipe<br/><br/><!-- http.cpp --><!---->
      
      
=== Gateway: Trap when interworking Call Completion ===
=== PBX: Registration with Name/Number did not work correctly if default device not first ===


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|}
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Trap when interworking Call Completion.<br/><br/>LOG CALL 6 A:Call    ->                        / PRI2::->*::<br/>R_CALL free error c18a59b8<br/><!---->
For a registration with name or number, the information if the PBX password shall be used was always taken from the first device regardless if this was the default device (hw-id identical to name) or not<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== phone: ip241: memory leak when ip230x extension module is attached and used  ===
=== PBX: Partial Rerouting was prohibited in Alerting State (CFNR) ===


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|}
|}
one packet lost per keystroke on extension module<!---->
This was done under the assumption the partial rerouting is not supported as CFNR by public networks.<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== PBX: Dynamic group function keys did not work with non-ascii characters in group name ===
=== PBX Waiting: Call to operator with Twin Phone Checkmark did not work anymore ===


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Name was not correctly converted to internal utf-8 representation<!---->
Collateral damage of<br/><br/>fix: #82858: PBX Waiting: Evaluate Busy on ... Calls for calls to an operator<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
      
      
=== phone: ip222,ip232: trace USB media data only if explicitely requested ===
=== SoftwarePhone: Product string ===


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|}
|}
use "config add USB-HOST AUDIO /trace" if data is needed<!---->
The product string is changed, used e.g with the PBX registration.<br/><br/><!--<br/>Changed file:<br/>portkernel.cpp<br/>--><!---->
      
      
=== H.323: Potential Trap in case of special malformed RAS registration ===
=== phone: provide complete dialog info to a phoneapp ===


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|}
|}
A NULL pointer access could happen<!---->
Both group indications and dialog infos are signaled via a group indication facility. For dialog infos the parked_to_alerting member was overloaded to provide the info as expected by the existing phoneapp.<br/>Now the parked_to_alerting member is passed to a phoneapp as received.<br/><br/><!--<br/>phonesig.cpp phonesig_if.h app_reg.cpp app_gui.cpp app_fkey.cpp app_disp.cpp<br/>--><!---->
      
      
=== SIP: Removed b=TIAS attribute from media description ===
=== IP222 IP232 IP241: Force same speed of the switch ports for 1000M/100M scenarios (configuration option added) ===


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|}
|}
Causes trouble on some SIP providers (neotel.at)<!---->
In scenarios with frequent transistions of the attached PC to sleep  renegotiating the link speed may be undesired. For this case the force same speed mechanism can be disabled.<br/><br/>Other changes:<br/>1000M is only changed to 100M if the other port runs at 100M. The previous version changed from 1000M to 100M if the other port runs at 100M or 10M.<br/><br/>The statistics can be collected from the PC port or from the LAN prot or from both.<br/><br/>Packet forwarding on the PC port is disabled if the port is down to avoid misleading collision counter behaviour.<br/><br/><!-- orchid_drv.cpp orchid_drv.h orchid_drv.xsl orchid_drv_s.xsl --><br/><br/><!---->
      
      
=== DHCP server leases with hostnames containing non-ascii latin1 characters (for example Umlauts) could not be displayed ===
=== CDR fixes for external call detection ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=78382 78382]
|[http://mantis.innovaphone.com/view.php?id=84211 84211]
|}
|}
In pre-V9 firmware hostnames were stored latin1-encoded. Names contaning non-ascii latin1 charaters must be converted to UTF8 before display.<!---->
The type="ext" attribute was not set reliably. Additionaly an attribute pseudo was added to the <user/> tag to indicate the type of object the CDRis created for.<br/><!--<br/>pbx_api.h<br/>pbx.cpp<br/>pbx_exec.cpp<br/>pbx_wait.cpp<br/>pbx_bc.cpp<br/>pbx_mobility.cpp<br/>--><!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
=== MyPBX: Changing password did not work any more ===
=== http client : authentication was not retried after a failure when the offending request was repeated in the same session ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=78391 78391]
|[http://mantis.innovaphone.com/view.php?id=84217 84217]
|}
|}
Collateral damage of fix #75453: PBX/IP-DECT: User password length<!---->
When a httpclient user repeated a failing request in the same session the authentication was not tried again. Thus a change of the client side URL password or a change of the server side password had no effect until a new session was started.<br/><br/><!-- httpclient_i.cpp --> <!---->
      
      
=== phone: ip222, ip232: USB headset sometimes mute after disconnect ===
=== IP6000: Prevent blinking error LED on old IP6000 with HW-Build &lt;110 ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=78414 78414]
|[http://mantis.innovaphone.com/view.php?id=84227 84227]
|}
|}
depending on isochronous transfer state at disconnect time. seems to happen more often with relatively cheap headsets.<!---->
Conference DSP driver was started on old hardware that doesnt support the conference DSP<br/><!-- adsp2191_drv.cpp --><!---->
      
      
=== SIP: Be save against sudden death of SIP caller ===
=== IP222 IP232 IP241: pressing speaker key when phone is in handset mode switches to handsfree mode instead to toggle monitor mode ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=78460 78460]
|[http://mantis.innovaphone.com/view.php?id=84297 84297]
|}
|}
Lifetime of an INVITE trasnaction is not limited by any timeout<br/>after provisional response has been send/received.<br/>Sudden death of a caller make calls hang forever.<br/>Now overall lifetime of an INVITE server transaction is limited to 3 minutes.<br/>After expiration fimnal reject response is sent and call is released.<!---->
handset/headset plus speaker is not supported<br/><br/><!-- codec.h ac_codec3.h ac_codec3.cpp phonesig_if.h phonesig.h phonesig.cpp app_ctl.cpp --> <!---->
      
      
=== IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax failure after transfer ===
=== myPBX: Inconsistent display of group monitoring rights ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=78487 78487]
|[http://mantis.innovaphone.com/view.php?id=84350 84350]
|}
|}
channel was reconfigured on remote switch to T38, without a new activate fax relay command<!---->
In the visibility settings groups were not displayed if the membership is "dynamic out". But members of that group still have monitoring rights.<br/><!--<br/>pbx_client.cpp<br/>--><!---->
''Status:''
ac_dsp3.cpp
      
      
=== PBX Boolean: Avoid unnecessary load by boolean monitoring ===
=== phone: when scrolling directory search results sometimes one of  the numbers of a contact was not displayed ===


{|
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|[http://mantis.innovaphone.com/view.php?id=78504 78504]
|[http://mantis.innovaphone.com/view.php?id=84362 84362]
|}
|}
A call was sent to the monitoring endpoint every 10s<!---->
the tag characters assigned to the different numbers were not included in sort order. <!---->
      
      
=== IP241,IP222,IP232: No RTP send on outgoing call in some cases ===
=== phone: status messages for outbound external calls were sometimes garbled on display ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=78550 78550]
|[http://mantis.innovaphone.com/view.php?id=84365 84365]
|}
|}
No RTP send on outgoing call in some cases<!---->
happened when for a preceeding outbound internal call in alerting state presence info was displayed<br/><br/><!-- forms.h forms_gen.cpp forms_color.cpp app_ctl.h app_call.cpp app_disp.cpp --><!---->
      
      
=== IP6000: Traps in DSP driver under high load ===
=== IP0010: DSP didnt start with build 9061044 ===


{|
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|[http://mantis.innovaphone.com/view.php?id=78591 78591]
|[http://mantis.innovaphone.com/view.php?id=84388 84388]
|}
|}
under high load timing may change. Checks in driver relaxed to take this into account.<!---->
<!--ac_491.cpp adsp2191:drv.cpp--><!---->
      
      
=== PBX-SOAP: Limit on size for Admin call too small ===
=== Page Ldap/Replicator/Status didn't display in WebKit ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=78592 78592]
|[http://mantis.innovaphone.com/view.php?id=84400 84400]
|}
|}
Only 4k was allowed<!---->
Page Ldap/Replicator/Status didn't display in WebKit<br/><!--<br/>ldaprep_status.xsl<br/>--><!---->
      
      
=== SoftwarePhone: Default ToS value for RTP ===
=== phone_orchid: displaying both  dialed and connected number may be misleading, it's better to omit dialed number ===


{|
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|[http://mantis.innovaphone.com/view.php?id=78593 78593]
|[http://mantis.innovaphone.com/view.php?id=84422 84422]
|}
|}
Now the default ToS value for RTP packets is 0xB8.<!---->
when for example 022222222 was dialed and the network reported a connected number 03022222222 the display info "022222222 -> 03022222222" looked like a transfer.<br/><br/><!-- forms_color.cpp --><br/>  <!---->
      
      
=== PBX Filter: Maximum length increased from 13 to 17 digits ===
=== Hide LDAP Server Password For Viewer Accounts ===


{|
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|[http://mantis.innovaphone.com/view.php?id=78594 78594]
|[http://mantis.innovaphone.com/view.php?id=84557 84557]
|}
|}
13 digits could be too little if an international number including a prefix of an external line was to be checked<!---->
Was accessible for viewers<br/><!--ldapsrv.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Wrong call direction indication on PARTNER fkey ===
=== phone: enable directory search function key also in connected state ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=78642 78642]
|[http://mantis.innovaphone.com/view.php?id=84559 84559]
|}
|}
When partner calls someone else it my displayed as if partner is called.<br/>(Only if option "Show connected Party in busy State" is enabled)<br/><!---->
it's sometimes useful to browse the directories for a number while talking.<br/><br/><!-- app_fkey.cpp --> <!---->
      
      
=== IP222 IP232 IP241: Sporadic noisy sidetone ===
=== myPBX: Only send one command at a time ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=78649 78649]
|[http://mantis.innovaphone.com/view.php?id=84568 84568]
|}
|}
Analog and digital sidetonegain were enabled,now only the analog sidetonegain is used.<!---->
Queue commands instead of sending overlapping commands. This limits the number of open AJAX connections to two.<br/><!--<br/>client.js<br/>--><!---->
''Status:''
ac_codec3.cpp
      
      
=== Gateway: Handling of blind transfer gets stuck in routing ===
=== H.323: Accidential fallback to slowstart if faststart response received in PROGRESS ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=78709 78709]
|[http://mantis.innovaphone.com/view.php?id=84601 84601]
|}
|}
Routing of transfer call stops with: reason='incomplete'<br/>But during transfer no dialing digits will follow.<!---->
Only happened in some H.323/SIP interop scenarios<br/><!--<br/>h323sig.cpp<br/>--><!---->
      
      
= V9 Hotfix 10 (90600.12) =
=== IP222 IP232 IP241: Updated gain and equalizer setting ===
Changes included in Version 9 hotfix10
[http://mantis.innovaphone.com/view.php?id=78895 Definition]
 
== New Features ==
 
 
   
=== Linux: Enable/Disable configuration ===


{|
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|[http://mantis.innovaphone.com/view.php?id=84605 84605]
|}
|}
With the new feature #78786 the configuration is only saved in the downloaded configuration file with password. Now the information is also included in the configuration file with standard password and in the file downloaded by the update server.<!---->
Tuned for high MOS values according to ETSI ES202737 ES202738 ES202739 ES202740<br/><!-- ac_codec3.* platform/filter/* --><!---->
      
      
== Bug Fixes ==
=== IP0010 IP1060 IP3010 IP6010 IP22 IP24 IP28 IP302 IP305 IP800 IP6000: Minifirmware not shown on LED ===
 
 
   
=== PBX: Trap if enabling Unknown Registrations ===


{|
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|[http://mantis.innovaphone.com/view.php?id=84616 84616]
|}
|}
Collateral Damage from<br/><br/>fix: #77261: Phone->PBX LDAP Search returns normalised number to be dialled by phone<!---->
Minifirmware should blink long green short red. This happened only after a firmware/bootcode update.<br/><br/><!-- ip6010.cpp ip24.cpp ip800.cpp ip6000.cpp--><!---->
      
      
= V9 Hotfix 11 (90600.14) =
=== phone: ip222, ip232: status stage added to all USB control transfers ===
Changes included in Version 9 hotfix11
[http://mantis.innovaphone.com/view.php?id=79001 Definition]
 
== New Features ==
 
 
   
=== HTTP-Client: MD5-sess authentication ===


{|
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|[http://mantis.innovaphone.com/view.php?id=77773 77773]
|[http://mantis.innovaphone.com/view.php?id=84617 84617]
|}
|}
HTTP Digest Authentication with alogrithm=MD5-sess.<br/>Choose the first supported "WWW-Authenticate" line from 401 response headers.<br/><br/>Needed for new versions of IIS.<br/><br/><!---->
the status stage is mandatory as well for IN as for OUT control tranfers<br/><br/><!-- ohci.cpp ohci.h ohci_defs.h --><!---->
''Status:''
http://wiki.innovaphone.com/index.php?title=Support:DVL-Feature_Requests#HTTP_Client
      
      
=== Phone: Possibility to reject incoming SIP calls with customized reason phrase ===
=== H.323: More information on "Unexpected Message" event ===


{|
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|[http://mantis.innovaphone.com/view.php?id=84699 84699]
|}
|}
E.g. "SIP/2.0 480 Do not disturb" instead of  "SIP/2.0 480 Temporarily unavailable"<!---->
The message type and the state for which this message was unexpected is needed to find out what the problem is.<br/><!--<br/>h323sig.cpp<br/>--><!---->
      
      
=== Active Directory Replication: Editfield for Poll Timer added ===
=== SIP: Trap during channel handling ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=78631 78631]
|[http://mantis.innovaphone.com/view.php?id=84800 84800]
|}
|}
If change notifications cannot be received from an AD, a poll timer can be specified. A re-replication is going to take place after the poll timer expired.<!---->
Rare trap when re-assigning channels.<br/><br/><!--<br/>sip.cpp/h<br/>medialib.h<br/>--><!---->
      
      
=== X.509: Support for PKCS#12 files ===
=== IP232,IP222,IP241: Rendering errors when trying to use backround image with indexed colors ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=78820 78820]
|[http://mantis.innovaphone.com/view.php?id=84814 84814]
|}
|}
Support for certificate import using password encrypted PKCS#12 files.<br/><br/>Currently the following encryption types are supported:<br/><br/>pbeWithSHAAnd128BitRC4 (1.2.840.113549.1.12.1.1)<br/>pbeWithSHAAnd40BitRC4 (1.2.840.113549.1.12.1.2)<br/>pbeWithSHAAnd3-KeyTripleDES-CBC (1.2.840.113549.1.12.1.3)<br/>pbeWithSHAAnd128BitRC2-CBC (1.2.840.113549.1.12.1.5)<br/>pbewithSHAAnd40BitRC2-CBC (1.2.840.113549.1.12.1.6)<br/><!---->
PNG mode "indexed colors" is not supported.<br/>Trying to use an indexed color png leaves phone screen in bad shape.<br/><br/><!--<br/>read_png.cpp<br/>--><!---->
      
      
=== myPBX: Pass language code to reporting ===
=== NAT: Don't forward DNS requests from public network ===


{|
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|[http://mantis.innovaphone.com/view.php?id=78874 78874]
|[http://mantis.innovaphone.com/view.php?id=84842 84842]
|}
|}
For localization of the call list.<!---->
As kind of denial of service attack, bursts of incoming DNS requests were seen. The nat process was forwarding these requests to the public DNS. This is a useful function for DNS requests from the private network, but not for requests from the public network.<br/><br/>These DNS requests are now discarded<br/><!--<br/>socket.h<br/>udp4.cpp<br/>udp.cpp<br/>nat.cpp<br/>--><!---->
      
      
=== Phone: Show number of voice messages on label of MWI fkey ===
=== H.323: Media Negotiation did not work for call with reverse media and media response in CALL-PROC ===


{|
{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=78894 78894]
|[http://mantis.innovaphone.com/view.php?id=84848 84848]
|}
|}
Use "%u Message(s)" as label text for active state of MWI fkey.<!---->
Caused SIP interop problems with CUCM<br/><!--<br/>h323sig.cpp<br/>--><!---->
      
      
=== IP-DECT: Cisco SIP conferencing ===
=== Maximum LDAP PDU Size Too Small ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=84851 84851]
|}
|}
Cisco proprietary softkey events implemeneted to start a 3pty conference<!---->
Maximum LDAP PDU Size Too Small. Now internal maximum allocation unit plus a bit for encoding overhead.<br/><br/><!--<br/>ldapasn1.cpp<br/>--><!---->
      
      
=== IP-DECT: Feature Code for calling predefined service URIs ===
=== PBX Broadcast: Call to group members was not cleared when CFNR was executed ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79028 79028]
|[http://mantis.innovaphone.com/view.php?id=84857 84857]
|}
|}
User for Cisco compatibility<!---->
This was a collateral damage from<br/>fix: #79549: PBX Waiting/Broadcast: Incomplete CDRs if CFNR configured on object<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== Cisco Softkey Features, Status updates and Park ===
=== IPVA: V10 Code Merge Aiming To Stabilize FW Upload To DRAM ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79030 79030]
|[http://mantis.innovaphone.com/view.php?id=84862 84862]
|}
|}
suported_mask needs more flags<br/>Softkey Feature Park<!---->
Addresses a problem in conjunction with innovaphone's automated test environment<br/><!--<br/>ipva.cpp<br/>vlance_drv.cpp<br/>ide_drv.cpp<br/>--><!---->
      
      
=== IP-DECT: Cisco Park ===
=== PBX: Filters should only be applied to calls with media channels ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79032 79032]
|[http://mantis.innovaphone.com/view.php?id=84892 84892]
|}
|}
Send Park softkey event<!---->
A join group operation could be prohibited with filters. This was unexpected.<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== IP-DECT: Forward Cisco Status Line Updates related to a call ===
=== phone: ip222, ip232: raise alarm if an USB Headset does not respond on USB bus anymore ===


{|
{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=79036 79036]
|[http://mantis.innovaphone.com/view.php?id=84893 84893]
|}
|}
.<!---->
<!-- keys_ip222.cpp, ohci.cpp, ohci.h --><br/><!---->
      
      
=== IP-DECT: SRTP for an OEM device ===
=== SIP: Fix for media negotiation on calls re-routed from TONE interface to outbound SIP ===


{|
{|
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|-
|Id
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|[http://mantis.innovaphone.com/view.php?id=79061 79061]
|[http://mantis.innovaphone.com/view.php?id=84932 84932]
|}
|}
Now SRTP is configurable for an OEM device.<!---->
Helps on gateway interfaces configured for media-relay with exclusive codec.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== IP-DECT: Forward Cisco Status Line Updates not related to a call ===
=== IP-DECT: Call transfer timer ===


{|
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|-
|Id
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|[http://mantis.innovaphone.com/view.php?id=79104 79104]
|[http://mantis.innovaphone.com/view.php?id=84951 84951]
|}
|}
.<!---->
The call transfer timer is stopped with the call proceeding event now. This fixes a call transfer to e.g. a mobile user with a delayed alert.<br/><br/><!--<br/>Changed files:<br/>dectradio_call.cpp<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp<br/>--><!---->
      
      
=== Merge of v10 changes: Dynamic Timer Management/oem serial no ===
=== IP-DECT: Reverse phone book ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79127 79127]
|[http://mantis.innovaphone.com/view.php?id=84953 84953]
|}
|}
Better power management on some platforms<br/>Display of oem serial no<br/>stanard behaviour of delete on NULL<!---->
The reverse phone book does not work till hotfix 11. This is fixed again now.<br/><br/><!-- Changed files:<br/>dectradio.cpp<br/>dectradio_mastercall.cpp<br/>--><!---->
      
      
=== Merge of v10 changes: Memory Management ===
=== SIP: Loosing remote IP addresses when DNS becomes temorarily unavailable ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79129 79129]
|[http://mantis.innovaphone.com/view.php?id=84991 84991]
|}
|}
for OEM needs<!---->
Loosing remote IP addresses when DNS becomes temorarily unavailable.<br/>Results in interfaces without remote ip addresses.<br/><br/><!--<br/>sip.cpp/h<br/>--><br/><!---->
      
      
=== Merge of v10 changes: Firmware File check when updating improved ===
=== PBX Routing: A CFNR on a slave gateway object redirecting the call to the master failed ===


{|
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|-
|Id
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|[http://mantis.innovaphone.com/view.php?id=79130 79130]
|[http://mantis.innovaphone.com/view.php?id=85017 85017]
|}
|}
for OEM needs<!---->
This was a collateral damage of<br/><br/>fix: #77874: PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== Merge of v10 changes: More ciphers ===
=== SIP: Switch to T.38 did not work when interworking with H.323 slowstart (XCAPI) ===


{|
{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=79134 79134]
|[http://mantis.innovaphone.com/view.php?id=85047 85047]
|}
|}
for OEM SRTP and other Certificate formats<!---->
Switch to T.38 did not work when interworking with H.323 slowstart (XCAPI).<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== Merge of v10 changes: Internal interfaces enhanced ===
=== IP222 IP232 IP241: Headet volume adjust added, re-enable noise reduction in headset mode ===


{|
{|
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|-
|Id
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|[http://mantis.innovaphone.com/view.php?id=79135 79135]
|[http://mantis.innovaphone.com/view.php?id=85058 85058]
|}
|}
for OEM support<!---->
Headset volume adjust can be done with the webinterface at<br/> http://addr/dsp.xml<br/><br/>This feature can be used to adapt to different DHSG headsets.<br/><br/>The noise reduction in headset mode is re-enabled to avoid sporadic noise with <br/><br/><!-- ac_phone3.* ac_codec3.* ac_dsp3.mak --><!---->
      
      
=== Merge of v10 changes: ZipZip Tone definition updates ===
=== SIP: Memory leak ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79136 79136]
|[http://mantis.innovaphone.com/view.php?id=85083 85083]
|}
|}
should overlay normal voice<!---->
Memory leak.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== Merge of v10 changes: Library update ===
=== SIP: Fix for STUN problems ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79140 79140]
|[http://mantis.innovaphone.com/view.php?id=85118 85118]
|}
|}
.<!---->
Try alternative STUN server address if first fails and another was gathered by DNS.<br/><br/><!--<br/>sip.cpp/h<br/>siptrans.cpp/h<br/>--><!---->
      
      
=== Merge of v10 changes: permit DHCP mode change between client/disabled without reset ===
=== IP232,IP222,IP241: Do not hide configured fkey if neither text nor icon is displayed ===


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|[http://mantis.innovaphone.com/view.php?id=79152 79152]
|[http://mantis.innovaphone.com/view.php?id=85211 85211]
|}
|}
for OEM support<!---->
Do not hide configured fkey if neither text nor icon is displayed.<br/>E.g. Call Forwarding fkey with CF destination but no label text.<br/><br/><!--<br/>forms_color.cpp<br/>--><!---->
      
      
=== ASN.1 BER: Support for indefinite-length encoding ===
=== PBX: Slave with non-ASCII PBX name did not register at master ===


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|[http://mantis.innovaphone.com/view.php?id=79260 79260]
|[http://mantis.innovaphone.com/view.php?id=85235 85235]
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needed for PKCS#12 import<br/>only decoding<br/><!---->
Error in utf-8 to unicode convertion in this case<br/><!--<br/>pbx.cpp<br/>--><!---->
   
= V9 Hotfix 16 (9.061101) =
Changes included in Version 9 hotfix16
[http://mantis.innovaphone.com/view.php?id=85032 Definition]
 
== New Features ==
 
 
      
      
=== ASN.1 BER: Support for constructed octet strings ===
=== Media: Preparing for G.722.1 ===


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|[http://mantis.innovaphone.com/view.php?id=85316 85316]
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needed for PKCS#12<br/>only decoding, only one level of nesting<!---->
Decoding SDP containing G.722.1<br/><br/>\tv=0<br/>\to=- 5140 5141 IN IP4 10.138.6.91<br/>\ts=-<br/>\tc=IN IP4 10.138.6.91<br/>\tt=0 0<br/>\tm=audio 50000 RTP/AVP 122 8 0 18 121 101<br/>\ta=rtpmap:122 G7221/16000<br/>\ta=fmtp:122 bitrate=32000<br/>\ta=rtpmap:121 L16/16000<br/>\ta=rtpmap:18 G729/8000<br/>\ta=fmtp:18 annexb=no<br/>\ta=rtpmap:101 telephone-event/8000<br/>\ta=fmtp:101 0-15<br/><br/><!--<br/>channel.cpp/h<br/>sdpmsg.cpp<br/>--><!---->
      
      
=== IP810: Config Wizard ===
=== SIP: New config file option /tls-unchecked ===


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|[http://mantis.innovaphone.com/view.php?id=85449 85449]
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Mechanism needed to add platform specific files (IP810 uses same firmware as IP6010, IP3010, ...)<!---->
New config file option /tls-unchecked disables validation of remote server certificate.<br/><br/><!--<br/>sip.cpp/h<br/>siptrans.cpp<br/>--><!---->
      
      
=== Merge of v10 changes: Support for SIP specific phone features ===
=== IP-DECT: Configuration changes for OEM device ===


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|[http://mantis.innovaphone.com/view.php?id=79306 79306]
|[http://mantis.innovaphone.com/view.php?id=85522 85522]
|}
|}
for OEM support<!---->
For an OEM device the configuration update behaviour is changed.<br/><br/><!--<br/>Changed files:<br/>dectmaster.h<br/>dectmaster.cpp<br/>--><br/><!---->
''Status:''
checked in to 9.00
      
      
=== phone: permit to select the directories to be searched on indirect dialing ===
=== SIP: New config file option /fixed-contact-addr ===


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if there is for example a huge external directory used for inbound name resolution and dialing is restricted to internal partners it may be hard to find internal numbers via combined directory search.<br/><br/>  config add PHONE DIR-UI /dial-mask <selection><br/><br/>selection:<br/>  0x02 - local directory<br/>  0x04 - PBX directory<br/> 0x08 - external directory<br/><br/> 0x06 = local + PBX<br/>  0x0E = local + PBX + external<br/><!---->
New config file option /fixed-contact-addr to keep SIP client from changing it's Contact address into public address of NAT mapping after registration.<br/>(RFC-3581 Symmetric Response Routing)<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== IP-DECT: Signal waiting calls to handset ===
=== IP-DECT: DECT security (3) ===


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Was only acoustic information.<br/><br/><!--<br/>Changed files:<br/>fty.h<br/>fty.cpp<br/>dectradio_dectcall.cpp<br/>--><!---->
Some changes in preparation for the new feature "DECT security". This feature will be available later (IP1202).<br/><br/><!--<br/>Changed files:<br/>common/interface/signal.h<br/>common/interface/signal.cpp<br/>ascom/mobmaster/cryptomaster.h<br/>ascom/mobmaster/cryptomaster.cpp<br/>ascom/mobmaster/dectmobmaster.h<br/>ascom/mobmaster/dectmobmaster.cpp<br/>dect/master/dectmaster.h<br/>dect/master/dectmaster.cpp<br/>dect/radio/dectradio.h<br/>dect/radio/dectradio.cpp<br/>dect/localusers/dectlocalusers.h<br/>dect/localusers/dectlocalusers.cpp<br/>ascom/httpfiles/asc_dectusers.xsl<br/>ascom/httpfiles/_inn_asc_dectusers.xsl<br/>--><br/><!---->
      
      
=== IP-DECT: LDAP phonebook (IP1202) ===
=== Phones: New config file option /recording-without-remote-party-info ===


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The new feature LDAP phone book for the IP1202 is added now.<br/><!--<br/>Changed files:<br/>config_options.h<br/>config_options.cpp<br/>ascom/ascom_1202.mak<br/>ascom/httpfiles/_inn_dect.xml<br/>dect_comcerto/platform/config.h<br/><br/>New files:<br/>ascom/phonebook_ldap/phonebook_ldap.mak<br/>ascom/phonebook_ldap/phonebook_ldap.h<br/>ascom/phonebook_ldap/phonebook_ldap.cpp<br/>ascom/phonebook_ldap/phonebook_ldap.xsl<br/>ascom/phonebook_ldap/phonebook_ldap.xml<br/>ascom/phonebook_ldap/phonebook_ldap_hdr.xml<br/>ascom/phonebook_ldap/phonebook_ldap_request.h<br/>ascom/phonebook_ldap/phonebook_ldap_request.cpp<br/>ascom/phonebook_ldap/phonebook_ldap_search.h<br/>ascom/phonebook_ldap/phonebook_ldap_search.cpp<br/>ascom/phonebook_ldap/phonebook_ldap_entry.h<br/>ascom/phonebook_ldap/phonebook_ldap_entry.cpp<br/>ascom/phonebook_ldap/config_ldap.h<br/>ascom/phonebook_ldap/config_ldap.cpp<br/>ascom/phonebook_ldap/config_dl.h<br/>ascom/phonebook_ldap/config_dl.cpp<br/>--><!---->
For interop of recording feature with VOXTRON application.<br/>VOXTRON application gets confused by Diversion header in INVITE.<br/><br/>   config change PHONE SIG /recording-without-remote-party-info<br/><br/><!--<br/>phonesig.cpp/h<br/>--><!---->
      
      
=== IP-DECT: Three party conference with innovaphone PBX ===
=== phone_orchid: configuration prameters to adjust microphone and speaker volume ===


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|}
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Now it is possible to make a three party conference with DECT handsets with an innovaphone PBX (an innovaphone device with the CONF interface). The conferencing unit must be configured in the DECT master. The conference call is established with the feature code 'R' + '3'.<br/>This fix also includes a rework of the DECT radio module. It can handle more than one waiting or hold call now.<br/><br/><!--<br/>Changed files:<br/>common/style/work.css (innovaphone)<br/>common/interface/channel.h<br/>common/interface/channel.cpp<br/>common/interface/signal.h<br/>common/interface/signal.cpp<br/>dect/dectdisplay_if.h<br/>dect/master/dectmaster.h<br/>dect/master/dectmaster.cpp<br/>dect/master/dectmaster.xsl<br/>dect/localusers/dectlocalusers.h<br/>dect/localusers/dectlocalusers.cpp<br/>dect/radio/dectradio.mak<br/>dect/radio/dectradio.h<br/>dect/radio/dectradio.cpp<br/>dect/radio/dectradio_calls.xsl<br/>ascom/display/dectdisplay.h<br/>ascom/display/dectdisplay.cpp<br/>ascom/httpfiles/asc_dectmaster.xsl<br/>ascom/httpfiles/_inn_asc_dectusers.xsl<br/>ascom/httpfiles/_inn_asc_dectmaster.xsl<br/>kirk/display/display.h<br/>kirk/display/display.cpp<br/><br/>New files:<br/>dect/radio/dectradio_call.h<br/>dect/radio/dectradio_call.cpp<br/>dect/radio/dectradio_fty_call.h<br/>dect/radio/dectradio_fty_call.cpp<br/>dect/radio/dectradio_handover_in.h<br/>dect/radio/dectradio_handover_in.cpp<br/>dect/radio/dectradio_handover_out.h<br/>dect/radio/dectradio_handover_out.cpp<br/>dect/radio/dectradio_reg.h<br/>dect/radio/dectradio_reg.cpp<br/>dect/radio/dectradio_uninit_reg.h<br/>dect/radio/dectradio_uninit_reg.cpp<br/>dect/radio/dectradio_dectcall.h<br/>dect/radio/dectradio_dectcall.cpp<br/>dect/radio/dectradio_mastercall.h<br/>dect/radio/dectradio_mastercall.cpp<br/>dect/radio/dectradio_confcall.h<br/>dect/radio/dectradio_confcall.cpp<br/>--><!---->
These parameters set a volume correction factor which is applied at any volume level. Parameter changes are applied immediately even in an active call. <br/> <br/>config add AC-DSP0 HEADSET /mic-volume <mic-adjust> /spk-volume <spk-adjust><br/>config write<br/>config activate<br/><br/>   -20 <= <*-adjust> <= +20<br/><br/> <br/><!---->
      
      
=== Gateway: 'cn' attribute for test interfaces ===
=== IP-DECT: Radio reconnect handling with OEM PBX ===


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|}
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The 'cn' attribute is included in the test interfaces SIG0/1 for internal tests now.<br/><br/><!--<br/>Changed file:<br/>common/service/dummysig/dummysig.cpp<br/>--><!---->
The handling for the endpoint location update in combination of a radio reconnect is changed. This fixes the base station behaviour with an OEM PBX.<br/><br/><!--<br/>Changed file:<br/>dectmaster.cpp<br/>--><!---->
      
      
== Bug Fixes ==
=== ip1202: support capture of raw ethernet packets exchanged between MSP (aka DSP) and the  firmware ===
 
 
   
=== IP222/IP232/IP241: Ethernet link configuration ===


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|}
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In some case ethernet link configuration and display didnt work.<br/><br/>Now the autoneg status is shown if available, and the phy status if no autoneg didnt complete.<br/><br/>An option to use autoneg with fixed speeds and a link trace option (/ltrace) are added.<!---->
All exchange between the firmware running on the ACP (Application Command Processor) and the firmware running on the MSP (Media Strem Processor) is in ethernet packet format. The Mindspeed support prefers this trace format.<br/>The capture is enabled via <br/>   config add MSP0 /mtrace<br/>It includes as well command and RTP data packets and thus duplicates the RTP packets traced by the general "All TCP/UDP Traffic" option. <!---->
''Status:''
orchid_drv.cpp<br/>orchid_drv.h<br/>orchid_drv.xsl
      
      
=== Media Relay: Don't send dummy RTP data on incoming calls ===
=== HTTP-Client: Requests with specified credentials ===


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|[http://mantis.innovaphone.com/view.php?id=86133 86133]
|}
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Dummy RTP data is sent just in case a NAT router is within the media path to set a UDP mapping in case both legs of the call contain a NAT router.<br/><br/>An example for such a situation is a call coming from a SIP provider thru a NAT router to the PBX, which forwards the call back out to the SIP provider. The NAT router won't get RTP data from inside to set the mappings.<br/><br/>The dummy RTP was sent to all legs of the call, but it is better to send it to outgoing call legs only, because endpoints calling in may turn off a local ringback tone when receiving dummy RTP<!---->
Additional function for OEM httpclient::auth_request<br/>It uses the specified credentials and ignores the username and password from the configuration.<br/><!--<br/>Files:<br/>common/interface/httpclient.h<br/>common/interface/httpclient.cpp<br/>common/protocol/httpclient_i.h<br/>common/protocol/httpclient_i.cpp<br/>common/protocol/httpclient_session.cpp<br/>test/os/test_httpclient.cpp<br/><br/>Tests:<br/>test/10.00/unit-tests/httpclient/authentication/300-get-specified-basicauth.xml<br/>--><!---->
      
      
=== H.323/SIP: Avoid delayed SDP within outgoing calls as far as possible ===
=== Phones: New config option "Allow User Settings at Phone" ===


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If media renegotiation is needed, to one side of the call an request for a media proposal (in SIP terms, this is an INVITE without SDP) is sent. The media proposal (in SIP terms SDP offer) is then forwarded to the other side.<br/><br/>The request for an offer should if possible not sent with the initial call, because there is equipment which does not handle 'delayed SDP'<!---->
Allow changing of User Settings even if 'Protect Configuration at Phone' is activated.<br/><br/><!--<br/>phone_protect.xsl<br/>phone_admin_if.h<br/>phone_admin_ui.h<br/>phone_admin_ui.cpp<br/>app_gui.cpp<br/>phone_user_config.cpp<br/>--><br/><br/><!--<br/>See also: http://class.innovaphone.com/moodle2/mod/forum/discuss.php?d=1795<br/>--><!---->
      
      
=== Media Relay: Don't terminate T.38 protocol in media relay, forward transparently ===
=== IP-DECT: New hidden Radio option 'Max RTP streams' ===


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This should add robustness and reduce CPU load<!---->
The new hidden option 'Max RTP streams' is added to the IP-DECT Radio module. The option is only visible for an OEM device, but can be used with config change command ("/max-rtp-streams <count>"). The feature is useful to limit the RTP streams for radios connected to the IP-DECT Master with a low data bandwidth. Conference calls are not limited with this feature.<!---->
      
      
=== Flash Directory: Display Error Messages ===
=== Gateway: New config option "No blind transfer" ===


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Flash Directory: Display Error Messages<br/>Shedding light on replication problems.<br/><br/>currently only in 10.00<!---->
New config option "No blind transfer" to keep Gateway from handling blind transfer requests.<br/>If set blind transfer requests are passed through.<br/>Handling is performed at the next signaling hop.<br/><br/><!--<br/>relay.cpp/h<br/>relay_general.xsl<br/>--><!---->
   
== Bug Fixes ==
 
 
      
      
=== RTP: No check for changed media destination because of received multicast ===
=== phone: ip222, ip232: USB Headset could not be disabled via Menu or by Headset Function in Enable mode ===


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When receiving RTP normally a check is done if the source of the RTP is the same as we are sending to. If this is not the case, we assume the destination of the RTP is behind a NAT router and we change the destination address to the source address of the received RTP.<br/><br/>This does not make sense for received multicast.<!---->
sometimes the user want's to use the phone as if no headset is connected, i.e. when for example redial key is pressed after a number has been entered or a list entry has been selected the call should be started in handsfree mode and not in headset mode.<br/>If now the headset is disabled via menu or the headset(enable) function key all headset functions are completely disabled and no calls are directed to the headset, the status bar displays an icon indicating the disabled state.   <br/><br/><!--<br/>phone_user.h<br/>phone_user.cpp<br/>app_ctl.h<br/>app_ctl.cpp<br/>app_disp.cpp<br/>app_form.cpp<br/>app_fkey.cpp<br/>app_gui.cpp<br/>phone_user_if.h <br/>forms.h<br/>forms_color.cpp<br/>icons.mak<br/>headset_disabled.png<br/>ac_codec_3.cpp<br/>--><br/><!---->
      
      
=== IP22 IP24 IP28 IP305: Faxempfang gelegentlich gestört ===
=== H.323: Interop Problem with CUCM SIP for Transfer ===


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DSP erhält zum falschen Zeitpunkt ein close-rtp.<!---->
For the transfer the CUCM first sets the call on hold and then requests a new media proposal from this call, which we cannot deliver. The request is just ignored, there should be an answer.<br/><!--<br/>h323ch.cpp<br/>--><!---->
''Status:''
ac_dsp3.cpp
      
      
=== IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax receive max packet size increased to 1024 ===
=== IP152: Call replacement (blind transfer) did not work ===


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For internet fax devices<!---->
Call replacement (blind transfer) ends up in no audio.<br/><br/><!--<br/>coldfire_afe.cpp<br/>--><!---->
''Status:''
ac_fax3.cpp
      
      
=== phone: ip222, ip232: USB controller sometimes hangs ===
=== Trap when disabling Gateway interface ===


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happened with certain headsets after fast connect/disconnect sequences<!---->
0:0204:814:3 - SIP_UDP.17 -> SIP_TRANSPORT.1 : SOCKET_RECVFROM_RESULT(87.173.157.2:5060,10.2.2.1:5060;95fd:75c0:9485:bc94:9485:bc30:9406:630c:5060,948a:2198:9476:5924:19:72:0:1:5060)<br/>      data(913),SIP_TRANSPORT<br/>      <br/>\tREGISTER sip:gwdl_ip800 SIP/2.0<br/>\tVia: SIP/2.0/UDP 87.173.157.2:5060;branch=z9hG4bKA072411261DD0135<br/>\tRoute: <sip:87.139.89.223;lr><br/>\tFrom: <sip:93@gwdl_ip800>;tag=4110839322<br/>\tTo: <sip:93@gwdl_ip800><br/>\tCall-ID: DECE171E57A1898F@192.168.178.1<br/>\tCSeq: 7998 REGISTER<br/>\tContact: <sip:93@87.173.157.2;uniq=D526C620EFAB7D1ACBBC469D33242><br/>\tAuthorization: Digest username="_!(schuetz)", realm="gwdl_ip800", nonce="7977d820e909d311", uri="sip:gwdl_ip800", response="6999f52d0e7d4ddbd18b23b59936aaa6", algorithm=MD5, cnonce="F56053DD72D3E1B3", qop=auth, nc=00000401<br/>\tExpires: 1800<br/>\tMax-Forwards: 70<br/>\tUser-Agent: AVM FRITZ!Box Fon WLAN 7112 (UI) 87.04.87 (Jun  7 2011)<br/>\tSupported: 100rel,replaces,timer<br/>\tAllow-Events: telephone-event,refer,reg<br/>\tAllow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH<br/>\tAccept: application/sdp, multipart/mixed<br/>\tAccept-Encoding: identity<br/>\tContent-Length: 0<br/><br/><br/>0:0204:815:5 - SIP-Client(SIP-CLIENT.0) <sip:93@gwdl_ip800;user=phone> ...<br/>0:0204:816:5 - SIP-Client(SIP-CLIENT.0) Idle->Registering<br/>0:0204:816:6 - SIP_RAS_APP.3 -> GK.0 : RAS_DISCOVERY, ip=87.173.157.2<br/>      data(6),GK<br/>      00 00 00 02  39 33                                  ....93          <br/>0:0204:816:7 - GK.0 -> SIP_RAS_APP.3 : RAS_DISCOVERY_CONFIRM<br/>0:0204:817:0 - SIP_RAS_APP.3 -> GK.0 : RAS_REGISTRATION(87.173.157.2:5060)<br/>      data(6),GK<br/>      00 00 00 02  39 33                                  ....93          <br/>0:0204:817:1 - GK.0 -> SIP_RAS_APP.3 : RAS_REGISTRATION_CONFIRM <br/>      data(32),SIP_RAS_APP<br/>      00 00 00 02  39 33 00 01  00 16 00 5f  00 21 00 28  ....93....._.!.(<br/>      00 73 00 63  00 68 00 75  00 65 00 74  00 7a 00 29  .s.c.h.u.e.t.z.)<br/>0:0204:817:1 - DEBUG this=948b325c event.sig=0<br/>0:0204:817:2 - DEBUG this->reg_reference=0 event.reference=948b3ebc<br/><!---->
''Status:''
checked in to 10.00, 9.00<br/>merged to 90600
      
      
=== SIP: SDP answer sometimes contains 2 media descriptions for audio ===
=== Presence note got lost ===


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|}
|}
... instead of one for audio and one for video.<br/><br/>Offer:<br/><br/>    v=0<br/>    o=- 1 2 IN IP4 130.30.1.111<br/>    s=CounterPath Bria<br/>    c=IN IP4 130.30.1.111<br/>    t=0 0<br/>    m=audio 1902 RTP/AVP 0 8 18<br/>    a=alt:1 1 : 1A98X9/s 31UwK5IA 130.30.1. 111 1902 <br/>    a=fmtp:18 annexb=yes<br/>    a=rtpmap:18 G729/8000<br/>    a=sendrecv<br/>    a=x-rtp-session-id:ACE43888C33A4AC8895C7F8F69380914<br/>    m=video 28212 RTP/AVP 115 34 123 124 <br/>    a=alt:1 1 : yjz/mbyQ aWGuD8xJ 130.30.1. 111 28212 <br/>   a=fmtp:115 QCIF=1;CIF=1;I=1;J=1;T=1<br/>   a=fmtp:34 QCIF=1;CIF=1<br/>   a=fmtp:123 profile-level-id=42801e; packetization-mode=0; max-mbps=48600<br/>    a=fmtp:124 profile-level-id=42801e; packetization-mode=1; max-mbps=48600<br/>    a=rtpmap:115 H263-1998/90000<br/>   a=rtpmap:34 H263/90000<br/>    a=rtpmap:123 H264/90000<br/>    a=rtpmap:124 H264/90000<br/>    a=sendrecv<br/>    a=x-rtp-session-id:694677A9F65D419FB7B9E451FFCA02CC<br/><br/><br/>Answer:<br/><br/>    v=0<br/>    o=- 3141 1 IN IP4 10.230.5.1<br/>   s=-<br/>    c=IN IP4 10.230.5.1<br/>    t=0 0<br/>    m=audio 16410 RTP/AVP 0 13<br/>    b=TIAS:64000<br/>    a=ptime:20<br/>    a=silenceSupp:off - - - -<br/>   a=sendrecv<br/>    m=audio 16410 RTP/AVP 0 13<br/>    b=TIAS:64000<br/>    a=ptime:20<br/>    a=silenceSupp:off - - - -<br/>    a=sendrecv<!---->
When configuring a presence on the phone (Main menu/User setup/Presence)<br/>the note is not save along with the selected activity<br/><br/><!--<br/>app_gui.cpp<br/>--><!---->
      
      
=== Conference: Permanent noise in conference channel ===
=== PBX: Potential trap on registration of a slave PBX, with Master GK-ID configuration changes ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=78838 78838]
|[http://mantis.innovaphone.com/view.php?id=85477 85477]
|}
|}
Permanent noise in conference channel.<br/>Increasing with every new participant.<!---->
The trap happened if on the Slave a Master GK-ID was configured, then the slave registered, and afterwards the Master GK-ID was removed again and the slave registered again and then was restarted once more.<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== PBX: When doing a show users, unnecessary data was sent to the browser ===
=== Presence: Note moves into 'tel' presence ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=78878 78878]
|[http://mantis.innovaphone.com/view.php?id=85482 85482]
|}
|}
The data could be reduced and thus speeding up the display of big users lists.<!---->
Presence note moves into 'tel' presence when changing activity by use of 'presence' fkey.<br/><br/><!--<br/>app_disp.cpp<br/>--><!---->
      
      
=== PBX Admin UI: Editing profiles was not possible if the user had too many contacts ===
=== IP232,IP222: Automatically enter input mode 'alpha' when entering directory search screen through fkey ===


{|
{|
Line 11,434: Line 11,251:
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=78883 78883]
|[http://mantis.innovaphone.com/view.php?id=85493 85493]
|}
|}
The form is now submitted using POST instead of GET.<!---->
Automatically enter input mode 'alpha' when entering directory search screen through fkey.<br/><br/><!--<br/>phone_dir_ui.cpp<br/>forms_color.cpp<br/>--><!---->
      
      
=== SIP: Wrong number of waiting messages (MWI) ===
=== SIP: Interop with Genband SBC ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=78890 78890]
|[http://mantis.innovaphone.com/view.php?id=85534 85534]
|}
|}
MWI: Number of voice messages not decoded from incoming NOTIFY(application/simple-message-summary).<br/>Was either 1 or 0.<!---->
Handling this kind of SDP offer:<br/><br/>\tv=0<br/>\to=IOTMSX1-0 17 2 IN IP4 206.165.51.38<br/>\ts=sip call<br/>\tc=IN IP4 0.0.0.0<br/>\tt=0 0<br/>\tm=audio 42076 RTP/AVP 0 8 4 18 101 13<br/>\ta=sendonly<br/>\ta=rtpmap:101 telephone-event/8000<br/>\ta=fmtp:18 annexa=yes<br/>\ta=fmtp:18 annexb=no<br/>\ta=fmtp:101 0-15<br/>\ta=ptime:20<br/>\ta=silenceSupp:off - - - -<br/><br/><!--<br/>sip.cpp<br/>--><br/><!---->
      
      
=== SIP: Send even anonymous Diversion header ===
=== PBX-Exec: Call at secretary for executive had wrong destination name/number ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=78954 78954]
|[http://mantis.innovaphone.com/view.php?id=85535 85535]
|}
|}
If diverting party's identity is unknown/hidden.<br/>Diversion: ;reason=user-busy<!---->
Call should be displayed as call to the scretary, diverted by the executive<br/><!--<br/>pbxb_exec.cpp<br/>--><!---->
      
      
=== SIP: Failed to decode presence XML from CUCM ===
=== Media: RTP did not work on switch from slowstart T.38 to EFC G.711 ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=78967 78967]
|[http://mantis.innovaphone.com/view.php?id=85548 85548]
|}
|}
Support for namespace "urn:ietf:params:xml:ns:pidf:status:rpid" added.<!---->
T.38 retransmission packets were sent, which caused problems with the RTP<br/><!--<br/>media.cpp<br/>--><!---->
      
      
=== IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax failure after transfer #2 ===
=== IP222 IP232 IP241: DSP Update to Version 680.05 ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=78969 78969]
|[http://mantis.innovaphone.com/view.php?id=85552 85552]
|}
|}
tone flag was not disabled<!---->
Sporadic problems with noise reduction fixed.<br/>Noise reduction can be enabled for Handset/Headset/Handsfree mode<br/><br/><!-- <br/>DSP code details at \\\\innovaphone.sifi\\dfs\\Dokumente\\Hardware\\G7xx\\Audiocodes\\firmware\\firmware_ac49x\\680.05<br/><br/>ac49x_680/* ac_phone3.cpp ac_phone3.h ac_494.cpp ac_494e.cpp ac491.cpp ac_firmware.h phone_orchid.mak phone_dsp.xsl --><!---->
''Status:''
ac_dsp3.cpp
      
      
=== H.323: Media problem with SIP call-completion on IP-DECT ===
=== SIP: Providing private RTP address although public RTP address is available ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=78970 78970]
|[http://mantis.innovaphone.com/view.php?id=85751 85751]
|}
|}
There was no media on the successful completion call (SDP recived in Progress was not forwarded)<!---->
During re-negotiation:<br/>Providing private RTP address to external endpoint although public RTP address is available.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== SIP: Send 'Connected Number' in P-Asserted-Identity header of 200/OK ===
=== SIP: Do not send "504 Server Time-out" after "200 OK" ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=78974 78974]
|[http://mantis.innovaphone.com/view.php?id=85788 85788]
|}
|}
Send 'Connected Number' in P-Asserted-Identity header of 200/OK<br/>if different from original called number.<!---->
Do not send "504 Server Time-out" after "200 OK".<br/><br/><!--<br/>siptrans.cpp<br/>--><!---->
      
      
=== SIP: Re-transmission of ACK with SDP answer ===
=== IP-DECT: Call transfer compatibility with SIP ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=78975 78975]
|[http://mantis.innovaphone.com/view.php?id=85795 85795]
|}
|}
Handling of re-transmissions of 200/OK with SDP offer.<br/>ACK must contain SDP answer.<!---->
For compatibility reasons with the SIP protocol no hold signal should be sent to the call transfer destination. This is fixed again.<br/><br/><!--<br/>Changed file:<br/>dectradio_call.cpp<br/>--><!---->
      
      
=== IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: DSP packet debug didnt show some packets, version endian was wrong ===
=== SIP: Handling of "sendonly" offer after "inactive" offer ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=78985 78985]
|[http://mantis.innovaphone.com/view.php?id=85857 85857]
|}
|}
cleanup<!---->
Switching from "inactive" into "recvonly".<br/>Giving REMOTE_RETRIEVE and HOLD_NOTIFY to app.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
''Status:''
ac_491.cpp<br/>ac_494.cpp
      
      
=== IP-DECT: Radio/Master calls switched to slowstart ===
=== SIP: Ignore From-URI in re-INVITE and UPDATE when "from-change" not supported ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=78987 78987]
|[http://mantis.innovaphone.com/view.php?id=85873 85873]
|}
|}
If first media answer was received with a PROGRESS message, the call leg from the radio to the master was switched to progress (no EFC features are defined for the Progress message).<br/><br/>This could cause media problems later in the call with hold/retrieve/transfer.<!---->
Ignore From-URI in re-INVITE and UPDATE when "from-change" not supported by remote side.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== SIP: REFER does not work as expected on Gateway interfaces without registration ===
=== Phones: Touching should start dialing of selected directory entry ===


{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79008 79008]
|[http://mantis.innovaphone.com/view.php?id=85959 85959]
|}
|}
Bad side effect of a previous fix in HF8:<br/>http://mantis.innovaphone.com/view.php?id=76309<br/><br/>Introducing new SIP interop tweak "Take Refer-To URI as Remote Target URI"<br/>to be found on the Gateway interface config (GWx).<br/>See http://mantis.innovaphone.com/file_download.php?file_id=66299&amp;type=bug<br/><br/><!--<br/>voip.h<br/>gk.cpp/h<br/>sip.cpp<br/>relay_edit_voip.xsl<br/>--><!---->
Touching should start dialing of selected directory entry<br/>instead of opening directory entry for editing.<br/>At least in Context of Wahlvorbereitung.<!---->
      
      
=== IP-DECT: Better handling when receiving a call, while a previous call is being released ===
=== SNMP-Traps: agent-addr wasn't reflecting altered IP adress ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79022 79022]
|[http://mantis.innovaphone.com/view.php?id=85970 85970]
|}
|}
Call to service number, which cause a callback can create problems.<br/><br/><!--<br/>Changed files:<br/>dectradio_call.h<br/>dectradio_call.cpp<br/>--><!---->
SNMP-Traps: The SNMP trap's agent-addr member carried an outdated ip adress if the DHCP leased ip address changed.<br/><!--<br/>snmp.cpp<br/>snmp.h<br/>--><!---->
      
      
=== Edss1 Interworking: Sending divertingLegInformation2 as PublicPartyNumber(was UnknownPartyNumber) ===
=== SIP: Optimization when validating local media address ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79025 79025]
|[http://mantis.innovaphone.com/view.php?id=86029 86029]
|}
|}
Such that the type of number resulting from an interface cgpn-map can also be  asn.1 encoded.<!---->
Optimization when validating local media address.<br/>Keep number of SOCKET_GET_LOCAL_ADDR low.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== Gateway: Mapping of Message Center Number of MWI ===
=== DHCP: client IP connectivity lost when a renew/rebind request for the currently assigned address was refused by the server ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79039 79039]
|[http://mantis.innovaphone.com/view.php?id=86030 86030]
|}
|}
Mapping of Message Center Number of MWI according to CGPN mapping.<!---->
When a client renew/rebind request is refused by the server providing the current address the client starts a new discovery. But in case of success the new address was not set and the client could not be reached anymore. <!---->
      
      
=== Qsig Interworking: Apply interface CGPN map to DGPN ===
=== ENUM: Port in SIP-URI was not honored ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79060 79060]
|[http://mantis.innovaphone.com/view.php?id=86037 86037]
|}
|}
Qsig Interworking: Apply interface CGPN map to DGPN<!---->
Port in SIP-URI was not honored in regex of DNS result.<br/><br/><!--<br/>enumlib.cpp/h<br/>relay.cpp<br/>sip.cpp<br/>--><!---->
      
      
=== PBX: A CFB at a PBX object was executed on call to busy user ===
=== IP-DECT: Transferred/rerouted call display ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79199 79199]
|[http://mantis.innovaphone.com/view.php?id=86041 86041]
|}
|}
It should only be executed, if the calls exceeded the Busy On number.<!---->
The remote party number of transferred and rerouted calls are not correctly shown in the handset's display. This fixes the display of CTI initiated calls.<br/><br/><!--<br/>Changed file:<br/>dectradio_call.cpp<br/>--><!---->
      
      
=== PBX: Mapping of Message Center Number of MWI ===
=== IP-DECT: MAC-alias change of OEM device ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79232 79232]
|[http://mantis.innovaphone.com/view.php?id=86047 86047]
|}
|}
Mapping of Message Center Number of MWI.<!---->
The MAC-alias of an OEM device was changed and this results in conflicts within several DECT modules. Different product short names of the same device are correctly accepted now.<br/><br/><!--<br/>Changed files:<br/>dectmaster.h<br/>dectmaster.cpp<br/>cryptomaster.cpp<br/>dectmobmaster.h<br/>dectmobmaster.cpp<br/>cryptomaster_registration.h<br/>registration_context.h<br/>registration_context.cpp<br/>--><br/><!---->
      
      
=== PBX-SOAP: Conference Id was missing in call-info for parked calls ===
=== ip1202: Send Inband DTMF did not work ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79243 79243]
|[http://mantis.innovaphone.com/view.php?id=86052 86052]
|}
|}
This caused problems with the innovaphone operator<!---->
the "DECT/Config/Master/Send Inband DTMF" checkmark had no effect. <!---->
      
      
=== PBX-SOAP: UserClear cause should not be used for remote party ===
=== SIP: Support for multiple audio media descriptions ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79249 79249]
|[http://mantis.innovaphone.com/view.php?id=86083 86083]
|}
|}
This can result in strange call clearing without tones<!---->
One without encryption and one with encryption:<br/><br/>v=0<br/>o=OpenStage-Line_0 968610650 446118927 IN IP4 10.30.1.144<br/>s=SIP Call<br/>c=IN IP4 10.30.1.144<br/>t=0 0<br/>m=audio 5012 RTP/AVP 9 8 0 18 101<br/>a=rtpmap:9 G722/8000<br/>a=rtpmap:8 PCMA/8000<br/>a=rtpmap:0 PCMU/8000<br/>a=rtpmap:18 G729/8000<br/>a=rtpmap:101 telephone-event/8000<br/>a=silenceSupp:off - - - -<br/>a=fmtp:18 annexb=no<br/>a=fmtp:101 0-15<br/>a=sendrecv<br/>m=audio 5010 RTP/SAVP 9 8 0 18 101<br/>a=rtpmap:9 G722/8000<br/>a=rtpmap:8 PCMA/8000<br/>a=rtpmap:0 PCMU/8000<br/>a=rtpmap:18 G729/8000<br/>a=rtpmap:101 telephone-event/8000<br/>a=silenceSupp:off - - - -<br/>a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ujVU8G6kgknZnPflRwx8tadNskkp9glas/DFCbC3<br/>a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:+uoFBNVVhDY5OOGvMOdAlvpxvc98hX/VeProhlwH<br/>a=fmtp:18 annexb=no<br/>a=fmtp:101 0-15<br/>a=sendrecv<br/><br/><br/><!--<br/>sip.cpp/h<br/>sdpmsg.cpp/h<br/>--><!---->
      
      
=== PBX: Include error in resulting XML if submit-object with insufficient rights was attempted ===
=== Change order of links on debug page ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79289 79289]
|[http://mantis.innovaphone.com/view.php?id=86121 86121]
|}
|}
Problem with innovaphone operator.<!---->
Make "Tracing" the first item so that no leak check is triggered when opening the page.<br/><!--<br/>debug.xml<br/>maintenance.xml<br/>--><!---->
      
      
=== phone: ip222,ip232,ip241: wrong AM/PM time display on status line ===
=== Phones: Using config option "CGPN" as CGPN on outbound SIP calls ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79302 79302]
|[http://mantis.innovaphone.com/view.php?id=86122 86122]
|}
|}
15:19 PM was displayed instead of 03:19 PM<!---->
Using config option "CGPN" as CGPN on outbound SIP calls.<br/><br/><!--<br/>phonesig.cpp<br/>--><!---->
''Status:''
checked in to 10.00, 9.00, 90600
      
      
=== Voicemail: Default for mwi.basicService now Speech(1) (was allServices(0)) ===
=== phone: ip222, ip232: recover from USB port failure probably caused by electric sparks ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79303 79303]
|[http://mantis.innovaphone.com/view.php?id=86125 86125]
|}
|}
Voicemail: Default for mwi.basicService now Speech(1) (was allServices(0))<br/><br/>May from now on be overriden by variable "$_pbxmwiservice".<br/><!---->
An USB port failure is indicated when a Plantronics DA45 headset adapter is plugged and a certain kind of table lamp (halogen) is switched on or off.<br/>It happens independent of current state of the headset (idle or in call)<br/>but only with the abovementioned adapter.<br/>The exact reason is not known yet, may be it's an electric spark from the switch of the lamp or some pulse.<br/>The fix is to reset the port and to restart the plugin process, a possibly active call is terminated.<br/><br/><!-- keys_ip222.cpp ohci.cpp ohci.h ohci_defs.h --> <br/><!---->
      
      
=== PBX: Local objects could not be called from Nodes with escapes as expected ===
=== phone: coder settings of a "Create Registration"  function key were not applied to the created registration ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79317 79317]
|[http://mantis.innovaphone.com/view.php?id=86164 86164]
|}
|}
Was not possible to call at all, or the number was wrong<!---->
<!-- app_fkey.cpp --><!---->
      
      
=== PBX-Trunk: Handling for Incomplete, Invalid, Busy, No Anser destinations for Media calls only ===
=== Disable leak check if debug flag is not set ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79319 79319]
|[http://mantis.innovaphone.com/view.php?id=86165 86165]
|}
|}
For example this should not be done for presence/dialog subscriptions<!---->
The leak check is only allowed if the config flag CPU /debug is set.<br/><!--<br/>command.cpp<br/>leaks.xml<br/>--><!---->
      
      
=== Refactored some ASN.1 BER Handling ===
=== AD Replication: Configuration Buffer Increased ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79357 79357]
|[http://mantis.innovaphone.com/view.php?id=86211 86211]
|}
|}
Became necessary in the run of: #79260: ASN.1 BER: Support for indefinite-length encoding<!---->
Was too small for many maps<br/><!--<br/>ldaprep.cpp/.h<br/>--><!---->
      
      
=== PBX: Conference &amp;amp; VM ===
=== whistling tone in all ip28 a/b ports on incoming call ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79359 79359]
|[http://mantis.innovaphone.com/view.php?id=86212 86212]
|}
|}
The last connected user is not disconnected although it is configured. It occurs if a VM PBX object forwards the call to the BC Conference object (like the innovaphone conferencing script). It is fixed now.<!---->
This phenomena occured after few day uptime after sending CLIP<!---->
      
      
=== IP-DECT: Subscription could get lost randomly with logout/login cycle ===
=== ip72 firmware did not boot anymore since V9hotfix11 ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79397 79397]
|[http://mantis.innovaphone.com/view.php?id=86246 86246]
|}
|}
Problem with user database handling<br/><!--<br/>dectusers.cpp<br/>dectusers.h<br/>--><!---->
<!-- ip220.cpp --><!---->
      
      
=== H.323: Media problem if PBX rtp-proxy is activated for CFNR to external destination ===
=== SIP: Avoid re-configuration of DSP channel when processing re-INVITE ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79430 79430]
|[http://mantis.innovaphone.com/view.php?id=86316 86316]
|}
|}
no voice<br/><!--<br/>h323ch.cpp<br/>h323sig.cpp<br/>--><!---->
Fix is required for interop with SIP devices sending re-INVITE for session-refresh,<br/>but incrementing version field in SDP body, altough there is no change in SDP.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== HTTP: Could not have sockets with same ports but different remote addresses ===
=== phone: headset function key mode 'control' could be configured via WEB interface only ===


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|}
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This was a general problem with the new TCP stack, used together with IP6. Problem only happened for HTTP because only for HTTP this new stack is currently used.<br/><br/><!--<br/>ip_user.cpp<br/>tcp.cpp<br/>--><!---->
must be possible locally at the phone too<br/><br/><!-- phone_user_func.cpp phonetxt.h phonetxt-utf8.cpp phonetxt-latin1.cpp phonetxt-utf8.base --><!---->
      
      
=== ISDN: Fix for call completion interworking ===
=== phone: Partner state provided via dialog info was not reset when the subscription call was released  because of network errors ===


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|[http://mantis.innovaphone.com/view.php?id=86330 86330]
|}
|}
CCBS/CCNR does not work in some cases<br/><br/><!--<br/>q950.cpp<br/>--><br/><br/><!---->
<!-- phonesig.cpp app_reg.cpp --><!---->
      
      
=== SIP: From-URI may got lost after call transfer ===
=== Gateway: Routing on blind transfer call starts from wrong IF ===


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UPDATE may be send out missing From-URI.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
Transfered endpoint was used as source interface on routing.<br/>Better use transfering endpoint as source on routing of (blind) transfer call.<br/>Also transfer-to endpoint missed ctSetup.<br/>Also transfered endpoint missed ctComplete.<br/><br/><!--<br/>relay.cpp<br/>sip.cpp<br/>--><!---->
      
      
=== AD Replication stuck after connection loss ===
=== Linux: Start-up failures ===


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Didn't reconnect. Side effect of after-hf10 fix<br/><!--<br/>adrep.cpp<br/>--><!---->
Linux start-up is improved. This fixes hanging Linux start-ups caused by compact-flash failures.<br/><br/><!--<br/>Changed files:<br/>ip6010.h<br/>ip6010.cpp<br/>--><!---->
      
      
=== PBX Waiting/Broadcast: Incomplete CDRs if CFNR configured on object ===
=== IP-DECT: GUI Master Configuration ===


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If the CFNR was executed, the CDR for the call to the Waiting Queue/Broadcast object ended without release.<br/><!--<br/>pbx.cpp<br/>pbx_wait.cpp<br/>pbx_bc.cpp<br/>--><!---->
Disabling LDAP Directory Search fixed for the IP1202.<br/><br/><!--<br/>Changed file:<br/>_inn_asc_dectmaster.xsl<br/>--><!---->
      
      
=== PBX: Web User interface problem with IE and non-ascii PBX names ===
=== Linux: GUI in viewer mode ===


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For example groups at a user assigned to a PBX with non-ascii characters could not be edited.<br/>The problem is a bug in IE XSL translation which does special handling of href attributes. Same thing with onclick attribute works.<br/><!--<br/>pbx_objs_left.xsl<br/>pbx_objs_right.xsl<br/>--><!---->
The Linux General page is fixed for the viewer only mode.<br/><br/><!--<br/>Changed files:<br/>config_options.cpp<br/>linux.xsl<br/>--><!---->
      
      
=== ip3010 TEL port not working in NT mode ===
=== SIP: PRACK after CANCEL contains bad RAck header value ===


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TEL port of ip3010 gateways configured in NT mode do not get Physical Link up indication. This problem applies to V9hotfix7 up to V9hotfix10.<!---->
CSeq or original INVITE transaction was damaged.<br/>But only if CANCEL has been sent right before PRACK.<br/>CANCEL is sent before PRACK only if SDP answer of provisional response is invalid.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
''Status:''
ip6010.cpp
      
      
=== SIP: Max forward value of 32 could be too small for some provider ===
=== PBX: Conference trap ===


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For a starting value of max-forwards a value of 32 was used, because this is the maximum value in H.323. This was too small for some sip providers. Starting value now increased to 64 and on H.323 the half value is transmitted.<br/><!--<br/>h323sig.cpp<br/>signal.cpp<br/>signal.h<br/>--><!---->
Potential trap in the PBX BC conference object is fixed.<br/><br/><!--<br/>Changed files:<br/>pbx_bc_conf.h<br/>pbx_bc_conf.cpp<br/>--><!---->
      
      
=== SIP: Re-negotiation for T38 did not work in media-relay scenarios ===
=== SNMP: Obsoleted Enterprise-Specific Trap "innoIsdnFailure" ===


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Bad SDP answer was generated.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
This SNMP trap is no longer necessary. Meanwhile it is covered more consistently by "innoDiagAlarm" and "innoDiagAlarmClear".<br/><!--<br/>gk.cpp/.h<br/>--><!---->
''Status:''
Introduced by<br/> #77277: SIP: SDP answer must have the same number of media descriptions as received offer
      
      
= V9 Hotfix 12 (9061009) =
=== Linux: GUI removed from IP1060 ===
Changes included in Version 9 hotfix12
[http://mantis.innovaphone.com/view.php?id=79736 Definition]
 
== New Features ==
 
 
   
=== myPBX: Syslog for sessions and application sharing ===


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Syslog entries should be useful for debugging session timeouts and problems with the external application sharing solution.<br/><!--<br/>pbx_client.cpp<br/>pbx_client.h<br/>log.xsl<br/>--><!---->
The IP1060 does not support the Linux Application Platform. The GUI is removed now. Also a trap is fixed, if somebody tries to enable Linux.<br/><br/><!--<br/>Changed files:<br/>linux.xml<br/>ip6010/platform/config.h<br/>ip6010/platform/ip6010.cpp<br/>ip6010/platform/tabs.xml<br/>ip6000/platform/ip6000.cpp<br/>--><!---->
      
      
=== SIP: Pass display names of &lt;dialog-info&gt; to phoneapp ===
=== Voicemail: &lt;pbx-disc&gt; failed sometimes ===


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Pass display names of <dialog-info> to phoneapp.<br/><br/><!--<br/>fty.cpp/h<br/>sip.cpp<br/>phonesig.cpp<br/>--><!---->
In case of unconnected calls that were subject of a prior <pbx-fwd><br/><!--<br/>pbx_vm.cpp<br/>--><!---->
      
      
=== Gateway: Overlap dialing timeout configurable ===
=== phone: ip222, ip232:  handset/handsfree speaker was not switched off when changing over to headset mode ===


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The default of 4s is not good for all applications. Now configurable between 500ms and 6000ms.<br/><!--<br/>relay_edit_route.xsl<br/>gk.cpp<br/>gk.h<br/>gk_if.h<br/>relay.cpp<br/>relay.h<br/>--><!---->
When the headset talk-key or the phone headset-control-key is pressed while the phone is in handset or handsfree mode the phone changes to headset mode, i.e. headset micro and speaker are activated. The handset or handsfree speaker should be switched off then.<br/><br/><!-- ac_codec3.cpp --><!---->
      
      
=== IP-DECT: OEM Configuration of Cisco Features changed ===
=== phone_orchid: pressing the speaker key while in headset mode did not switch over to handsfree mode ===


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should depend on installed license<br/><!--<br/>dectmaster.h<br/>dectmaster_if.h<br/>dectfty.cpp<br/>dectfty.h<br/>dtmffty.cpp<br/>--><!---->
When the speaker key is pressed while the phone is in headset mode the headset should be deactivated and the phone should enter handsfree mode.<br/><br/><!-- app_ctl.cpp --><!---->
   
= V9 Hotfix 17 (9061152) =
Changes included in Version 9 hotfix17
[http://mantis.innovaphone.com/view.php?id=86708 Definition]
 
== New Features ==
 
 
      
      
=== myPBX launcher: Automatically move main window into visible area of screen ===
=== Voicemail: &lt;pbx-query-obj type="filter"/&gt; couldn't read template distributed filter ===


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The window might be outside the visible area if the user changed the screen resolution or disconnectes one of the screens.<br/><br/>If the main window is not visible on any screen, it is now moved to the center of the primary screen, when the user clicks the tray icon.<br/><!---->
Switched to a different internal api method<br/><!--<br/>shelved:<br/>pbx_vm.cpp<br/>--><!---->
      
      
=== PBX-SOAP: FindUser optional argument 'nohide' ===
=== AD Replication: Refactoring Poll-Timer-based Replication ===


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Needed if the FindUser is used to look for users which have the 'Hide from LDAP property' set.<!---->
-Ridded usnChanged filter term<br/>-Added two status messages<br/><!--<br/>adrep.cpp<br/>--><!---->
''Status:''
<!--<br/>pbx_xml.cpp<br/>pbx900.wsdl<br/>pbx10_00.wsdl<br/>-->
      
      
=== PBX Mobility: Pick mobile call on fixed phone ===
=== PBX Mobility: Take precaution against hanging calls because of misbehaving mobile phone or client ===


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by dialing number of Mobility object on fixed phone<br/><!--<br/>pbx_mobility.cpp<br/>pbx_mobility.h<br/>pbx.cpp<br/>--><!---->
Use a 20s timer to terminate any call to the mobility object if no internal call is initiated<br/><!--<br/>pbx_mobility.cpp<br/>pbx_mobility.h<br/>--><!---->
      
      
=== Gateway: Allow setting of system time from ISDN time ===
=== IPVA: Query Physical Link Status. ===


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In ISDN connect messages a public network is sending a local date and time. By a configuration option this can now be used to update the system time<br/><!--<br/>relay_edit_phys,xsl<br/>gk.cpp<br/>gk.h<br/>gk_if.h<br/>relay.cpp<br/>relay_api.h<br/>--><!---->
-The physical link status wasn't evaluated so far. Now it is.<br/>-Added statistics page 'Interfaces/EthX/Statistics'<br/><!--<br/>vlance/vlance_drv.cpp<br/>vlance/vlance_drv_s.xls<br/>platform/interfaces.xml<br/>--><br/><!---->
      
      
=== IP-DECT: New Master/Radio behavior for license incompatibility ===
=== IP TOS value is now traced with Wireshark ===


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The behavior of the DECT Master is changed, if an unlicensed Radio tries to register in. This is only used for an OEM license model.<br/><br/><!--<br/>Changed files:<br/>dectmaster.h<br/>dectmaster.cpp<br/>dectradio.h<br/>dectradio.cpp<br/>asc_dectradio.xsl<br/>asc_dectmaster_radios.xsl<br/>--><!---->
This value is now correctly traced within IP4 UDP/TCP traces.<br/><br/><!--<br/>debug.cpp/h<br/>ip_user.cpp/h<br/>tcp.cpp/h<br/>udp.cpp/h<br/>ip4_user.h<br/>ipproc.cpp/h<br/>tcp4.cpp/h<br/>udp4.cpp/h<br/>ip6_icmpsock.cpp/h<br/>test_debug.cpp/h<br/>test_tcp.cpp/h<br/>test_udp.cpp/h<br/>--><!---->
      
      
=== PBX: Better handling of presence information without activity ===
=== PBX Trunk: New destination for rejected calls ===


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This type of presence information could contain a note which refers to the future<br/><!--<br/>pbx.cpp<br/>--><!---->
Calls which are rejected (busy after alert), can be redirected to a configurable destination.<br/><!--<br/>pbx_trunk.cpp<br/>pbx_trunk.h<br/>pbx_edit_trunk.xsl<br/>--><!---->
      
      
=== RPCAP trace: Indicate transmit/received packets by setting the remote mac address to 00-90-33-00-00-00 ===
=== SIP: Offer media encryption as separate media description ===


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When reading a trace it is currently not obvious if a packet is sent or received, we need to find out the devices IP address, e.g. by reading the config. If the devices on MAC adress is used a source only if a packet is sent and as destination only if a packet is received this process is simplified.<br/><br/>This is now done for UDP/TCP traces!<!---->
Interop issue.<br/>New config file option "config change SIP /separate-encryption".<br/><br/>Offer:<br/>\tv=0<br/>\to=- 10 1 IN IP4 172.16.16.156<br/>\ts=-<br/>\tc=IN IP4 172.16.16.156<br/>\tt=0 0<br/>\tm=audio 16390 RTP/AVP 9 8 0 18 101 13<br/>\tc=IN IP4 172.16.16.156<br/>\ta=rtpmap:101 telephone-event/8000<br/>\ta=fmtp:18 annexa=yes<br/>\ta=fmtp:18 annexb=no<br/>\ta=fmtp:101 0-15<br/>\ta=ptime:30<br/>\ta=silenceSupp:off - - - -<br/>\ta=sendrecv<br/>\tm=audio 16390 RTP/SAVP 9 8 0 18 101 13<br/>\tc=IN IP4 172.16.16.156<br/>\ta=rtpmap:101 telephone-event/8000<br/>\ta=fmtp:18 annexa=yes<br/>\ta=fmtp:18 annexb=no<br/>\ta=fmtp:101 0-15<br/>\ta=ptime:30<br/>\ta=silenceSupp:off - - - -<br/>\ta=sendrecv<br/>\ta=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:VoGZgKwImVwTkJg4jBeYFBafl/CyJpfMX66WqDMZ<br/><br/>Instead of:<br/>\tv=0<br/>\to=- 9 1 IN IP4 172.16.16.156<br/>\ts=-<br/>\tc=IN IP4 172.16.16.156<br/>\tt=0 0<br/>\tm=audio 16386 RTP/SAVP 9 8 0 18 101 13<br/>\tc=IN IP4 172.16.16.156<br/>\ta=rtpmap:101 telephone-event/8000<br/>\ta=fmtp:18 annexa=yes<br/>\ta=fmtp:18 annexb=no<br/>\ta=fmtp:101 0-15<br/>\ta=ptime:30<br/>\ta=silenceSupp:off - - - -<br/>\ta=sendrecv<br/>\ta=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:mDPFKhNdhm6fhkeyPbAI9uzZyTmtA4t9fy1MwXiD<br/><br/><br/><!--<br/>sip.ccp/h<br/>--><!---->
      
      
=== SIP: Support for Alert-info:&lt;Bellcore-dr1&gt; ===
=== IP-DECT: DECT security (4) ===


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Proprietary tagging of internal calls<br/><br/><!--<br/>sip.cpp<br/>--><!---->
Some changes in preparation for the new feature "DECT security". This feature will be available later (IP1202).<br/><br/><!--<br/>Changed file:<br/>dectusers.cpp<br/>--><!---->
      
      
=== phone: ip222,ip232: more USB headsets supported ===
=== SIP: Do registration refresh more ahead of expiration ===


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for a complete list see http://wiki.innovaphone.com/index.php?title=Reference9:Concept_USB_Headset<br/><br/><!--<br/>keys_ip222.cpp<br/>--><!---->
Instead of refreshing 2 secs before registration expiration,<br/>do the refresh after 98,33% of the registration ttl.<br/><br/>TTL=120secs -> Refresh 2secs before expiration<br/>TTL=3600secs -> Refresh 60secs before expiration<br/><br/><!--<br/>sip.cpp<br/>--><!---->
''Status:''
checked in to 10.00, 9.00
      
      
=== Phones: Switch for phoneapp to disable auto-answer ===
=== phone: Modification of function key on phone can be disabled for any single key ===


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Disable/enable auto-answer support on phoneapp level.<br/><br/><!--<br/>phonesig_if.h<br/>phonesig.cpp<br/>--><!---->
A "Disable Modification on Phone" checkmark will be provided in the edit menu for  each key. If checked the key cannot be edited on the phone anymore.<br/>This mechanism works in addition to the phone local key type mask set via<br/>   "Phone/Protect/Function keys not modifiable on the phone"<br/>A key of a type NOT marked as ``not editable´´ in this mask can be made ``not editable´´ by setting the above mentioned checkmark<br/>A key of a type marked as ``not editable´´ remmains not editable, independent of the checkmark setting.<br/><br/><!-- phone_config.cpp phone_config.h phone_edit.cpp phone_user_func.cpp fkey_edit_*.xsl --><br/><br/><!---->
      
      
=== PBX: DECT attributes for DECT security ===
=== phone: "Spare" function key to reserve key positions for administrative purposes ===


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In preparation for the new DECT feature DECT security there will be new attributes for the endpoint data which must be taken over. With this fix the innovaphone PBX supports the new attributes if the user is edited.<br/><br/><!--<br/>Changed files:<br/>pbx_admin.cpp<br/>pbx_edit_object.xsl<br/>--><br/><!---->
This new key permits to reserve key positions for later definition by administrator.<br/><br/><!-- phone.mak phone_config.h phone_config.cpp phone_edit.cpp fkey_edit_spare.xsl app_disp.cpp app_fkey.cpp phone_user_func.cpp phone_strings_utf8.h phone_strings_latin1.h phone_text_tables.h --><!---->
      
      
=== phone: ip222, ip232: Jabra  USB Headset feature "Reject incoming call" supported now ===
=== IP-DECT: Remote hold in conference mode ===


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A double tap on talk button rejects a ringing call with 'busy'<br/><br/><!--<br/>keys_ip222.cpp<br/>--><!---->
If a hold notify message is received from a remote party and the conference mode is active, now the message is forwarded to the conference unit. This prevents the music on hold in conference calls. The state is also shown in the radio call list.<br/><br/><!--<br/>Changed files:<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp<br/>dectradio_calls.xsl<br/>--><br/><!---->
      
      
=== IP-DECT: Signal waiting calls to handset ===
=== Webdav: Write information into trace if DELETE fails because file is in open state ===


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New event to the DECT system to add a waiting call to the call list.<br/>This is step two of the implementation and related to the case #79349.<br/><br/><!--<br/>Changed files:<br/>fty.h<br/>fty.cpp<br/>dectmaster.cpp<br/>dectradio_call.cpp<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp<br/>h450.h<br/>h450.cpp<br/>h450asn1.h<br/>h450asn1.cpp<br/>--><br/><!---->
Write information about Webdav session holding open file handle.<br/><br/><!--<br/>http.h<br/>servlet.h<br/>servlet_webdav.cpp/h<br/>--><!---->
      
      
=== IP-DECT: Anonymous endpoint information ===
=== H.323: Better error handling in case of incompatible SRTP parameters ===


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|}
Information about anonymous endpoint is added to the event to the DECT system. In preparation for the new feature "easy subscription".<br/><br/><!--<br/>Changed files:<br/>signal.h<br/>signal.cpp<br/>dectradio.cpp<br/>--><!---->
Was needed for DECT OEM.<br/><!--<br/>h323ch.cpp<br/>--><!---->
      
      
=== IP-DECT: DECT security ===
=== PBX: No CDR was generated for enblock call, which was rejected because of filter config ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=80424 80424]
|[http://mantis.innovaphone.com/view.php?id=87869 87869]
|}
|}
Some changes in preparation for the new feature "DECT security". This feature will be available later.<br/><br/><!--<br/>Changed files:<br/>signal.h<br/>signal.cpp<br/>dummysig.cpp<br/>dectusers_if.h<br/>dectusers.h<br/>dectusers.cpp<br/>dectmaster.h<br/>dectmaster.cpp<br/>dectradio.h<br/>dectradio.cpp<br/>dectlocalusers.h<br/>dectlocalusers.cpp<br/>dectmobmaster.h<br/>dectmobmaster.cpp<br/>asc_dectusers.xsl<br/>asc_dectradio.xsl<br/>asc_dect_edit_user.xsl<br/>--><!---->
A CDR with cause 52 (outgoing call barred) is generated<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_api.h<br/>--><!---->
      
      
=== Phone: Accept MWI from Exchange Server ===
=== Voicemail: Add &lt;pbx-getcallinfo out-leg2-orig="..."&gt; ===


{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=80446 80446]
|[http://mantis.innovaphone.com/view.php?id=87880 87880]
|}
|}
MS Exchange Server sends unsolicited NOTIFY(message-summary) to served user<br/>with served user's number as destination and origin.<br/>But phones expect to receive MWI message center number as origin.<br/>MWI fkey would not light up.<br/><br/><!--<br/>app_reg.cpp<br/>--> <!---->
Allows to query the divertingLegInformation2.orignalCalled number<!---->
      
      
=== SIP: New config option /take-zero-addr-for-hold ===
=== IP-DECT: Fault logging for Master module ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=80516 80516]
|[http://mantis.innovaphone.com/view.php?id=88005 88005]
|}
|}
From now on "c=IN IP4 0.0.0.0" is no longer accepted as hold signaling.<br/>Config option /take-zero-addr-for-hold is introduced to get back old handling.<br/><br/><!--<br/>sip.cpp<br/>sip.h<br/>--><!---->
Call and channel fault logging for the IP-DECT Master module are added.<br/><br/><!--<br/>Changed files:<br/>signal.h<br/>signal.cpp<br/>log_if.h<br/>log_if.cpp<br/>dectmaster.h<br/>dectmaster.cpp<br/>fault_dect.xsl<br/>--><!---->
      
      
=== IP-DECT: User log in, endpoint data added ===
=== Voicemail: &lt;pbx-prepcallinfo leg2=".." leg2-name=".."/&gt; ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=80620 80620]
|[http://mantis.innovaphone.com/view.php?id=88351 88351]
|}
|}
During the user log in with a change of the IP-DECT Master the endpoint data (product id/software version) is sent to this new one now.<br/><br/><!--<br/>Changed files:<br/>dectlocalusers.cpp<br/>dectradio.cpp<br/>--><!---->
New attributes allow to prepare <pbx-fwd> in such, that the supplementary service divertingLegInformation2 is going to be sent.<!---->
      
      
=== SIP: Pass display information to application ===
=== PBX: Allow configuration of default presence/dialog-info visibility for group members ===


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|[http://mantis.innovaphone.com/view.php?id=80632 80632]
|[http://mantis.innovaphone.com/view.php?id=88352 88352]
|}
|}
Pass display information received in Call-Info header in 200/OK for BYE to app.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
Active group members got full presence/dialog-info because this matched the visibility be group-indications. However this is not desired always, so it can now be configured to restrict this.<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_general.xsl<br/>config_options.cpp<br/>config_options.h<br/>--><!---->
      
      
=== IP-DECT: Idle display update ===
=== IP-DECT: Unused OEM modules removed ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=80654 80654]
|[http://mantis.innovaphone.com/view.php?id=88413 88413]
|}
|}
Handling of idle display update message in the call release message is added. Used with an OEM PBX.<br/><br/><!--<br/>Changed file:<br/>dtmffty.cpp<br/>--><!---->
Unused OEM modules has been removed now.<br/><br/><!--<br/>Changed files:<br/>config.h<br/>dect.cpp<br/>--><!---->
      
      
=== myPBX launcher: Configurable hotkey action ===
=== Gateway: Make interop flag "Ack incoming call" configurable on UI ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=80684 80684]
|[http://mantis.innovaphone.com/view.php?id=88499 88499]
|}
|}
Options are:<br/>- Copy selected phone number to myPBX<br/>- Show myPBX<br/><!--<br/>Workaround for #76582<br/>ctiwin.cs<br/>Config.resx<br/>Config.Designer.cs<br/>Config.cs<br/>--><br/><!---->
This was a hidden interop flag, but was now needed multiple times, so it is more efficient to have it configurable on the UI<br/><!--<br/>gk.cpp<br/>gk.h<br/>relay_edit_phys.xsl<br/>--><!---->
      
      
=== myPBX launcher: Autocomplete configured URLs ===
=== IP-DECT: Logging for handover calls ===


{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=80689 80689]
|[http://mantis.innovaphone.com/view.php?id=88536 88536]
|}
|}
If the user configures just an IP address, it shall be replaced by the full myPBX URL.<br/><br/>For example 192.168.0.10 will be replaced by http://192.168.0.10/PBX0/MY/client.htm.<br/><!--<br/>Config.cs<br/>--><!---->
Logging events for IP-DECT handover calls added.<br/><br/><!--<br/>Changed files:<br/>dectradio_call.h<br/>dectradio_call.cpp<br/>dectradio_mastercall.h<br/>dectradio_dectcall.h<br/>dectradio_dectcall.cpp<br/>dectradio_handover_out.h<br/>dectradio_handover_out.cpp<br/>dectradio_handover_in.cpp<br/>--><!---->
      
      
== Bug Fixes ==
== Bug Fixes ==
Line 12,146: Line 11,949:


      
      
=== PBX: CFNR at Gateway object with incomplete destination ===
=== make update script parsing more tolerant to suspicious line ends ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79605 79605]
|[http://mantis.innovaphone.com/view.php?id=84349 84349]
|}
|}
A CFNR at a gateway object is executed if there is no registration. Any additional digits dialed should be added to the CFNR destination. This did not work if the original CFNR destination was incomplete and only completed with additional digits dialed.<br/><!--<br/>pbx.cpp<br/>--><!---->
Some WEBDAV tools garble line end when a text file is stored after editing. Last seen \\r\\r<br/> instead of \\r<br/>. Any sequence consisting only of \\r <br/> chars should be read as one line end because empty lines have no meaning in an update script.<br/><br/><!-- update.cpp update.h --> <!---->
      
      
=== PBX Waiting: Filter did not work anymore for CFNR ===
=== SIP: Do not interwork holdNotific and retrieveNotific while on hold ===


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|[http://mantis.innovaphone.com/view.php?id=79610 79610]
|[http://mantis.innovaphone.com/view.php?id=86736 86736]
|}
|}
Collateral damage from<br/><br/>fix: #75465: PBX: CFNR Loop check detected loops that weren't<br/><!--<br/>pbx.cpp<br/>--><!---->
Interworking problem with VOXTRON/XCAPI<br/>Do not interwork holdNotific and retrieveNotific after remoteHold.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== LDAP Server caused crash on port-scanner attack ===
=== Disable LDAP Server When Erasing Flash Directory ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79634 79634]
|[http://mantis.innovaphone.com/view.php?id=86763 86763]
|}
|}
LDAP Server caused crash on port-scanner attack<br/><!--<br/>ldapsrv.cpp<br/>ldapsrv.h<br/>--><!---->
During an upload of a complete configuration the command "mod cmd FLASHDIR0 erase-all" will erase all flash directory content. Replication clients are going to receive nil-responses making them assume a certain entry does no longer exist.<br/><br/>Replication clients are now barred from accessing the LDAP server as long as the box didn't process the post-upload reset.<br/><!--<br/>interface/ldapapi.h<br/>interface/ldapapi.cpp<br/>service/ldap/ldapsrv.h<br/>service/ldap/ldapsrv.cpp<br/>service/ldap/flashdir.h<br/>service/ldap/flashdir.cpp<br/>--><br/><!---->
      
      
=== myPBX: Quotes in connected names caused java script errors ===
=== IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: fax bad signal quality events are sent on good fax connections ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=79636 79636]
|[http://mantis.innovaphone.com/view.php?id=86788 86788]
|}
|}
When the connected name of a call contained the ' character there was a script error.<br/><!--<br/>client.js<br/>client_dtmf.js<br/>client_util.js<br/>--><!---->
Sometimes during fax transfer a bad signal quality (e.g.50) is reported, even if the connection is good. This happens during the TCF phase, in the image phase the signal quality is fine ( e.g. 3) <br/><br/><!-- ac_fax3.cpp --><br/><!---->
      
      
=== PBX: Status displayed on boolean function key could be wrong ===
=== Mis-typing on Maintenance/Diagnostics/Counters ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79643 79643]
|[http://mantis.innovaphone.com/view.php?id=86812 86812]
|}
|}
In case of severe network problems, it could happen that the status displayed on a Boolean function key was wrong and was only corrected when the boolean status changed.<br/><!--<br/>pbx_bool.cpp<br/>pbx_gi.cpp<br/>--><!---->
Show "kbit/s" instead of "kBbit/s".<br/><br/><!--<br/>log_counter.xsl<br/>--><!---->
      
      
=== DHCP Server Identifier was cleared after editing the DHCP-Server page ===
=== Gateway: #11 could not be dialed on analog interfaces with feature codes enabled ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79651 79651]
|[http://mantis.innovaphone.com/view.php?id=86819 86819]
|}
|}
The value of "IP4/ETHx/DHCP/Server Identifier" was cleared when the OK or Renew button was pressed on the "IP4/ETHx/DHCP-Server" page. This bug was introduced with V9hotfix5.<br/><br/><!-- <br/>dhcp_server.xsl<br/>--><br/><br/><!---->
This is a featiure code used on DECT systems and it was not disabled on analog interfaces<br/><!--<br/>relayfty.cpp<br/>dtmffty.cpp<br/>--><!---->
      
      
=== SIP: Wrong Contact-URI in outgoing NOTIFY ===
=== Gateway: Potential trap when recording is configured and a transfer happens ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79700 79700]
|[http://mantis.innovaphone.com/view.php?id=86837 86837]
|}
|}
Contact-URI should match the Request-URI of the SUBSCRIBE.<br/>Also the Message-Account URI in "simple-message-summary" was wrong as result of the wrong Contact-URI.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
The problem is with a transfer, which is executed in the Gateway, not the PBX<br/><!--<br/>relay.cpp<br/>--><!---->
      
      
=== Config: Could not dynamically set or reset /trace on the LICENSE module ===
=== PBX: XML Export/Import did not work, if DTMF Feature objects present ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79709 79709]
|[http://mantis.innovaphone.com/view.php?id=86870 86870]
|}
|}
Missing handling of MODULE_UPDATE.<br/><br/><!--<br/>license.cpp<br/>--><!---->
The automatically generated user objects caused a problem. This could result in a config that caused the PBX to restart in a loop. The export/import was fixed and the PBX does not restart because of the corrupt config any more.<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_api.h<br/>pbx_admin.cpp<br/>--><!---->
      
      
=== PBX: Allow call completion only if recall is not prohibited by CFU ===
=== PBX: Trap if a Hold was attempted for a call without media ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79739 79739]
|[http://mantis.innovaphone.com/view.php?id=86874 86874]
|}
|}
A CFU loop results in a rejection with busy. A subsequent call completion attempt was allowed and a recall possible was signaled right away. This was very confusing.<br/><!--<br/>pbx.cpp<br/>--><!---->
Could be caused by a misbehaving application or voip device<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== SIP: Handling of Call Pickup ===
=== PBX: Name beginning with '*' caused problems with SOAP applications ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79741 79741]
|[http://mantis.innovaphone.com/view.php?id=86882 86882]
|}
|}
Failed to handle INVITE with Replaces as Call Pickup.<br/>Must redirect Pickup call to alerting party.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
A search for such a name was treated as wildcard search for all users. The name '*' is now a reserved name, which cannot be used for an object.<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== myPBX: Problem with checking browser capabilities in Firefox ===
=== Linux: IP0010 available again ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79757 79757]
|[http://mantis.innovaphone.com/view.php?id=86903 86903]
|}
|}
When checking the supported browser features, Firefox thows an uncaught exception if cookies are deactivatd by the user. Therefore myPBX is stuck in the "Loading" screen instead of displaying a configuration hint.<br/><!--<br/>client.js<br/>client_util.js<br/>--><!---->
With V9 hotfix 16 Linux can not be started on the IP0010. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>linux.cpp<br/>--><br/><!---->
      
      
=== IP241,IP222,IP232: Show lengthy number information on Partner fkey ===
=== phone: call diversion override via indirect dialing could not be disabled ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79779 79779]
|[http://mantis.innovaphone.com/view.php?id=86944 86944]
|}
|}
Toggle between 'show head' and 'show tail' presentation.<br/><br/><!--<br/>forms_color.cpp/h<br/>--><!---->
In the indirect dialing screen the right arrow key opens a menu with different options how to place the call.<br/>"Dial - No Diversion" ssets up a call which will ignore the diversions active on the target phone. This menu item can be supressed via the "Fine grained function locking" bit PHONE_LOCK_DIVERSION_OVERRIDE 0x04000000<br/><br/><!-- phone_dir_ui.cpp phone_config.h --><!---->
      
      
=== SIP: DNS resolving of STUN server failed ===
=== PBX Waiting: CFNR with number filter did not work ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79788 79788]
|[http://mantis.innovaphone.com/view.php?id=86959 86959]
|}
|}
Wrong STUN server port used (5060 instead of real port).<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
The CFNR was not executed<!---->
      
      
=== Fax: Channel memory leak ===
=== PBX Conference: Calling Party was missing in the CDRs for calls to conference members ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79815 79815]
|[http://mantis.innovaphone.com/view.php?id=86977 86977]
|}
|}
Memory leak is fixed in fax channel now.<br/><br/><!--<br/>Changed file:<br/>t30.cpp<br/>--><!---->
Only the conference object itself was present as forwarding party<br/><!--<br/>pbx_bc_conf.cpp<br/>--><!---->
      
      
=== PBX: Export to CSV, utf-8 byte order mark was missing ===
=== SIP: Rare trap when cancelling call ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79844 79844]
|[http://mantis.innovaphone.com/view.php?id=86994 86994]
|}
|}
Without the utf-8 BOM the file was not correctly interpreted as utf-8 by Microsoft Excel and other applications.<br/><!--<br/>pbx_admin.cpp<br/>--><!---->
When call abort interferes with re-routing in gateway application.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== PBX: For registrations containing name and number, use name to identify the device ===
=== PBX: Replication from a dyn PBX was not possible ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79846 79846]
|[http://mantis.innovaphone.com/view.php?id=87065 87065]
|}
|}
It did depend on the sequence of the name and the number. If the number was first, the first device was selected and the name was ignored. Now the name is used to select the device regardless of sequence.<br/><br/>This is a problem with endpoints which always send name and number for registration.<br/><br/><!--<br/>pbx.cpp<br/>--><!---->
Configuration of dyn PBX id now possible for replication<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_general.xml<br/>--><!---->
      
      
=== SIP: CLEARMODE does not work since Hotfix10 ===
=== TLS: Ignore incoming HelloRequest messages ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79858 79858]
|[http://mantis.innovaphone.com/view.php?id=87091 87091]
|}
|}
Invalid SDP answer is send:<br/><br/>   v=0<br/>   o=- 4 3 IN IP4 10.28.108.8<br/>   s=-<br/>   c=IN IP4 0.0.0.0<br/>   t=0 0<br/>    m=audio 0 RTP/AVP 125 101<br/>    a=rtpmap:125 CLEARMODE/8000<br/>    a=rtpmap:101 telephone-event/8000<br/><br/><!--<br/>sip.cpp<br/>--><!---->
When receiving a HelloRequest the TLS client dopped the connection and sent an "Unexpected message" alert.<br/><br/>Now the HelloRequest is ignored and an "No renegotiation" warning is sent.<br/><br/><!--<br/>tls.cpp<br/>unit-tests/tls/basic/520-data-hellorequest.xml<br/>--><!---->
      
      
=== memory violation in http client when processing an URI &gt; 8000 byte ===
=== PBX: Call, which was rejected with busy, because of CF loop, could not be forwarded by Trunk Busy destination ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79860 79860]
|[http://mantis.innovaphone.com/view.php?id=87150 87150]
|}
|}
happens when the digest authentication header is constructed <br/><br/><!--<br/>httpclient_i.cpp<br/>--><br/><br/><br/>  <!---->
Such a call was always disconnected<br/><!--<br/>pbx.cpp<br/>--><!---->
''Status:''
checked in to 10.00, 9.00
      
      
=== PBX Broadcast: Forking/Mobility configured at Broadcast object did not work ===
=== IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: DSP Update to Version 680.05 ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79863 79863]
|[http://mantis.innovaphone.com/view.php?id=87179 87179]
|}
|}
Mobility can be used now for forking with delay<br/><!--<br/>pbx_bc.cpp<br/>pbx_bc.h<br/>pbx.cpp<br/>--><!---->
The jitter buffer performance has been improved.<br/>The fax/modem bypass performance has been improved.<br/>IP28 firmware size was reduced ( unused code is not linked ).<br/>IP28 G279 didnt work with more than 4 channels.<br/><br/><!-- <br/>DSP code details at \\\\innovaphone.sifi\\dfs\\Dokumente\\Hardware\\G7xx\\Audiocodes\\firmware\\firmware_ac49x\\680.05<br/><br/>ac49x_680/*  ac_494e.cpp ip24.mak ip6010.mak --><!---->
      
      
=== Potential trap in packet libraray ===
=== IP222 IP232 IP241: Disable PC port didnt work ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79866 79866]
|[http://mantis.innovaphone.com/view.php?id=87250 87250]
|}
|}
Ther was a not obvious size limit for packet::put_head and packet::put_tail, which caused a trap if the size was exceeded. This could happen if a CDR exceeded a certain size.<br/><!--<br/>packet.cpp<br/>--><!---->
<!-- orchid_drv.cpp --><!---->
      
      
=== SIP: Domain Name System (DNS) names compared case sensitive ===
=== IP-DECT: Login feature with disabled local R-key handling ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79887 79887]
|[http://mantis.innovaphone.com/view.php?id=87269 87269]
|}
|}
Domain Name System (DNS) names are "case insensitive"<br/><br/><!--<br/>sip.cpp<br/>--><!---->
The login feature is fixed now if the local R-key handling is disabled.<br/><br/><!--<br/>Changed file:<br/>dectradio_call.cpp<br/>--><!---->
      
      
=== SIP: Interworking with "AUDC-IPPhone" ===
=== Webdav: DELETE may fail if file has been opened for reading before ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79892 79892]
|[http://mantis.innovaphone.com/view.php?id=87288 87288]
|}
|}
RFC-3551 4.5.2<br/>  Even though the actual sampling rate for G.722 audio is 16,000 Hz,<br/>  the RTP clock rate for the G722 payload format is 8,000 Hz because<br/>   that value was erroneously assigned in RFC 1890 and must remain<br/>  unchanged for backward compatibility.  The octet rate or sample-pair<br/>  rate is 8,000 Hz.<br/><br/>Codec description must be: a=rtpmap:9 g722/8000<br/>but "AUDC-IPPhone" sends: a=rtpmap:9 G722/16000<br/>in SDP offer.<br/><br/><!--<br/>sdpmsg.cpp<br/>--><!---->
If reading (GET) stops but HTTP session remains open, the file remains in state 'open'<br/>and subsequent DELETE request fails with "500 Internal Server Error".<br/>Close file and re-try to delete.<br/><br/><!--<br/>servlet_webdav.cpp/h<br/>--><!---->
      
      
=== SIP: Problems with DNS resolving of proxy adresses ===
=== HTTP: Chunked transfer fails if the chunk header is not in a single packet ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=79907 79907]
|[http://mantis.innovaphone.com/view.php?id=87292 87292]
|}
|}
If resolving of the primary proxy failes, the backup proxy is never resolved.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
<!-- http.cpp --><!---->
      
      
=== IP-DECT: Reset required if Radio password changed ===
=== PBX: Conference and hanging VM script ===


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|[http://mantis.innovaphone.com/view.php?id=87312 87312]
|}
|}
Now reset required is shown if the Radio password for the Master registration is changed.<br/><br/><!--<br/>Changed file:<br/>dectradio.cpp<br/>--><!---->
Wrong media initializing in the BC Conference object causes a hanging VM script if a file should be played. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>pbx_bc_conf.cpp<br/>--><!---->
      
      
=== Phone: PBX (operator) initiated outbound call was connected but mute when another call was ringing on phone already ===
=== IP-DECT: No voice with early handover ===


{|
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|[http://mantis.innovaphone.com/view.php?id=87504 87504]
|}
|}
the phone was not switched from ring mode to handsfree/headset mode <br/><br/><!--<br/>phonesig.cpp<br/>--><!---->
This fixes outgoing calls without voice if early handover is done before the media channel is initialized.<br/><br/><!--<br/>Changed files:<br/>dectmaster.cpp<br/>dectradio_call.cpp<br/>dectradio_dectcall.cpp<br/>--><br/><!---->
      
      
=== SIP: Interworking with HD audio client ===
=== phone: ip222, ip232: multiple reports indicating offhook were misinterpreted and could drop a call ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=79953 79953]
|[http://mantis.innovaphone.com/view.php?id=87506 87506]
|}
|}
Support for unknown codecs.<br/>Passing transparentley.<br/><br/><!--<br/>channel.cpp/h<br/>sdpmsg.cpp/h<br/>sip.cpp<br/>--><!---->
Wireless USB headsets may send more than one report indicating headset offhook state in conjunction with different wireless link states. If the interval between the first and the second indication was very short (8 ms) the second indication was misinterpreted and the just setup call was dropped (observed with a Jabra PRO 930 after plugin). <!---->
      
      
=== Phone: Out-Of-Memory-Trap ===
=== SIP: Fix for Contact-URI in 200/OK for SUBSCRIBE ===


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|}
|}
If LOG server is configured but not reachable<br/>the device will buffer arising LOG entries until a limit of 300kB.<br/>This limit was to high for old black/white telephones.<br/>Now not more then 1% of DRAM size is used for LOG buffer.<br/><br/><!--<br/>logging.cpp<br/>--><!---->
Fix for Contact-URI in 200/OK for SUBSCRIBE and NOTIFY requests.<br/>May contain double port attribue:<br/><br/>SUBSCRIBE sip:8011@172.20.11.53:2053 SIP/2.0<br/><br/>Contact: <sip:8011@172.20.11.53:2053:2053;maddr=172.20.11.53;transport=UDP><br/><br/><!--<br/>sip.cpp<br/>--><!---->
''Status:''
Fixed in 10.00 and 9.00
      
      
=== PBX: When editing a Node object it was changed to a PBX object ===
=== IP-DECT: Disturbances GUI info change ===


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|[http://mantis.innovaphone.com/view.php?id=87522 87522]
|}
|}
This was a collateral damage of fix: #78878: PBX: When doing a show users, unnecessary data was sent to the browser<br/><!--<br/>pbx.cpp<br/>--><!---->
The text of the GUI page disturbances is changed.<br/><br/><!--<br/>New file:<br/>_inn_disturbances.xsl<br/>Changed file:<br/>httpfiles.mak<br/>--><!---->
      
      
=== ISDN: Problem configuring negative volume levels ===
=== IP-DECT: Web UI  info page, version and release state ===


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|[http://mantis.innovaphone.com/view.php?id=87538 87538]
|}
|}
Collateral Damage from<br/>79028: IP-DECT: Feature Code for calling predefined service URIs<br/><!--<br/>config_options.cpp<br/>dect_comcerto/platform/config.h<br/>--><!---->
Now the version and the release state are shown on the web UI info page of the IP1202.<br/><br/><!--<br/>Changed file:<br/>dect_comcerto.cpp<br/>--><!---->
      
      
=== Dect: AD replication makes IPEI disappearing on all users ===
=== PBX: Blind transfer to WQ by IP-DECT caused hanging calls ===


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|[http://mantis.innovaphone.com/view.php?id=80015 80015]
|[http://mantis.innovaphone.com/view.php?id=87542 87542]
|}
|}
Side effect of a previous fix #72672 aiming at ad-replicated and deleted objects coming back to life.<br/><!--<br/>flashdir.cpp<br/>--><!---->
The call-leg to the transfering phone was not cleared by the PBX, so if the phone did not clear this call, it was hanging for ever. Other phones clear such a call after a timeout, but this is only a workaround, the call must be cleared by the PBX<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== myPBX: Setting CFx with boolean did not work in IE8 ===
=== H.323: Problem sending real big signaling messages ===


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|[http://mantis.innovaphone.com/view.php?id=80018 80018]
|[http://mantis.innovaphone.com/view.php?id=87543 87543]
|}
|}
Using Internet Explorer 8 the selected Boolean object was not saved.<br/>Internet Explorer 8 requires option tags to have a value attribute.<br/><!--<br/>client_diversions.js<br/>--><!---->
Happened with a configuration with more the 60 Join Group function keys on the phone. The message was not sent and the operation failed.<br/><!--<br/>h323sig.cpp<br/>--><!---->
      
      
=== IP-DECT: Handover with IP1202 ===
=== PBX-SOAP: UserCall on WQ did not take "Send Number" into account and call was not marked as internal ===


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|[http://mantis.innovaphone.com/view.php?id=87559 87559]
|}
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Only one handover is possible since the last fix with the IP1202. It is fixed now.<br/><br/><!--<br/>Changed files:<br/>dectradio.cpp<br/>dectradio_call.cpp<br/>dectradio_dectcall.cpp<br/>--><!---->
Calls should be send with "Send Number" as source if configured<br/><!--<br/>pbx_xml.cpp<br/>--><!---->
      
      
=== Name and Number configured for a registration must be completely replaced by registration result ===
=== PBX: CFB configured at PBX object did not patch number correctly ===


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|[http://mantis.innovaphone.com/view.php?id=87581 87581]
|}
|}
If for example Number and Name is configured but the registration result provides the Number only, the configured Name must not be used in further signaling operations (diversion queries ...) <br/><br/><!--<br/>phonesig.cpp<br/>--><!---->
In case the called endpoint was in different node then PBX object, the number did not contain all the needed prefixes<br/><!--<br/>pbx.cpp<br/>--><!---->
''Status:''
checked in to 10.00, 9.00
      
      
=== PBX: Possible trap on calls from misconfigured nodes/PBXs (node parent loop) ===
=== SIP: Trap when logging out dect user ===


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|[http://mantis.innovaphone.com/view.php?id=87606 87606]
|}
|}
If a node or PBX is configured with a parent node configured to itself in the most simple case, a call from an endpoint configured for this node to a destination which cannot be found in this node, will cause a trap.<br/><br/>This is a collateral damage of fix: #79317: PBX: Local objects could not be called from Nodes with escapes as expected<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>--><!---->
Trap when logging out dect user.<br/><br/>0:0287:977:1 - MASTER_EP-SIG.0 default(948f3e18): serial_event(100) src=DECTMASTER-CALL.0 mod=SIP<br/>0:0287:977:1 - Assertion failed line 790 in common/os/os.cpp, object deleted<br/>0:0287:977:2 - assert-ep 94004474  called from 94055498<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305 IP222 IP232 IP241: receveid RTP packets limited to 480bytes ===
=== PBX: Called number was missing in CDRs for calls to busy endpoint ===


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|[http://mantis.innovaphone.com/view.php?id=87620 87620]
|}
|}
increased to 640 Bytes to allow G711 80ms.<br/><br/><!-- ac_phone3.h  ac_dsp3.h --><br/><!---->
In case there was no other event then rel-to/from and the call was dialed with overlap sending<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== DHCP Survivability mode doesn't work after a DHCP restart ===
=== SIP: Handset type information also in Subscribe User-Agent ===


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|[http://mantis.innovaphone.com/view.php?id=87621 87621]
|}
|}
``Survivability´´ mode is used by WLAN phones. In this mode the settings of the 'saved lease' (IP addr, mask, ...) are used until a fresh lease is received. The 'saved lease' is the last lease received from a server, it is kept over a reboot.<br/><br/>When WLAN coverage is lost for a while and then regained a DHCP restart is requested to get a fresh lease from a server in a possibly different network. If this happened while using the 'saved lease' the phone lost it's (saved) IP address.  <!---->
Handset type information also in Subscribe User-Agent as sent in REGISTER request.<br/><br/><!--<br/>siptrans.cpp/h<br/>sip.cpp/h<br/>--><br/><!---->
''Status:''
checked in to 10.00, 9.00
      
      
=== AC-DSP3: Switch trace off if the DSP Host interface shows an error ===
=== H.323: Media Negotiation did not work for Hold/Retrieve after a DECT conference ===


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|[http://mantis.innovaphone.com/view.php?id=87635 87635]
|}
|}
<!-- ac_dsp3.cpp --><!---->
No media after retrieve<br/><!--<br/>h323ch.cpp<br/>--><!---->
      
      
=== SIP: Keep backup registration while calls are active ===
=== PBX: Conference trap ===


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|[http://mantis.innovaphone.com/view.php?id=87662 87662]
|}
|}
Terminating backup registration will active calls (through backup system) get disconnected.<!---->
A trap in the BC conference PBX object is fixed.<br/><br/><!--<br/>Changed file:<br/>pbx_bc_conf.cpp<br/>--><!---->
      
      
=== SIP: Content of Allow-Events header must be treated case-insensitive ===
=== PBX Waiting: Missing ringback on call forward after announcement ===


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|}
|}
Event names are case insensitive.<br/><br/><!--<br/>sipmsg.cpp<br/>--><!---->
This was a collateral damage of<br/><br/>fix: #81370: PBX Waiting: Call state shows "Disconnecting" after switch from announcement 1 to announcement 2<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
      
      
=== IP-DECT/Analog Features: Call Park should be done on the last active call ===
=== Gateway: Routing on blind transfer call starts from wrong IF (undo of  #86386) ===


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|[http://mantis.innovaphone.com/view.php?id=87678 87678]
|}
|}
Szenario is an active call, then a waiting call comes in, which is accepted, then call park is executed. This call park should be done on the accepted waiting call and not the original.<br/><!--<br/>dtmffty.cpp<br/>--><!---->
Rollback of #86386: Gateway: Routing on blind transfer call starts from wrong IF<br/><br/>There are other installations where the old behaviour is the expected behaviour.<br/><br/><!--<br/>relay.cpp<br/>--><!---->
      
      
=== IP-DECT/Analog Features: Call completion ===
=== PBX Waiting: DTMF overlap dialing or blind transfer to same Waiting object was rejected with busy ===


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|[http://mantis.innovaphone.com/view.php?id=87681 87681]
|}
|}
The type of the call is changed back to normal state if the call completion is executed, and facility conversion is added for the call completion state. <br/>This fixes the reusing of features for a call completion callback call, used if IP-DECT/analog features are enabled. This also fixes missed remote hold and retrieve events to the gatekeeper.<br/><br/><!--<br/>Changed file:<br/>dtmffty.cpp<br/>--><!---->
Even if this was caused by a CFB or CFU on the dialed destination<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
      
      
=== SIP: Handling of 488 for encrypted media ===
=== IP-DECT: No idle display update with own message ===


{|
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|}
|}
If SRTP offer is rejected by remote endpoint with "SIP/2.0 488 Not Acceptable Here"<br/>we should re-try offer without encryption keys.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
A message with the same sender and receiver name overwrites the handset's idle display. This feature is removed now.<br/><br/><!--<br/>Changed file:<br/>dect_comcerto/platform/config.h<br/>--><!---->
      
      
=== Phones: Show 'tel' presence on configuration screen ===
=== Voicemail: &lt;pbx-fwd&gt;, Forward Timer Didn't Fire For Connected Calls ===


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|}
|}
Show latest presence on 'Presence' fkey,<br/>but show 'tel' presence on phone config menu.<br/><br/><!--<br/>app_ctl.h<br/>app_gui.cpp<br/>app_reg.cpp<br/>phonesig_if.h<br/>phonesig.cpp<br/>--><!---->
Voicemail: <pbx-fwd>, Forward Timer Didn't Fire For Connected Calls<br/><!-<br/>pbx_vm.cpp<br/>--><!---->
      
      
=== PBX-Waiting: Calls to a Waiting queue object in altering state could not be cleared with SOAP ===
=== IP-DECT: Trap with rejected handovers ===


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|}
|}
A SOAP UserClear issued for the call monitored at the Waiting Queue object didn't do anything<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
A trap occurs in the rare situation if the handover is rejected by the radio.<br/><br/><!--<br/>Changed file:<br/>dectradio.cpp<br/>--><!---->
      
      
=== Phones: Sometime fkey could not be deleted ===
=== H.323: Q.931 User Info could not be sent in incoming call proceeding state ===


{|
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|}
|}
Sometime fkey could not be deleted when using the fast edit mode (long fkey press).<br/><br/><!--<br/>phone_user_func.cpp<br/>phone_config.cpp<br/>--><!---->
This is used by the Fax interface<br/><!--<br/>h323_tbl.tbl<br/>--><!---->
      
      
=== IP-DECT: Handset display ===
=== SIP: Parsing of multiple Allow lines was not implemented ===


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|[http://mantis.innovaphone.com/view.php?id=87753 87753]
|}
|}
Names with special characters are not correctly shown in the handsets. This is fixed in the IP1200 now.<br/><br/><!--<br/>Changed file:<br/>dectdisplay.cpp<br/>--><br/><!---->
Now all Allow lines are parsed (not only first one).<br/><br/><!--<br/>sipmsg.cpp/h<br/>--><!---->
      
      
=== IP210 IP230 IP240: Handset receiver volume increased, especially at lower frequencies ===
=== syslog packets were sent with the initially assigned source address even after a new address had been assigned ===


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|[http://mantis.innovaphone.com/view.php?id=87754 87754]
|}
|}
<br/>For better performance with jinlida receiver.<br/><br/><!-- inca_afe.cpp --><!---->
When a device runs as DHCP client the IP-adress assigned to an interface may change either because the DHCP-Server rejects a renew request and provides a new lease or because a WLAN device enters another network.<br/>An IP-address change may also happen when the DHCP mode of a device is changed from 'disabled' to 'client' without reboot.<br/>The source address of syslog packets does reflect such changes now. <br/><br/><!-- logging.cpp logging.h -->  <br/><!---->
      
      
=== Gateway: Pass through ctSetup facility ===
=== Linux: Trap of IP810 if Linux is started ===


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|[http://mantis.innovaphone.com/view.php?id=80437 80437]
|[http://mantis.innovaphone.com/view.php?id=87763 87763]
|}
|}
Lync sends diverting party information inside Referred-By header.<br/>Referred-By is interworked to ctSetup facility.<br/>ctSetup facility needs to be passed through by Gateway application.<br/><br/><!--<br/>relay.cpp/h<br/>--><!---->
The IP810 with the V9 hotfix 16 firmware traps if Linux is started. Please do not use this firmware. It is fixed again now.<br/><br/><!--<br/>Changed file:<br/>config.h<br/>--><!---->
      
      
=== Linux: IP address with external DHCP server ===
=== PBX: CFNR to same node but different PBX failed on Slave ===


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|[http://mantis.innovaphone.com/view.php?id=80471 80471]
|[http://mantis.innovaphone.com/view.php?id=87772 87772]
|}
|}
Assigning a IP address to Linux by a external DHCP server is not working if the network interface which is used is configured with a fix IP address (DHCP disabled). This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dhcp.cpp<br/>--><!---->
The PBX routing did not work correctly in this case<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== IP232: Invalid text on second line of fkey label ===
=== PBX administration: Viewer could change phone configuration of users ===


{|
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|[http://mantis.innovaphone.com/view.php?id=80484 80484]
|[http://mantis.innovaphone.com/view.php?id=87774 87774]
|}
|}
Invalid text on second line of fkey label when changing registrations.<br/><br/><!--<br/>forms_color.cpp<br/>--><!---->
<!-- pbx_phone.cpp pbx_phone.h --><!---->
      
      
=== PBX: Routing problem from sub-slave to master to object in sub-slave node, but registered to master ===
=== SIP: Don't try to dns-resolve IP adresses ===


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|[http://mantis.innovaphone.com/view.php?id=87781 87781]
|}
|}
Collateral damage from #77874: PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX<br/><!--<br/>pbx.cpp<br/>--><!---->
Don't try to dns-resolve IP adresses.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== SIP: Calls may remain in clearing state ===
=== phone: Function key options which need to be enabled in Phone/Userx/Preferences could be edited on phone although disabled ===


{|
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|[http://mantis.innovaphone.com/view.php?id=80623 80623]
|[http://mantis.innovaphone.com/view.php?id=87792 87792]
|}
|}
SIP calls may remains undeleted.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
Partner-Intrude: hide when "Phone/Userx/Preferences/Enable Call Intrusion" is not checked  <br/>Dial-Announce: hide when "Phone/Userx/Preferences/Announcement Calls/Outgoing/Allow" is not checked<br/><br/><!-- phone_user_func.cpp phone_user_config.h --><br/><!---->
      
      
=== SIP: Code optimization ===
=== SIP: Error handling huge INVITE requests ===


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|[http://mantis.innovaphone.com/view.php?id=80635 80635]
|[http://mantis.innovaphone.com/view.php?id=87811 87811]
|}
|}
Reduce number object constructions/destructions during message encoding.<br/><br/><!--<br/>siptrans.cpp<br/>--><!---->
Failed to compose provisional and final response.<br/>Lots of error messages in trace and truncated responses.<br/><br/>  ERROR: SIP message buffer (2049) exceeded! (40,35,941b0c90)<br/><br/><!--<br/>siptrans.cpp<br/>--><!---->
      
      
=== PBX: Update of conference id did not always work after multiple transfer accross PBXs ===
=== Gateway: MOH from call on hold in CONF interface calls ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=80656 80656]
|[http://mantis.innovaphone.com/view.php?id=87823 87823]
|}
|}
A scenario which did not work was A calls B, B does consultation to C, B Transfers, C does consultation to D, C transfers with B on different PBX then A.<br/>After this the conference id on the call on A should be identical to the conference id on D. This was not the case.<br/><br/>SOAP/TAPI applications which are keeping track of transfered calls could have a problem with this.<br/><!--<br/>pbx.cpp<br/>--><!---->
If a new call joins the conference or a call on hold retrieves the conference, and there are calls on hold in the conference, the music on hold can be heard in the new or retrieving calls. Now this is fixed.<br/>This affects all devices with a CONF interface, but not the IP800 and the IP305.<br/><br/><!--<br/>Changed files:<br/>adsp2191_drv.h<br/>adsp2191_drv.cpp<br/>--><!---->
      
      
=== AD Replication: LDAP filter encoding failed, when Poll Timer was configured ===
=== IP241,IP222,IP232: Presence note may is not enough truncated on 'presence' fkey ===


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|[http://mantis.innovaphone.com/view.php?id=87825 87825]
|}
|}
AD Replication: LDAP filter encoding failed, when Poll Timer was configured<br/><!--<br/>adrep.cpp<br/>--><br/><!---->
Presence note may is not enough truncated on 'presence' fkey.<br/>Esp. when there's also an icon to be displayed.<br/><br/><!--<br/>forms_color.cpp<br/>--><!---->
      
      
=== PBX: More consistent use of conferenceId after transfer ===
=== PBX Trunk: If a call was forked to a trunk with no-name option, name was removed from the original call as well ===


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|[http://mantis.innovaphone.com/view.php?id=87843 87843]
|}
|}
The conference ID is used (SOAP/TAPU, CDRs) to associate different call legs to the same call. After a transfer two calls, which have been seperate are connected, so one of the call legs has to change its conference ID, so that the resulting call has a single conference ID again.<br/><br/>There was a complicated logic implemented in the PBX to decide which conference ID should be used, this is now changed to a simple logic: The conference ID of the call on which the transfer is performed, is used.<br/><br/>Example:<br/><br/>A calls B, B does a consuktation to C, and B transfers A to C - This means the transfer is performed on call leg A, so the conference ID of the original call A-B is used for A-C<br/><!--<br/>pbx.cpp<br/>--><!---->
A global flag was set on the incoming call, which was wrong.<br/><!--<br/>pbx_trunk.cpp<br/>--><!---->
      
      
=== Gateway: Routing of incoming SIP calls may not work ===
=== IP230 IP240: Handset receiver volume increased ===


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|}
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... because the To-URI is not reduced to a called number (CDPN).<br/>Gateway routing is based on CDPN and does not work for URIs.<br/><br/>Before this fix:<br/>.LOG CALL 2 Alloc<br/>.LOG CALL 2 A:Call    ->                        / GW8::->*::<br/>.LOG CALL 2 B:Call    :081604998@212.13.249.90->:018108680@192.168.5.230 / GW8::->GW1::<br/><br/><br/>After this fix:<br/>.LOG CALL 1 Alloc<br/>.LOG CALL 1 A:Call    ->                        / GW8::->*::<br/>.LOG CALL 1 B:Call    081604998->018108680      / GW8:081604998:->GW1:018108680:<br/><br/><br/><!--<br/>sip.cpp<br/>--><!---->
receiver volume adapted to changed equalizer settings from fix #80403<br/><br/><!-- inca_afe.cpp --> <br/> <!---->
      
      
=== SIP: Memory leak when closing signaling interface ===
=== phone: ip222, ip232: better handling of spurios headset disconnects ===


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|}
|}
Memory leak when closing signaling interface while DNS request is pending on a call.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
Sometimes USB headsets get disconnected from USB port because of certain electric pulses. To overcome this problem the headset port is reset and the media stream routed to the handset. If the headset comes up (logical plugged) again in a reasonable time the media stream is routed to the headset again. Otherwise the media stream remains on the handset and the call can be continued by taking off the handset.<br/><br/><!-- keys_ip222.cpp ohci.cpp --><br/><!---->
      
      
=== Phone: Message function key cannot be configured ===
=== PBX Trunk: Diverting as Calling Feature should replace the name as well ===


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|[http://mantis.innovaphone.com/view.php?id=87913 87913]
|}
|}
Configuration is not saved<br/><br/><!--<br/>phone_edit.cpp<br/>fkey_edit_message.xsl<br/>--><!---->
With this feature the Trunk object uses a Diverting Number as calling party number. But not only the number, but also the Name and Name Id should be replaced<br/><!--<br/>pbx_trunk.cpp<br/>pbx.cpp<br/>--><!---->
      
      
=== SIP: Diversion information was wrong on incoming SIP calls ===
=== Phones: Allow lcd_dump.bmp to be retrieved with viewer credentials ===


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|[http://mantis.innovaphone.com/view.php?id=87945 87945]
|}
|}
Parsing of History-Info header was wrong, but only if multiple History-Info headers were present.<br/><br/><!--<br/>sipmsg.cpp/h<br/>--><!---->
Allow lcd_dump.bmp to be retrieved with viewer credentials.<br/><br/><!--<br/>lcd_dump.bmp<br/>--><br/><!---->
      
      
=== PBX: Routing problem with nodes/escapes/slaves with calls to extern numbers in other PBX defined nodes ===
=== IP-DECT: Wrong radio list after MAC-alias change ===


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|}
|}
Could happen that the number dialed when sending the call to the extern interface was not adjusted correctly<br/><!--<br/>pbx.cpp<br/>--><!---->
The radio list can be wrong after the MAC-alias change. This fixes a bug of the feature "MAC-alias change of OEM device" (#86047). This is only relevant for OEM devices.<br/><br/><!--<br/>Changed file:<br/>dectmaster.cpp<br/>--><!---->
      
      
=== Build Number format changed ===
=== SIP: Keep Contact-URI when registering via TCP ===


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|}
|}
The build number of the hotfixes changes from the 90600.xx format to the 9.061xxx format. This is due to organizational changes without any other significance.<br/><!--<br/>box.cpp<br/>--><!---->
Keep Contact-URI when registering via TCP, even if rport is present in 200/OK for REGISTER<br/><br/><!--<br/>sip.cpp<br/>--><br/><!---->
      
      
=== IP-DECT: Memory leak for SIP calls ===
=== PBX-SOAP: UserCall with 'cn' as destination to a user in different node did not work ===


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|[http://mantis.innovaphone.com/view.php?id=88008 88008]
|}
|}
buffer for received name-id was not freed. Could happen with other facilities as well.<br/><!--<br/>dtmffty.cpp<br/>--><!---->
The call is done using the number, but the number needs to be adjusted according to the nodes<br/><!--<br/>pbx_xml.cpp<br/>--><!---->
      
      
=== SIP: Transcation handling was wrong ===
=== OS: Quota mechanism did not work for CPU time consumed by timer handling ===


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|[http://mantis.innovaphone.com/view.php?id=80989 80989]
|[http://mantis.innovaphone.com/view.php?id=88034 88034]
|}
|}
Double delete on a INVITE client transaction object.<br/><br/>0:0027:988:5 - SIP_TAC_INVITE:serial::delete(805b0c80) caller=800440ec<br/><br/><!--<br/>siptrans.cpp<br/>--><!---->
This could result in a MAX_BUSY_TICK restart if many timer expired at the same time<br/><!--<br/>os.cpp<br/>command.cpp (for test only)<br/>command.h (for test only)<br/>--><!---->
      
      
=== PBX: Adjusting called number, when sending call to extern ===
=== HTTP: Chunked transfer sporadic fails with webdav ===


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|[http://mantis.innovaphone.com/view.php?id=81005 81005]
|[http://mantis.innovaphone.com/view.php?id=88078 88078]
|}
|}
depending on the node of the extern object and the called node, the called party number has to be adjusted (escapes added, prefixes added/removed). This did not work unders some conditions.<br/><!--<br/>pbx.cpp<br/>--><!---->
<!-- http.cpp --><!---->
      
      
= V9 Hotfix 13 (9061024) =
=== IP6000 IP2000: Webinterface for ETH1 Link setting didnt work ===
Changes included in Version 9 hotfix13
[http://mantis.innovaphone.com/view.php?id=81128 Definition]
 
== New Features ==
 
 
   
=== myPBX: Interface for tracing ===


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|}
|}
The web application can now write messages into the trace file of the myPBX launcher.<br/><!--<br/>client.js<br/>client_util.js<br/>--><!---->
..<br/><!-- interfaces.xml --><!---->
      
      
=== Oem Code: Accelerating Boot Snmp Traps ===
=== SIP: Decoding problem on application/simple-message-summary ===


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|[http://mantis.innovaphone.com/view.php?id=88188 88188]
|}
|}
Oem Code: Accelerating Boot Snmp Traps<!---->
application/simple-message-summary may get decoded wrongly.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== PBX: Handling of call limits at PBX objects improved ===
=== Voicemail: Possibility To Switch Off An Internal Automatism ===


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|}
|}
A call from a PBX, which is sent back to the same PBX is not counted anymore. This can happen because of node-extern.<br/>Incoming calls at master, which are above the limit are rejected now. They can be rerouted on the slave with "Route Master calls if no Master to"<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_api.h<br/>--><!---->
An internal automatism could lead to VM-calls getting disconnected after 15s.<br/>The new URI variable "$_divconn=false" turns off auto-connection for diverted/transferred calls:<br/>http://wiki.innovaphone.com/index.php?title=Howto:Configure_the_innovaphone_Voicemail#URL_Query_String_Variables<br/><!--<br/>pbx_vm.cpp/h<br/>--><!---->
      
      
=== SIP: New config option for endpoints not refreshing their registration during call ===
=== SIP: DNS priority value not honored always ===


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|}
|}
Interoperability:<br/>New config option /keep-active-endpoints for endpoints not refreshing their registration during call.<br/><br/>AVM FRITZ!Box Fon WLAN 7270 v2 (UI) 54.05.21 (Apr  2 2012)<br/><br/><!--<br/>sip.cpp/h<br/>siptrans.cpp<br/>--><!---->
DNS priority value not honored, if SRV query returns names without address in additional records.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== UI hint where to update Linux AP ===
=== SIP: Do not take "9564+4631559300" as E.164 number ===


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|[http://mantis.innovaphone.com/view.php?id=88254 88254]
|}
|}
Changed a hint on the upload/update tab and added a link to the update/upload page on the firmware side.<br/><br/><!--<br/>lang*.inc<br/>application.php<br/>linux.xsl<br/>--><!---->
Do not take "9564+4631559300" as E.164 number.<br/><br/><!--<br/>sip.cpp<br/>siputil.cpp/h<br/>--><!---->
      
      
=== SIP: New config option /no-certificate-check ===
=== PBX Boolean: Unpredictable behaviour if more the 16 times are entered ===


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|[http://mantis.innovaphone.com/view.php?id=88263 88263]
|}
|}
New config option to disable validation of remote certificate name<br/>when opening outbound TLS connection.<br/><br/><!--<br/>sip.cpp/h<br/>siptrans.cpp<br/>--><!---->
There was a limit of 16 times, which was not checked everywhere. The limit is now 32 and checked, so that no more of 32 times can be entered.<br/><!--<br/>pbx_bool.cpp<br/>pbx_bool.h<br/>--><!---->
      
      
=== IP-DECT: Own priority for idle display set by messages ===
=== IP222 IP232 IP241: Restart on Jitter buffer overrun ===


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|}
|}
The idle display set by messages has got an own priority now.<br/><br/><!--<br/>Changed files:<br/>dtmffty.h<br/>dtmffty.cpp<br/>--><!---->
..<br/><!-- ac494e.cpp --><!---->
      
      
=== SoftwarePhone: Support for Jabra PRO 930 ===
=== SIP: Fix for auto-answer handling ===


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|}
|}
Support for Jabra PRO 930 with product id 0x1016 added.<br/><br/><!--<br/>Changed file:<br/>hid.tbl<br/>--><!---->
Fix for auto-answer handling.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== SIP: New config option "No Inband Disconnect" on GW interfaces ===
=== IP-DECT: SIP/Intop - rejected call transfer by target ===


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|[http://mantis.innovaphone.com/view.php?id=88356 88356]
|}
|}
Introduced VOIP_OPTION_NO_INBAND_DISC.<br/>Option was missing on Gateway interfaces.<br/><br/><!--<br/>voip.h<br/>sip.cpp<br/>gk.cpp/h<br/>relay_edit_voip.xsl<br/>--><!---->
If the call transfer target rejects the call in ringing state, no fall-back to the initiator call is done and it is not released. This is fixed now. It is only important for a third party PBX.<br/><br/><!--<br/>Changed files:<br/>dectradio_call.cpp<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp<br/>--><!---->
      
      
=== SIP: New config option /product-id-format ===
=== H.323: Problem with Media Re-Negotiation on a DECT handover call ===


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|}
|}
Change the User-Agent string from<br/>  User-Agent: (innovaphone IP232/10.00 dvl [90910/90879/501])<br/>into<br/>  User-Agent: innovaphoneIP232x90910x501<br/>with /product-id-format 1<br/><br/><!--<br/>sip.cpp/h<br/>siptrans.cpp<br/>sipmsg.cpp<br/>--><!---->
The DECT handover call works a little special concerning media renegotiation in a way that local preferences are never honored (the real media negotiation takes place between the original radio and the remote endpoint, the handover radio is just told the result. This special mode did not work correctly<br/><!--<br/>h323ch.cpp<br/>dectradio_ho_out.cpp<br/>dectradio_ho_out.h<br/>--><!---->
      
      
=== phone: ip222,ip232: more USB headsets supported ===
=== PBX Waiting: User Information Message from announcement interface accidentally forwarded to caller ===


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|[http://mantis.innovaphone.com/view.php?id=88426 88426]
|}
|}
Sennheiser and some more Jabra Headsets added. Look for "(since V9hotfix13)" in<br/>http://wiki.innovaphone.com/index.php?title=Reference9:Concept_USB_Headset<br/><br/><!-- keys_ip222.cpp --><!---->
The announcement interface uses User Information signaling messages to send status information for example at the end of the announcement. This was forwarded to the caller by accident. Usually this does not do any harm, but on some ISDN networks it could result in clearing of the call because of unexpected message.<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
      
      
=== Debug information on assertion ===
=== H.323: Potential Max-Busy-Ticks restart ===


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|}
|}
More debug information on default event handler.<!---->
The H.323 state machine could enter a endless loop<br/><!--<br/>h323sig.cpp<br/>--><!---->
      
      
=== phone: ip222,ip232: support for advanced USB headset functions (redial, reject call, accept waiting call) ===
=== IP-DECT: No media after conference and toggled to held call ===


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|}
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A lot of USB headsets generate special indications to request redialing of last number dialled, to reject a ringing call, to accept a waiting call and to put the active call on hold or to switch between an active and an held call.<br/><br/><!--<br/>keys_gen.cpp<br/>keys_ip222.cpp<br/>forms_color.cpp<br/>forms_impl.cpp<br/>forms_keyboard.cpp<br/>keyboard.h<br/>app_call.cpp<br/>app_reg.cpp<br/>app_ctl.cpp<br/>app_ctl.h<br/>--><br/><br/><!---->
If the DECT user leaves the conference mode and toggles to an held call, there is no voice. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>decradio_mastercall.h<br/>decradio_mastercall.cpp<br/>--><!---->
      
      
== Bug Fixes ==
=== Admin IP: static routes configured at an ETH interface disappear when NAT or VLAN config of this interface is updated ===
 
 
   
=== IP222  equalizer update ===


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|}
|}
- handsfree speaker equalizer enabled<br/>- handset mic and receiver equalizer smoothed<br/>- ADC gain reduced, input gain increased ( after ec ) to avoid clipping<br/>- halfduplex mode disabled<br/><br/><!-- ac_codec3.cpp platform/filter/* --><!---->
Routes configured under "IP4/ETHn/IP/Static IP Routes" were cleared when the "IP4/ETHn/NAT" page or the "ETHn/VLAN" page was left by pressing "OK".<br/><br/><!-- ip_ethif_nat.xsl ip_ethif_vlan.xsl -->   <!---->
      
      
=== IP222 IP232 IP241: repeated ethernet link status 1000M wrong ===
=== IP232,IP222,IP241: Reduce memory requirements of display rendering ===


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|[http://mantis.innovaphone.com/view.php?id=88485 88485]
|}
|}
.<br/><br/><br/><!-- orchid_drv.cpp --><!---->
Reduce memory requirements of display rendering.<br/><br/><!--<br/>lcd_map.cpp/h<br/>forms_color.cpp/h<br/>read_png.cpp<br/>--><!---->
      
      
=== IP222 IP232 : Default LCD backlight standby brightness reduced ===
=== PBX: Call Completion monitoring call should be terminated only after the CC ringout call is alerting ===


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|[http://mantis.innovaphone.com/view.php?id=80031 80031]
|[http://mantis.innovaphone.com/view.php?id=88489 88489]
|}
|}
<br/><!-- config.h --><!---->
This is needed for QSIG interoperability<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>--><!---->
      
      
=== IP241 handsfree equalizer update, IP222 handsfree mic equalizer update ===
=== Viewer was able to download config with standard password ===


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|[http://mantis.innovaphone.com/view.php?id=88503 88503]
|}
|}
- IP241 handsfree speaker equalizer enabled<br/>- IP241 handsfree micro equalizer enabled<br/>- IP222 handsfree micro equalizer enabled <br/>- All three filter calculated with measured frequency response <br/>- IP241 ADC gain reduced, input gain increased ( after ec ) to avoid clipping<br/><br/><br/><!-- ac_codec3.cpp platform/filter/* --><!---->
This was a security hole<br/><!--<br/>cfg-standard.txt<br/>--><!---->
      
      
=== phone: ip222, ip232: Pressing Talk button on USB Headset when there is an active and a held call does not transfer ===
=== Gateway: Hanging calls in state "Clearing" ===


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|[http://mantis.innovaphone.com/view.php?id=88527 88527]
|}
|}
Instead of transfer the active call is disconnected and the held call is retrieved<!---->
Hanging calls in state "Clearing".<br/>Caused by failed call replacement.<br/><br/><!--<br/>relay.cpp<br/>--><!---->
''Status:''
checked in to 10.00
      
      
=== IP241,IP222,IP232: Show name of diverting party in incoming calls ===
=== IP-DECT: Potential buffer overrun ===


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{|
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|[http://mantis.innovaphone.com/view.php?id=80839 80839]
|[http://mantis.innovaphone.com/view.php?id=88535 88535]
|}
|}
Show name of diverting party in incoming calls (not only number).<br/><br/><!--<br/>forms_color.cpp/h<br/>call-ctrl-ring.png<br/>call_diverted.png<br/>call_transfered.png<br/>backgrounds_320x240.mak<br/>backgrounds_480x272.mak<br/>icons.mak<br/>--><!---->
Buffer overrun check added in dectusers module for command 'show'.<br/><br/><!--<br/>Changed file:<br/>dectusers.cpp<br/>--><!---->
      
      
=== PBX-SOAP: UserRedirect - original called number got lost ===
=== SIP: Missing UPDATE on call pickup ===


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|}
|}
For example if an application used an Waiting Queue object to monitor for incoming calls and redirected these calls to agents. The agent receiving the call could not see if the call was diverted to the waiting queue already.<br/><!--<br/>pbx_xml.cpp<br/>--><!---->
The caller does not received UPDATE containing the connected party information.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== SIP: Send 200/OK for MESSAGE(text/plain) when accepted by application ===
=== SIP: Handle one way of DTMF only ===


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|}
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Application gives "NormalCallClearing" to SIP stack.<br/>SIP stack better sends "200 OK" instead of "603 Decline".<br/><br/><!--<br/>sip.cpp<br/>--><!---->
Handle DTMF received via signaling message (INFO)<br/>or DTMF received via RTP (tlephone-event).<br/>Not both at the same call.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== SIP: Un-escape content of XML elements ===
=== PBX Conference: No Media in case media offer only received with alert/connect from called member ===


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|}
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XML element content requires some resevered characters to be escaped (<>).<br/>These escape sequences (&amp;gt; or &amp;lt;) must be un-escaped onthe receiving end.<br/><br/><!--<br/>xml.cpp/h<br/>sip.cpp<br/>soap.cpp<br/>fault_handler.cpp<br/>webex.cpp<br/>pbx_bool.cpp<br/>pbx_dtmf.cpp<br/>pbx_mwi.cpp<br/>pbx_vm.cpp<br/>pbx_wait.cpp<br/>--><!---->
This could happen if a broadcast object was called by a conference<br/><!--<br/>pbx_bc_conf.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Two status symbols may overlay each other ===
=== IP222 IP232 IP241: Adjust equalizer above 6.3Khz for lower noise level ===


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|}
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Symbol for "Call diversion" and symbol for "PIN-locked" overlay each other.<br/>Hide "Call diversion" as long as phone is PIN-locked.<br/><br/><!--<br/>forms_color.cpp<br/>--><!---->
to reduce noise with G722 codec<br/><br/><!-- ac_codec3.cpp platform/filter --><br/><!---->
      
      
=== SIP: Handling of multiple 401/407 responses ===
=== SIP: Memory leak ===


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Implemented handling of up to 3 401/407 responses per transaction.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
Memory leak on postponed RETRIEVE_NOTIFY.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== SIP: Trap while releasing call ===
=== SIP: Invalid SDP answer if SDP offer contains RED ===


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Timer fires during call release and causes trap.<br/><br/><!--<br/>sip.cpp<br/>sipstate.cpp<br/>--><!---->
Invalid SDP answer if SDP offer contains RED.<br/>Only on local media.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
= V9 Hotfix 18 (9061158) =
Changes included in Version 9 hotfix18
[http://mantis.innovaphone.com/view.php?id=88994 Definition]
 
== New Features ==
 
 
      
      
=== H.323: Memory leak when sending special OEM H.235 key elements ===
=== IPVA: Enhancement for innovaphone testbed: Indicate whether Upload to DRAM took place ===


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|}
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Only happens if Avaya SRTP is enabled in OEM products<br/><!--<br/>h323ch.cpp<br/>--><!---->
Required for innovaphones' automated testbed<br/><!--<br/>ipva.cpp<br/>--><!---->
      
      
=== H.323: Potential Trap in case of high load and media renegotiation/call clearing collision ===
=== Announcement Calls with DTMF feature code object ===


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There is a very unlikely situation when media-renegotiation is started and then the call is cleared, which could cause a message related to the media-renegotiation to be sent to a already deleted call object. High load could make this situation more likely.<br/><!--<br/>h323sig.cpp<br/>--><!---->
A new DTMF feature code allows to make announcement calls to a dialed number.<br/><br/><!--<br/>pbx_dtmf.cpp<br/>pbx_dtmf.h<br/>pbx_edit_dtmf-ctrl.xsl<br/>--><!---->
      
      
=== IP241 handset equalizer ===
=== Alarm and Event forwarding to a SYSLOG server ===


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|}
|}
Enable IP241 handset micro and speaker equalizer.<br/>Same values as on IP222.<br/>Reduces noise in G711 mode. <!---->
If "Services/Logging/Alarm and Event Forward Server/Type" is set to SYSLOG the xml-formatted alarm and event info is sent to the Server(s) specified under "Services/Logging/Alarm and Event Forward Server/Address".<br/><br/><!-- logging.cpp logging.h logging.xsl fault_handler.cpp fault_handler.h --><br/><!---->
   
== Bug Fixes ==
 
 
      
      
=== IP1060/3010/6010/22/24/28/302/305: sequence number to DSP changed from byte to word, caused jiter buffer warnings ===
=== TLS: Do not ignore early SOCKET_RECV ===


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|}
|}
<!-- ac_dsp3.h--><!---->
Allow SOCKET_RECV from application between SOCKET_CONNECT and SOCKET_CONNECT_COMPLETE. This event flow is used by the HTTP client.<br/><!--<br/>tls.cpp<br/>unit-tests/tls/basic/102-handshake-client-normal-early-recv<br/>--><br/><!---->
      
      
=== IP222/232/241/1060/3010/6010/22/24/28/302/305: min jitter buffer changed to 10ms ===
=== Kerberos: Admin UI trap when having too many Kerberos hosts ===


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was 35ms, now 10ms as on ip6000/800<br/><br/><!-- ac_dsp3.cpp ac_phone3.cpp --><!---->
The problem occured if many Kerberos hosts (~1000) were registered on the server. In this case the box trapped due to an XML encoding problem when opening the page General/Kerberos or PBX/Config/Security.<br/><!--<br/>kerberos_db.cpp<br/>kerberos_ldap.cpp<br/>--><!---->
      
      
=== IP-DECT: Duplicate call setup ===
=== SIP: Provide display name in 200/OK for SUBSCRIBE ===


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The IP-DECT Master sends in some circumstances a call twice to the same radio in the same time. This affects only the IP1202 and OEM devices, not the IP1200, and is fixed now.<br/><br/><!--<br/>Changed file:<br/>dectmaster.cpp<br/>--><!---->
Add display name to To header in 200/OK for SUBSCRIBE.<br/><br/><!--<br/>sip.cpp/h<br/>sipmsg.cpp/h<br/>siptrans.h<br/>--><!---->
      
      
=== IP-DECT: OEM configuration option ===
=== IP22 IP24 IP28 IP305: Sometimes the DSP stops after sending CLIP ===


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The configuration format of an OEM configuration option was changed for the backward compatibility.<br/><br/><!--<br/>Changed file:<br/>dectmaster.cpp<br/>--><!---->
..<br/><!-- ac_494e.cpp ac494004ce3_680.00.pf.01.h ac498004ce1_680.00.pf.01.h --><!---->
      
      
=== phone_orchid: micro mute when a waiting call was connected after an outbound call setup had  been cancelled ===
=== PBX: Configuration UI broken for checkmarks on PBX/Config/General and PBX/Config/myPBX ===


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- A goes offhook<br/>- B calls A, a waiting call from B is indicated on A<br/>- A goes onhook, phone rings<br/>- A goes offhook again and is connected to B<br/>- A hears B, B doesn't hear A <br/><br/><!-- ac_codedc3.cpp --><!---->
The checkmarks on these two pages where all cleared when the other page was edited.<br/>Collateral damage of<br/><br/>fix: #88352: PBX: Allow configuration of default presence/dialog-info visibility for group members<br/><!--<br/>config_options.cpp<br/>pbx_general.xsl<br/>--><!---->
      
      
=== SIP: Pass display information to application ===
=== phone: ip222,ip232: the destination of an acticve diversion was not saved when the diversion was deactivated at  phone ===


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|}
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Pass display information to application when registration comes up.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
The destination (name or number) disappered when the diversion was deactivated. It was also not visible via the WEB interface.<br/><br/><!-- app_gui.cpp --><!---->
      
      
=== IP0010 IP1060 IP3010 IP6010: During long reset the ready LED was blinking orange and the Ethernet LEDs were swaped ===
=== Gateway: Missed FAX pages received with the FAX interface ===


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|}
|}
Now the ready LED shows the green blinking during long reset.<br/>The Ethernet LEDs are initialized directly after reset to overwrite the default setting that swaps link and speed.<br/><br/><!-- <br/>ip6010.cpp platform_mv78x00.c ether_mv78x00.c boot_xxx.y<br/><br/>Note: the IPxx10 uses always the 3 pin LED<br/>The 2 pin LED of the IP6000 is not used any more. <br/>The 3 pin LED needs both GPIOs to be zero in off state, with the 2 pin LED both GPIOs can be zero or one in off state.<br/>--><br/><!---->
If the FAX interface is used to receive a FAX document with ECM mode and the transmitting terminal appends additional EOLs, the page counter is wrong and document pages are not written. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>t30.cpp<br/>facsimile_g3.h<br/>facsimile_g3.cpp<br/>facsimile_ECM.h<br/>facsimile_ECM.cpp<br/>--><!---->
   
= V9 Hotfix 19 (9061180) =
Changes included in Version 9 hotfix19
[http://mantis.innovaphone.com/view.php?id=89318 Definition]
 
== New Features ==
 
 
      
      
=== X.509: Creating certificate containing IPv4 address did not work ===
=== PBX-SOAP: Present normalized number of peer also ===


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|}
|}
When creating a certificate or a request with an IPv4 address, the IPv4 address was mapped to an IPv6 address. This was not the expected behaviour.<br/><br/>172.16.10.32 -> ::ffff:172.16.10.32<br/><br/>Now both IPv4 and IPv6 addresses can be used.<br/><!--<br/>x509.cpp<br/>--><!---->
The SOAP API presents the adjusted number of the peer (called/calling), which is the shortest possible number which can be dialed to call this. It is the same number as displayed on the phone. Sometimes an application needs to know the normalized number of the peer, which is the number in the context of the root node. This number is sent as additional number with the identifier "norm"<br/><!--<br/>pbx_xml.cpp<br/>pbx.cpp<br/>--><!---->
      
      
=== ip222,ip232: waiting calls mute after having been accepted/connected on USB headsets, mainly seen with Jabra LINK 14201-30 ===
=== IP232,IP222,IP241: Config option to adjust LCD brightness in idle state ===


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|}
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In some cases only the accepted call was mute and the next call was OK again but the Jabra LINK 14201-30 lost the USB connection in most cases.<br/>Delaying the HID-commands sent to the headset solves this problem.<br/><br/><!--<br/>keys_ip222.cpp<br/>--> <br/> <!---->
-> Main Menu -> Phone Setup -> LCD light (idle state)<br/>Can be tuned down to zero.<br/><br/><!--<br/>9.00\\phone_orchid\\box\\forms\\lcd_lib\\forms_color.cpp<br/>9.00\\phone_orchid\\box\\forms\\lcd_lib\\forms_color.h<br/>9.00\\phone_orchid\\box\\forms\\lcd_lib\\forms_color_mod.cpp<br/>9.00\\phone_orchid\\common\\interface\\forms.cpp<br/>9.00\\phone_orchid\\common\\interface\\forms.h<br/>9.00\\phone_orchid\\phone\\app\\app_form.cpp<br/>9.00\\phone_orchid\\phone\\app\\app_gui.cpp<br/>9.00\\phone_orchid\\phone\\app\\app_gui.h<br/>9.00\\phone_orchid\\phone\\phone_user_if.h<br/>9.00\\phone_orchid\\phone\\phonetxt.h<br/>9.00\\phone_orchid\\phone\\txt\\phonetxt-latin1.cpp<br/>9.00\\phone_orchid\\phone\\txt\\phonetxt-utf8.base<br/>9.00\\phone_orchid\\phone\\txt\\phonetxt-utf8.cpp<br/>9.00\\phone_orchid\\phone\\user\\phone_user.cpp<br/>9.00\\phone_orchid\\phone\\user\\phone_user.h<br/>9.00\\phone_orchid\\platform\\config.h<br/>--><!---->
      
      
=== PBX: Objects list filter for numbers did not work correctly anymore ===
=== H.323: Automatically connect signaling TCP if NAT router is detected ===


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only top level nodes were displayed<br/>Collateral damage of fix<br/><br/>79982: PBX: When editing a Node object it was changed to a PBX object<br/><!--<br/>pbx.cpp<br/>--><!---->
When regestering an endpoint from a private network to a PBX within the public network, the signaling TCP connection must be established and maintained by the endpoint. Otherwise calls to the endpoint are not possible.<br/><!--<br/>h323sig.cpp<br/>h323ras.cpp<br/>h323.h<br/>h323asn1.h<br/>--><!---->
      
      
=== SIP: Pass display information to application ===
=== PRI-QSIG: Interop config for channel numbering ===


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|}
The QSIG standard defines to use Channel numbers (1-30) instead of timeslot (1-15, 16-31) as it is defined for EDSS1. There are many 'old' QSIG implementations around, which do it wrong. The QSIG-ECMA1 protocol setting is used for these 'old' implementations and the QSIG-ECMA2 setting for standard conform inplementations.<br/><br/>With the QSIG-ECMA1 also 'old' facility coding is used. There is also the combination of standard facility coding and timeslots for channels around so an independent mechanism to configure the channel numbering is needed.<br/><!--<br/>q931.cpp<br/>q931.h<br/>isdn_interop.xsl<br/>--><!---->
=== Other new Features ===
 
{|
|-
|valign=top nowrap=true|[http://mantis.innovaphone.com/view.php?id=89367 89367]
|valign=top nowrap=true|jfr
|case independence for the characters of the Basic Russian Alphabet added
|}
|}
Pass display information received in REGISTER response to app.<br/><br/><!--<br/>sip.cpp/h<br/>--> <!---->
 
   
== Bug Fixes ==
 
 
      
      
=== IP241,IP222,IP232: Show H323-ID instead of Display Name if Display Name is not available ===
=== SIP/TLS: Rejecting server certificate ===


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Like on old b/w phones.<br/><br/><!--<br/>forms_color.cpp/h<br/>--><!---->
Validating server certificate against configured domain name.<br/>Must be validated against configured proxy domsina name.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== IP-DECT: Memory leak with special Cisco features ===
=== SIP: Wrong branch value in Via header in ACK request ===


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|}
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Only with special OEM features.<br/><!--<br/>dtmffty.cpp<br/>--><!---->
Branch value in Via header in ACK request must be new after 200 response.<br/>Branch value in Via header in ACK request must be same after non-200 response.<br/><br/><!--<br/>siptrans.cpp/h<br/>--><!---->
      
      
=== SIP: Problems with DNS resolving of proxy adresses ===
=== H.323: RTP-DTMF did not work on exclusive coder/media relay configurations ===


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|}
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Not always re-tried when failed at startup.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
Problem for DTMF on SIP trunks<br/><!--<br/>h323ch.cpp<br/>--><!---->
      
      
=== IP241,IP222,IP232: Rendering errors on Fkey configuration screen ===
=== AD Replication: A Buffer for Processing The Paged Result Cookie Was Too Small ===


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|}
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When scrolling or leaving Fkey configuration screen.<br/>List (Toggle) controls were not rendered correctly.<br/><br/><!--<br/>form_color.cpp<br/>--><!---->
1KB wasn't enough, now 2KB.<br/><!--<br/>interface/ldapapi.cpp/.h<br/>--><!---->
      
      
=== IP-DECT: Mobility Master registration ===
=== LDAP Expert: "Next"-Browsing through DB failed ===


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|}
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The Mobility Master does not accept clients with the OEM name IP1202. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dectmobmaster.cpp<br/>--><!---->
URI encoding error<br/><!--<br/>fdirui_right.xsl<br/>--><!---->
      
      
=== HTTP: Possible trap on many simultaneous sessions ===
=== phone: ip222, ip232,ip241:  the notification tone indicating a new message when a call is active was sent to remote ===


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|}
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Sorting the TCP sessions did not work correctly, which caused an assertion because it could happen that it was not possible to remove a session<br/><!--<br/>ip_user.cpp<br/>--><!---->
instead to notify the receiver of the message the remot party did hear the tone<br/><br/><!-- phonesig.cpp app_call.cpp --><!---->
      
      
=== IP-DECT: System GUI, disabled local coder options ===
=== SIP: "Supported: timer" missing in UPDATE message ===


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|}
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If the PARI function (only IP1202) of the IP-DECT Master is disable, configuration changes on the System GUI do not effect anything. The settings for the local coder are disabled on this GUI page now.<br/><br/><!--<br/>Change file:<br/>_inn_asc_dectusers.xsl<br/>--><!---->
"Supported: timer" missing in UPDATE message.<br/><br/><!--<br/>sip.cpp/h<br/>siptrans.cpp/h<br/>--><!---->
      
      
=== IP-DECT: System settings not to dynamically connected radios ===
=== phone: ip222,ip232: humming noise in USB headset speaker in outbound call setup phase, disappears once connected ===


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|}
|}
System settings should not be sent if the radio-master connection is a dynamic one (IP1202 only). This is fixed now.<br/><br/><!--<br/>Changed files:<br/>dectusers_if.h<br/>dectusers.h<br/>dectusers.cpp<br/>dectmaster.cpp<br/>--><!---->
sometimes a humming noise was heard in the USB headset speaker in the setup phase of an outbound call. it disappeared as soon as the call was connected. <br/><!-- ac_phone3.cpp --><!---->
      
      
=== Support for old versions of Jabra BIZ 2400 headset with 48 kHz output sampling frequency ===
=== phone:  an intrusion call set up via Partner function key could not be cleared at the intruding phone via TAPI ===


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|[http://mantis.innovaphone.com/view.php?id=89443 89443]
|}
|}
the newer versions come with the usual 16 kHz sampling frequency but have the same signature (vendor/product) as the older ones.<!---->
The TAPI interpreted the recording state as a conference<br/><br/><!-- app_call.cpp --><!---->
      
      
=== SoftwarePhone: Trap with trace and mute ringer event ===
=== log message forwarding to another innovaphone device did not work since V9hotfix15 (on the receiving device) ===


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A trap occurs if the trace is enabled and the ringer mute option is toggled.<br/><br/><!--<br/>Changed file:<br/>phone_V5/phone.h<br/>--><!---->
this problem is located on the reciving device, it does not depend on the the version of the forwarding device.<br/><br/><!-- logging.cpp logging.h --><!---->
      
      
=== IP222 IP232 IP241: Force same speed of the switch ports for 1000M/100M scenarios ===
=== IP22 IP24 IP28 IP305: DSP debug code added ===


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In scenarios that operate one Ethernet port with 1000M and the other with 100M the switch througput was low.<br/> Now the 1000M port is reconfigured to 100M, and the throughput is high.<br/><br/><!-- orchid_drv.* --><!---->
..<br/><!-- ac_494.cpp --><!---->
      
      
=== SIP: Support for multiple audio media descriptions ===
=== PBX Mobility: Conference ID not set for outgoing calls initiated by myPBX ===


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One without encryption and one with encryption:<br/><br/>v=0<br/>o=OpenStage-Line_0 968610650 446118927 IN IP4 10.30.1.144<br/>s=SIP Call<br/>c=IN IP4 10.30.1.144<br/>t=0 0<br/>m=audio 5012 RTP/AVP 9 8 0 18 101<br/>a=rtpmap:9 G722/8000<br/>a=rtpmap:8 PCMA/8000<br/>a=rtpmap:0 PCMU/8000<br/>a=rtpmap:18 G729/8000<br/>a=rtpmap:101 telephone-event/8000<br/>a=silenceSupp:off - - - -<br/>a=fmtp:18 annexb=no<br/>a=fmtp:101 0-15<br/>a=sendrecv<br/>m=audio 5010 RTP/SAVP 9 8 0 18 101<br/>a=rtpmap:9 G722/8000<br/>a=rtpmap:8 PCMA/8000<br/>a=rtpmap:0 PCMU/8000<br/>a=rtpmap:18 G729/8000<br/>a=rtpmap:101 telephone-event/8000<br/>a=silenceSupp:off - - - -<br/>a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ujVU8G6kgknZnPflRwx8tadNskkp9glas/DFCbC3<br/>a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:+uoFBNVVhDY5OOGvMOdAlvpxvc98hX/VeProhlwH<br/>a=fmtp:18 annexb=no<br/>a=fmtp:101 0-15<br/>a=sendrecv<br/><br/><br/><!--<br/>sip.cpp/h<br/>sdpmsg.cpp/h<br/>--><!---->
This could cause several problems:<br/>- When the call was sent to a local user with multiple registrations, the call to each registration had a different conferenceID, so myPBX could not match these calls to actually being only a single call, so multiple calls were dissplayed<br/>- The CDRs created for this call could not be matched<!---->
      
      
=== IP232: Show more of collapsed fkeys ===
=== SIP: Don't tell application that registration is down when handling redirect response ===


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140px when expanded.<br/>100px when collapsed (80px before).<br/><br/><!--<br/>forms_color_mod.cpp<br/>--><!---->
Don't tell application that registration is down when handling redirect response for REGISTER.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== SIP: Double Replaces header in INVITE after REFER ===
=== SIP: No T.38  parameter when indicating capabilitity only ===


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Reject by "Swyx IpPbxSrv/8.1.0.246" with "400 Bad Request(Multiple values in single-value header Replaces)"<br/><br/><!--<br/>sip.cpp<br/>--><!---->
No T.38  parameter when indicating capabilitity only.<br/><br/>Offer<br/>\tv=0<br/>\to=- 2 1 IN IP4 172.16.16.124<br/>\ts=-<br/>\tt=0 0<br/>\tm=audio 16386 RTP/SAVP 8 101 13<br/>\tc=IN IP4 172.16.16.124<br/>\ta=rtpmap:101 telephone-event/8000<br/>\ta=fmtp:101 0-15<br/>\ta=ptime:20<br/>\ta=silenceSupp:off - - - -<br/>\ta=sendrecv<br/>\ta=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:WJl714I+mpSr47ld1YjYMf8t9xQo0xYHUng1CnDi<br/>\tm=image 0 udptl t38<br/>\tc=IN IP4 172.16.16.124<br/><br/>Instead of<br/>\tv=0<br/>\to=- 2 1 IN IP4 172.16.16.124<br/>\ts=-<br/>\tt=0 0<br/>\tm=audio 16394 RTP/SAVP 8 101 13<br/>\tc=IN IP4 172.16.16.124<br/>\ta=rtpmap:101 telephone-event/8000<br/>\ta=fmtp:101 0-15<br/>\ta=ptime:20<br/>\ta=silenceSupp:off - - - -<br/>\ta=sendrecv<br/>\ta=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:sPngj1zoO9wI1pc1tMTymWCzTgwMoamPuJAFIhga<br/>\tm=image 0 udptl t38<br/>\tc=IN IP4 172.16.16.124<br/>\ta=T38FaxVersion:0<br/>\ta=T38MaxBitRate:14400<br/>\ta=T38FaxFillBitRemoval:0<br/>\ta=T38FaxTranscodingMMR:0<br/>\ta=T38FaxTranscodingJBIG:0<br/>\ta=T38FaxRateManagement:transferredTCF<br/>\ta=T38FaxUdpEC:t38UDPRedundancy<br/><br/><br/><!--<br/>sdpmsg.cpp<br/>--><!---->
      
      
=== PBX Mobility: No response for call to busy mobile phone, if no fixed phone ===
=== OS: Potential trap when starting a timer, which is already deleted ===


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If a user is called with mobility configured and no fixed phone and the mobile phone was busy, then the call did not complete and was hanging as if the number was not complete. The call should be answer with busy instead.<!---->
Could result in a strange trap with PBX transfer/recall<!---->
      
      
=== Chrome compatibility issue on PRI statistics page ===
=== phone:  while recording was running for a call this call ould not be cleared via SOAP/TAPI ===


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If clear was clicked an error message was displayed<!---->
The TAPI assumed the intruding phone to be in a conference<br/><br/><!-- app_call.cpp --><!---->
      
      
=== PBX Admin UI: Not possible to change DECT parameters for users with cf/grp admin rights ===
=== PBX Admin UI: Reconfiguring user visibility (Access) did not work correctly ===


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The user interface prohibited the editing of DECT parameters based on the rights of the object to be edited not based on the login<!---->
If a setting for a user/group was deleted, it could happen that some of the settings (Group, Online, Presence, Dialog, Ids) where copied to the next entry.<br/><!--<br/>pbx_admin.cpp<br/>--><!---->
      
      
=== phone: ip222,ip232: when a call is released from remote while another call is waiting the waiting call cannot be accepted. ===
=== Voicemail: Trap ===


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When the call is released from remote a buys tone is generated for two seconds. Therafter phone rings to indicate that the waiting call can be accepted now.<br/>When trying to accept this call by pressing the headset talk button the call was disconnected instead.<br/><br/><!-- keys_ip222.cpp --><!---->
If Trace-Checkmark activated, a diverted call carrying an mwi interrogate was processed<br/><!--<br/>pbx_vm.cpp<br/>--><!---->
      
      
=== SIP: Memory leak in SIP stack ===
=== Possible Trap While Getting Post-Mortem Log ===


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Memory leak in SIP stack.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
Possible Trap While Getting Post-Mortem Log<br/><!--<br/>httpclient.cpp<br/>signal.cpp<br/>--><!---->
      
      
= V9 Hotfix 14 (9061046) =
=== Media: Redirecting SRTP streams for NAT clients only after successful SRTP authentication ===
Changes included in Version 9 hotfix14
[http://mantis.innovaphone.com/view.php?id=82178 Definition]
 
== New Features ==
 
 
   
=== phone: support PBX-directory access via TLS ===


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A "Use TLS" checkmark has been added to the PBX directory config of the phone. If set, port 636 is used instead of port 389.<br/><br/><!-- phone_config.cpp, phone_config.h, phone_edit.cpp, reg_edit_dir.xsl, phone_dir.cpp --><!---->
Media endpoints support NAT.<br/>If receiving RTP/SRTP from an address other than negotiated one<br/>media endpoints redirecting their media stream towards source of incoming media stream.<br/>In case of SRTP, this NAT workaround is only executed if incoming media stream has passed authentication.<br/>For securitiy reasons.<br/><br/><!--<br/>media.cpp<br/>--><!---->
      
      
=== PBX-Mobility: Data Call Thru ===
=== AD Replication: Merge v10 code into v9 code ===


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To speed up dialing with the mobility client, the called number can be posted to the PBX via HTTP before the call<br/><!--<br/>pbx_admin.cpp<br/>pbx_api.h<br/>pbx.h<br/>pbx.cpp<br/>pbx_mobility.h<br/>pbx_mobility.cpp<br/>--><!---->
Some recent refactoring works weren't in sync. A state machine didn't reach "Completed".<br/><!--<br/>service/ldap/<br/>--><!---->
      
      
=== IP-DECT: DECT security (2) ===
=== phone: if a call ringing while the handset was offhook was accepted via SOAP/TAPI the call was not cleared when going onhook ===


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Some changes in preparation for the new feature "DECT security". This feature will be available later (IP1202).<br/><br/><!--<br/>New files:<br/>ascom/mobmaster/registration_context.h<br/>ascom/mobmaster/registration_context.cpp<br/>ascom/mobmaster/cryptomaster.h<br/>ascom/mobmaster/cryptomaster.cpp<br/>ascom/mobmaster/cryptomaster_registration.h<br/>ascom/mobmaster/cryptomaster_registration.cpp<br/>ascom/mobmaster/crypto_key_table.h<br/>ascom/mobmaster/crypto_key_table.cpp<br/>ascom/mobmaster/crypto_key_table_entry.h<br/>ascom/mobmaster/crypto_key_table_entry.cpp<br/>ascom/httpfiles/asc_dectcryptomaster.xsl<br/>ascom/httpfiles/asc_dectcryptomaster_mobmasters.xsl<br/><br/>Changed files:<br/>common/interface/signal.cpp<br/>dect_comcerto/platform/config.h<br/>ip1201/platform/config.h<br/><br/>dect/dectusers_if.h<br/>dect/users/dectusers.cpp<br/>dect/master/dectmaster.cpp<br/>dect/localusers/dectlocalusers.cpp<br/><br/>ascom/mobmaster/mobmaster.mak<br/>ascom/mobmaster/dectmobmaster.h<br/>ascom/mobmaster/dectmobmaster.cpp<br/>ascom/httpfiles/httpfiles.mak<br/>--><!---->
This could happen when the handset was lifted and kept lifted after the disconnect key was prressed. an inbound call arriving in this state could be accepted via SOAP/TAPI and was connected to the handset but the call could not be cleared by going onhook. Only the disconnect key did clear the call. <br/>Now the call is cleared as expected when going onhook.     <!---->
      
      
=== IP-DECT: No fall-back after unattended call transfer ===
=== phone_orchid: pressing speaker key in handset/headset mode switches to handsfree mode, pressing again returns to previous mode ===


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Now there is no fall-back after an unattended call transfer and the behavior is consistent with the other call transfer types (attended, semi-attended). To switch back to the hold call the R-key must be pressed.<br/><br/><!--<br/>Changed file:<br/>dectradio_call.cpp<br/>--><!---->
handset/headset plus speaker is not supported on phone_orchid, the previous solution where the connection was dropped when the speaker key was pressed again  (see #84297) was perceived as irritating.<br/><br/><!-- ac_codec3.cpp ac_codec3.h phonesig.cpp app_ctl.cpp --><br/><br/><!---->
      
      
=== IP-DECT: Idle display update ===
=== IP22 IP24 IP28 IP305: Sometimes the DSP stops after sending CLIP (2) ===


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Handling of idle display update message in the call release message is added. Used with an OEM PBX.<br/><br/><br/><!--<br/>Changed files:<br/>dectmaster.cpp<br/>dectfty.cpp<br/>--><!---->
..<br/><!-- ac_494e.cpp ac494004ce3_680.00.pf.02.h  --><!---->
      
      
=== phone: ip222,ip232: the variable KEYS0/HID-MAP permits to map new USB headsets to builtin descriptors ===
=== SIP: Bug in handling of INVITE with Replaces ===


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Sometimes USB headsets come with a signature different from the signature of similar headsets which are already supported.<br/>A "vars create KEYS0/HID-MAP p <map>" maps the new signature to an existing one.<br/><map> format is<br/> manufacturer:product=manufacturer:product<br/>the second manufacturer:product tuple is the signature of an already supported headset,<br/>'manufacturer' and 'product' are plain 4 digit hex numbers without a "0x" prefix.<br/><br/><!-- keys_ip222.cpp --><!---->
Bug in handling of INVITE with Replaces.<br/>Results in hanging call.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== IP2x2, IP241: Coder Preferences for prefered coder G.722 suboptimal ===
=== IP-DECT: Trap with call transfer ===


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In case G.7222 was selected as prefered coder and the called endpoint did not support G.722, as next best coder G.729 was selected. This is typically not what is desired in such a case, G.711 is the better alternative in this case<br/><!--<br/>ac_dsp3.cpp<br/>ac_phone3.cpp<br/>--><!---->
A trap occurs if a call transfer is received in the IP-DECT radio module. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dectradio.cpp<br/>--><!---->
      
      
=== myPBX: Show version of launcher in the list of sessions ===
=== Fix for MIPS counter ===


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The version of the used launcher is now shown on the page PBX/myPBX.<br/><!--<br/>ctiinterface.cs<br/>pbx_client_sessions.xsl<br/>pbx_client.h<br/>pbx_client.cpp<br/>client.js<br/>client_util.js<br/>--><!---->
MIPS counter was incorrect on IP1201 and IP4001<br/><br/><!--<br/>ip1201.cpp<br/>ip4001.cpp<br/>--><!---->
''Status:''
checked-in: 10.00<br/>checked-in: 9.00
      
      
=== H.323: Support for Avaya SRTP with AES128/80 ===
=== SIP: Media negotiation fails on calls into Waiting Queue ===


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Needed for OEM<br/><!--<br/>h323ch.cpp<br/>h323lib.cpp<br/>h323lib.h<br/>--><!---->
Media negotiation fails on calls into Waiting Queue if<br/>caller put call on hold before WQ agent accepts the call.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== SoftwarePhone: HID Support for new headsets ===
=== 'power-off loop' relay switching function failure ===


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With this new version the following headsets are supported for call control:<br/>- Jabra GO 6430 (Jabra LINK 350 USB with firmware 5.4.17 or later) with product id 0xa342. Please select the first device.<br/>- Jabra SUPREME UC (Jabra LINK 360 USB) with product id 0xa346. Please select the first device.<br/>- Jabra PRO 9470 with product id 0x1042.<br/>- Sennheiser VoIP USB headset (SH 350 IP) with product id 0x0008.<br/>- Sennheiser DW Office with product id 0x740a. Please select the first device.<br/>- Sennheiser CEHS-CI 02 (USB adapter cable) with product id 0x0030. Please select the second device.<br/><br/><!--<br/>Changed files:<br/>hid.tbl<br/>sphone_keys.cpp<br/>--><!---->
Especially POE-switches with higher supply voltages than 48V lead to a decreased timespan of powering the build-in relays of a ip6010/ip810 gateway. The detection of a power-fail condition is therefore derived from the POE ICs which react earlier and thus increases powering time of the relays.<!---->
      
      
=== PBX: Description was missing for DECT System object ===
=== myPBX: Default group visibility was not displayed correctly ===


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A configurable description is useful for the DECT System object as for all other objects<br/><!--<br/>pbx_edit_dect.xsl<br/>--><!---->
The default group visibility can be configured from v9hotfix17. Regardless of that configuration myPBX showed full visibility in the visibilty settings.<br/><!--<br/>pbx_api.h<br/>pbx.h<br/>pbx_client.cpp<br/>--><!---->
      
      
=== Permit logging to a second (shadow) log server ===
=== IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: DSP Update to Version 680.07 ===


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"Services/Logging/Log Server/Log Server Shadow/Address" defines the adress of a second server. <br/>"Services/Logging/Log Server/Log Server Shadow/Enable" starts/stops logging to the second server.<br/>Except the address the configuration for the second server is copied from the first server.<br/><br/><!-- logging.cpp, logging.h, logging.xsl platform/config.h --><br/><!---->
Fixed modem bypass with slow modems<br/><br/><!-- <br/>DSP code details at <br/>\\\\innovaphone.sifi\\dfs\\Dokumente\\Hardware\\G7xx\\Audiocodes\\firmware\\firmware_ac49x\\680.07<br/><br/>ac49x_680/ac49*.h  ac_494e.cpp ac_491.cpp ac_dsp3.cpp<br/><br/>Also AC494 DSP clock changed from 100 to 125MHz and report interval changed to 5seconds. <br/>--><!---->
      
      
== Bug Fixes ==
= V9 Hotfix 20 (9061198) =
Changes included in Version 9 hotfix20
[http://mantis.innovaphone.com/view.php?id=90101 Definition]
 
== New Features ==




      
      
=== Incorrect disk usage calculation for more than 4GB ===
=== Linux: Memory allocation changed for IP810 ===


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The disk usage calculation was wrong, if more than ~4 GB of the card were used.<br/><br/><!--<br/>fat32.cpp<br/>fat32.h<br/>fat32.xsl<br/>--><!---->
The memory allocation for the IP810 is changed to 128MB/384MB for innovaphone/Linux.<br/><br/>Important:<br/>Linux [[Support:Linux_Application_Platform_Hotfix_11_and_before_won't_run_on_Firmware_V9_hotfix_20_and_later_on_IP810|must be used with the kernel version 3.4.10 or later]]. This kernel is included in the Linux Application Platform V9.00 hotfix12 and later. The kernel is automatically updated with the Linux Application Platform V9.00 hotfix12.<br/><br/><!--<br/>Changed files:<br/>ip6010/platform/config.h<br/>ip6010/platform/startup.S<br/>ip6010/platform/ip6010.cpp<br/>--><!---->
      
      
=== PBX Waiting: Call state shows "Disconnecting" after switch from announcement 1 to announcement 2 ===
=== phone: new "Do Not Disturb" action "ring once" ===


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Announcement disconnect changed state<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
If "Phone/User-x/Preferences/Do Not Disturb/Action: ring once" is selected a new inbound call is indicated with a short tone only.<br/>Both the the tone and the duration of the tone can be configured under ""Phone/User-x/Preferences/Ring Tones/Do Not Disturb".<br/>If not configured the default ring tone is played for one and a half second. <br/><br/><!-- phone_config.cpp phone_edit.cpp reg_edit_general.xsl phone_config.h phone-strings-latin1.h phone-strings-utf8.h phone-text-tables.h app_ctl.h app_call.cpp app_gui.cpp phonetxt.h phonetxt-utf8.cpp phonetxt-latin1.cpp phonetxt-utf8.base<br/> --><!---->
      
      
=== PBX: Switch from Music on Hold to inband ringback on Alert after unpark oder transfer ===
=== PBX: Support of long user-user-informations by SOAP ===


{|
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|[http://mantis.innovaphone.com/view.php?id=81407 81407]
|[http://mantis.innovaphone.com/view.php?id=90029 90029]
|}
|}
To give the caller feedback that soon somebody may answer the call<br/><!--<br/>pbx.cpp<br/>--><!---->
Support of long user-user-informations (UUI) for SOAP sessions added. A long UUI is split into multiple short UUIs supported by Q.931.<br/>It is required by the FAX interface.<br/><br/><!--<br/>Changed files:<br/>pbx_xml.h<br/>pbx_xml.cpp<br/>--><!---->
      
      
=== PBX: Don't forward in-band info indicator if no media channel ===
=== Relay: Support of long user-user-informations by FAX ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=81879 81879]
|[http://mantis.innovaphone.com/view.php?id=90030 90030]
|}
|}
Not good to indicate in-band info available if no media channel can be negotiated<br/><!--<br/>pbx.cpp<br/>--><!---->
Support of long user-user-informations (UUI) for the FAX interface added. A long UUI is split into multiple short UUIs supported by Q.931.<br/><br/><!--<br/>Changed files:<br/>webfax.h<br/>webfax.cpp<br/>--><!---->
      
      
=== SIP: Trap when configuring user presence ===
=== Gateway: Support of a header line for FAX documents ===


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|[http://mantis.innovaphone.com/view.php?id=81996 81996]
|[http://mantis.innovaphone.com/view.php?id=90374 90374]
|}
|}
Trap may occur when configuring user presence.<br/><br/><!--<br/>sip_presence.cpp<br/>--><!---->
Support of a header line for FAX documents is added in the FAX interface.<br/><br/><!--<br/>Changed files:<br/>webfax.h<br/>webfax.cpp<br/>t30.h<br/>t30.cpp<br/>t4.h<br/>t4.cpp<br/>facsimile_g3.h<br/>facsimile_g3.cpp<br/>--><br/><!---->
      
      
=== HTTP-Server: Configuration of "Public compact flash access" did not work for all cases ===
=== SIP: Support for MESSAGE inside voice call ===


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|[http://mantis.innovaphone.com/view.php?id=82064 82064]
|[http://mantis.innovaphone.com/view.php?id=90408 90408]
|}
|}
E.g. /DRIVE/CF0/Neuer Ordner/ does not work, because HTTP request contains escaped sequences.<br/><br/><!--<br/>http.cpp<br/>--><!---->
Support for text messages inside voice call.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== phone: ip222,232.241: no notification tone on a successfull redial attempt ===
=== PBX-SOAP: Support for 'rc' and 'srce164' on UserCall on Waiting Queue ===


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|[http://mantis.innovaphone.com/view.php?id=82166 82166]
|[http://mantis.innovaphone.com/view.php?id=90537 90537]
|}
|}
On a failing or unanswered call the menu key opens the "Recall" menu. If "Redial" is selected the call is automatically redialed for 20 minutes in intervals depending on the result of the previous attempt. On success the user should be notified about the connection.<br/><br/><!-- ac_codec3.cpp --><br/><!---->
If a Waiting Queue is used for outgoing calls, these features can be usefull for some applications<br/><!--<br/>pbx_api.h<br/>pbx_xml.cpp<br/>pbx_wait.cpp<br/>pbx_wait.h<br/>--><!---->
      
      
=== IP-DECT: Trap during subscribing handsets ===
=== PBX: New configuration option 'Hide connected Number' at object ===


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|[http://mantis.innovaphone.com/view.php?id=82190 82190]
|[http://mantis.innovaphone.com/view.php?id=90693 90693]
|}
|}
A trap occurs during subscribing handsets on the IP1200. This is a fix for the previous fix #80424 and fixed again.<br/><br/><!--<br/>Changed file:<br/>dect.cpp<br/>--><!---->
In some cases it is desireable not to reveal the final destination of a call to a caller. For example a call center agent should not be called directly by the customer.<br/><!--<br/>pbx.cpp<br/>pbx_api.h<br/>pbx_admin.cpp<br/>pbx_edit_object.xsl<br/>pbx_wait.cpp<br/>--><!---->
      
      
=== IP-DECT: Wrong release code ===
=== IP-DECT: Configuration option 'Registration with system password' ===


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|[http://mantis.innovaphone.com/view.php?id=82195 82195]
|[http://mantis.innovaphone.com/view.php?id=91460 91460]
|}
|}
The IP-DECT sends the wrong release code "User not reachable" if the handset was not connected and the remote parts disconnects. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dectradio_dectcall.cpp<br/>--><!---->
The configuration option 'Registration with system password' is added. If ticked, all users are registered with the system password. This is useful, if the PBX users are only allowed to register with the PBX password.<br/><br/><!--<br/>Changed files:<br/>dectmaster.h<br/>dectmaster.cpp<br/>dectmaster.xsl<br/>_inn_asc_dectmaster.xsl<br/>--><!---->
      
      
=== IP-DECT: No delay with semi-attended call transfer and SIP ===
=== Voicemail: URL-En-/Decoding ===


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|[http://mantis.innovaphone.com/view.php?id=82197 82197]
|[http://mantis.innovaphone.com/view.php?id=91609 91609]
|}
|}
If the SIP protocol is used and the user do a semi-attended call transfer, the call transfer is directly confirmed again. The semi-attended call transfer is stored in the base station and executed as an attended call transfer if the target party connects.<br/><br/><!--<br/>Changed files:<br/>dectradio.h<br/>dectradio.cpp<br/>dectradio_call.h<br/>dectradio_call.cpp<br/>dectradio_dectcall.h<br/>dectradio_dectcall.cpp<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp <br/>--><!---->
new statement allows to URL-encode or URL-decode a string<br/><lib-enc string=".." string_out="$var" type="url"/><br/><lib-dec string=".." string_out="$var" type="url"/><br/><!--<br/>vm_script_if.cpp<br/>--><br/><!---->
      
      
=== IP-DECT: Display info with failed user log-in ===
== Bug Fixes ==
 
 
   
=== SIP: Auto answer with SDP in ACK ===


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|[http://mantis.innovaphone.com/view.php?id=82200 82200]
|[http://mantis.innovaphone.com/view.php?id=89539 89539]
|}
|}
If the user log-in fails, the cause is shown in the display.<br/><br/><!--<br/>Changed file:<br/>dectradio_call.cpp<br/>--><!---->
Auto answer results into no-audio when INVITE comes w/o SDP offer.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== IP-DECT: Short tone info by PBX only one time ===
=== SIP: Fix for overlap dialing with KPML ===


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|[http://mantis.innovaphone.com/view.php?id=82204 82204]
|[http://mantis.innovaphone.com/view.php?id=89581 89581]
|}
|}
Tone information with defined length and requested by the gatekeeper is played only one time to the handset. This is used by an OEM PBX and fixed now.<br/><br/><!--<br/>Changed file:<br/>dectradio_call.cpp<br/>--><!---->
Dialing digits entered before KPML subscription is established<br/>need to be queued until KPML subscription is established.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== IP-DECT: Hanging aborted semi-attended call transfers with SIP ===
=== SIP: Re-try INVITE after 407 even if no password configured ===


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|[http://mantis.innovaphone.com/view.php?id=82205 82205]
|[http://mantis.innovaphone.com/view.php?id=90024 90024]
|}
|}
If the SIP protocol is used, a semi-attended call transfer is done by the user and the call transfer can not be executed, the remaining call party is not disconnected. This is added now.<br/><br/><!--<br/>Changed files:<br/>dectradio.h<br/>dectradio.cpp<br/>dectradio_call.h<br/>dectradio_call.cpp<br/>dectradio_dectcall.h<br/>dectradio_dectcall.cpp<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp<br/>--><!---->
Calculate Digest with zero-length password and re-try INVITE.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== IP-DECT: LDAP server GUI description ===
=== IP232,IP222,IP241: Truncate directory entry information ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=82258 82258]
|[http://mantis.innovaphone.com/view.php?id=90027 90027]
|}
|}
The LDAP server GUI description is wrong and corrected now. (The LDAP server can not be a HTTP server.)<br/><br/><!--<br/>Changed files:<br/>dectmaster.xsl<br/>asc_dectmaster.xsl<br/>_inn_asc_dectmaster.xsl<br/>phonebook_ldap.xsl<br/>--><!---->
Truncate directory entry information to keep from overlapping with number type indication.<br/><br/><!--<br/>forms_color.cpp<br/>--><!---->
      
      
=== DHCP: Name registration at WINS was not refreshed as requested by TTL in registration response ===
=== IP-DECT: Don't show DTMF in radio call list ===


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|[http://mantis.innovaphone.com/view.php?id=82289 82289]
|[http://mantis.innovaphone.com/view.php?id=90034 90034]
|}
|}
If the DHCP-client gets a lease containing a WINS-server address and a NETBIOS node type P or M (1 or 2) the client tries to register it's NETBIOS-name (ipxxx-xx-xx-xx) with the WINS-server. The TTL returned by the server in the registration response determines when a name refresh has to be sent. <br/><br/><!-- dhcp.cpp, dhcp.h --> <!---->
User dialled digits during calls which are sent as DTMF should not be shown in the radio call list. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>dect_comcerto/platform/config.h<br/>ip1201/platform/config.h<br/>ip4001/platform/config.h<br/>dectradio.h<br/>dectradio.cpp<br/>dectradio_mastercall.cpp<br/>--><!---->
      
      
=== Phones: Presence info during ringing state may show garbage data ===
=== H.323: Allow media offers with 0.0.0.0 as address ===


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|[http://mantis.innovaphone.com/view.php?id=82306 82306]
|[http://mantis.innovaphone.com/view.php?id=90053 90053]
|}
|}
In case a presence update arrives at the phone while phone is in ringback state.<br/>Have been observed in conjunction with call forking with mobility only.<br/><br/><!--<br/>app_ctl.h<br/>app_call.cpp<br/>app_disp.cpp<br/>--><br/><!---->
Needed for SIP interoperability. Some third party SIP PBXs use addresses of 0.0.0.0 to indicate that they don't receive media. This may happens if an endpoint is put on hold. We did not forward such an offer and thus no Music on Hold was heard.<br/><!--<br/>h323ch.cpp<br/>--><!---->
      
      
=== myPBX: Support contact names containing a single quote ===
=== PBX: Busy On ... Calls at PBX objects did not take into acccount that a call may be routed back to Slave ===


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|[http://mantis.innovaphone.com/view.php?id=82323 82323]
|[http://mantis.innovaphone.com/view.php?id=90054 90054]
|}
|}
Configuring contacts with a H.323 id containing a single quote (') caused script errors in the web application.<br/><br/><!--<br/>client.js<br/>client_allows.js<br/>client_dirsearch.js<br/>client_util.js<br/>--><!---->
The busy on ... calls on PBX objects can be used to limit bandwidth usage between a master and a slave to a certain number of calls. Some calls are sent from a slave to the master and back to the slave if the routing decision cannot be done on the slave alone. This happens if escapes are used which overlap other obects (e.g. the local trunk). It is a common configuration the the E.164 routing scheme.<br/>With this fix, these calls are not counted for this purpose.<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_api.h<br/>pbx_wait.cpp<br/>pbx_wait.h<br/>pbx_trunk.h<br/>pbx_gw.h<br/>pbx_exec.h<br/>pbx_bc_conf.cpp<br/>pbx_bc_conf.h<br/>--><!---->
      
      
=== PBX: RTP-DTMF was disabled by Voice Mail object during re-negotiation ===
=== Voicemail: Send silence RTP during recording ===


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|[http://mantis.innovaphone.com/view.php?id=90095 90095]
|}
|}
RTP-DTMF acc. to RFC-2833 was disabled by Voice Mail object during re-negotiation.<br/><br/><!--<br/>pbx.cpp<br/>--><!---->
Send silence RTP during recording for some SIP carriers that do not send RTP without receiving RTP.<br/><br/><!--<br/>webmedia.cpp<br/>--><!---->
      
      
=== Gateway CDR with '0. 0' charge amount ===
=== IP2000: Prevent blinking error LED  IP2000 ===


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|[http://mantis.innovaphone.com/view.php?id=82359 82359]
|[http://mantis.innovaphone.com/view.php?id=90102 90102]
|}
|}
Should be '0.00' instead<br/><!--<br/>fty.cpp<br/>--><!---->
The firmware tried to load  the conference DSP, which is not available on the IP2000<br/><br/><!-- config.h --><br/><!---->
      
      
=== phone: call completion did not recover when DND(busy) was set at the phone requesting the call completion ===
=== SIP: Memory leak when receiving NOTIFY(message-summary) ===


{|
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|[http://mantis.innovaphone.com/view.php?id=82390 82390]
|[http://mantis.innovaphone.com/view.php?id=90106 90106]
|}
|}
when DND(busy) was set on the phone reqesting the call completion and was cleared some time later a "Recall possible" was not indicated anymore although a pending call completion was indicated on the called phone. <br/><br/><!-- phonesig_if.h, phonesig.cpp, phonesig.h, app_cc.cpp --><br/><!---->
Memory leak when receiving NOTIFY(message-summary)<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== ipv6: memory leak when sending fragmented packets ===
=== H.323: Support for registration from a private network thru NAT ===


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|[http://mantis.innovaphone.com/view.php?id=82394 82394]
|[http://mantis.innovaphone.com/view.php?id=90306 90306]
|}
|}
<!-- ip6_main.cpp, ip6_route.cpp --><!---->
In case an endpoint registers to a PBX from within a private network thru a NAT router, the signaling TCP connection must be maintained in order to be able to receive calls. When the registration is up a dummy call is sent to the PBX to establish the signaling TCP. This TCP connection is maintained after the dummy call is cleared. If this TCP connection is lost (e.g. NAT Router reset), the Registration is cleared and restarted, so that after the re-registration another dummy call is sent.<br/><br/>This is a fix for the previous fix<br/><br/>fix: #89497: H.323: Automatically connect signaling TCP if NAT router is detected<br/><br/>which did not work well<br/><!--<br/>h323sig.cpp<br/>h323.h<br/>voip.h<br/>gk.cpp<br/>relay.cpp<br/>relay_api.h<br/>relay.h<br/>--><!---->
      
      
=== H.323:No Media for calls with reverse media to a H.323/SIP exclusive Code Media Relay interface ===
=== SIP: SUBSCRIBE using old IP address in Contact field ===


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|[http://mantis.innovaphone.com/view.php?id=82408 82408]
|[http://mantis.innovaphone.com/view.php?id=90320 90320]
|}
|}
The execlusive coder/media relay config is used to avoid media negotiation problems with carrier which do not support media renegotiations. In case of a call with reverse media to such an interface, this did not work. This happens for example if a CFNR is configured at a Waiting Queue which redirects a call, which received an announcement from the Queue to such interface.<br/><!--<br/>h323ch.cpp<br/>--><!---->
If the IP address is changed at DHCP renew (or network change) the endpoint will immediately do a re-register to update the SIP Proxy with the new IP address.<br/>All SIP messages but SUBSCRIBE uses the new IP address in the Contact field.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== Kerberos: Allow editing multiple fields in admin UI ===
=== Voicemail: Memory Load High With Repeated Calls To &lt;store-getnext&gt; ===


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|[http://mantis.innovaphone.com/view.php?id=82425 82425]
|[http://mantis.innovaphone.com/view.php?id=90362 90362]
|}
|}
This is a fix for the page General/Kerberos in the admin UI.<br/><br/>Editing more that one item at a time could result in strange effects. Now it is possible to edit multiple fields.<br/><br/><!--<br/>kerberos_db.cpp<br/>kerberos_db.h<br/>kerberos_ldap.cpp<br/>kerberos_ldap.h<br/>--><!---->
The use-case does not happen within the standard voicemail, but within e.g. innovaphone conferencing.<br/><!--<br/>vm_script_if.cpp<br/>--><!---->
      
      
=== phone_inca: when a call completion was set up with CLIR active the called party was not displayed on a possible recall ===
=== PBX: CC Requests were sent with wrong number if a SendNumber was configured at the user ===


{|
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|[http://mantis.innovaphone.com/view.php?id=82435 82435]
|[http://mantis.innovaphone.com/view.php?id=90432 90432]
|}
|}
A sets CLIR, A calls B, B is busy<br/>A sets a CCBS request via Menu/Recall<br/>B goes on Hook<br/>A rings and sees 'anonymous' instad of the number of 'B', status line is empty (should show "Recall possible")<br/><br/><!-- forms_gen.cpp --><br/><!---->
Usually this is no problem only when interworking with some QSIG PBX's this causes the call-completion to fail.<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== IP-DECT: Display info with remote control call ===
=== phone_orchid: dialtone missing when recording is  active and the active call is held to open a consultation call ===


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|[http://mantis.innovaphone.com/view.php?id=90433 90433]
|}
|}
The display info shown after accepting a remote control call is fixed now.<br/><br/><!--<br/>Changed file:<br/>dectradio_call.cpp<br/>--><!---->
the consultation call could be established but there was no dialtone after pressing the R-key and no ringback tone after the number had been entered.<br/><br/><!-- ac_phone3.cpp ac_phone3.h codec.cpp codec.h phonesig_if.h phonesig.cpp phonesig.h app_call.cpp --><!---->
      
      
=== IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: enable modembypass ===
=== Possible trap when doing a leak check ===


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|[http://mantis.innovaphone.com/view.php?id=90451 90451]
|}
|}
Modembypass is enabled on all calls with disabled T.38 and coders G711A or G711U. <br/>Switch to modem bypass is indicated in the trace by "switch to modembypass".<br/>The feature can be disabled with http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl <br/><br/>Modembypass works best if T38 is disabled on both sides.<br/>If T38 is enabled on the called side the CED may trigger a T38 session, this changed back to voice and modem bypass is enabled (if G711 is active). The first modem tone is interupted, but we still have modembypass on both sides.<br/>If T38 is enabled on the calling side the calling side stays on regular G711.<br/><br/><br/><br/><!-- ac_dsp3.cpp ac_dsp3.h dsp.xsl --><br/><!---->
When many leaks exist or leak check is done when much tracing is turned on. The leak check itself could cause a watchdog trap, because the collecting of the leaks is done on highest priority so not even the timer interrupt could trigger the watchdog.<br/><!--<br/>os.cpp<br/>--><!---->
      
      
=== IP800 IP6000: reduce probability of false DTMF detection ===
=== IP-DECT: Busy state on maximum call count ===


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|[http://mantis.innovaphone.com/view.php?id=82471 82471]
|[http://mantis.innovaphone.com/view.php?id=90461 90461]
|}
|}
Change DTMF detector signal to noise ratio from 12db to 18db.<br/><!-- ac_dsp2.cpp --><br/><br/><!---->
The base station does not go to the busy state if the maximum call count is reached and the last call is an incoming call. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>dectradio.cpp<br/>dectradio_dectcall.cpp<br/>--><!---->
      
      
=== Gateway: 'Enable PCM' option added at the CONF interface of the IP800 ===
=== PBX: Support for Opticaller Data Callthru did not work ===


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|[http://mantis.innovaphone.com/view.php?id=82481 82481]
|[http://mantis.innovaphone.com/view.php?id=90480 90480]
|}
|}
The option 'Enable PCM' is available at the CONF interface of the IP800 now.<br/><br/><!--<br/>Changed file:<br/>config.h<br/>--><!---->
Support for Opticaller data callthru was added, but did not work<br/><!--<br/>pbx_admin.cpp<br/>--><!---->
      
      
=== Gateway: PCM mode of the CONF interface fixed ===
=== 802.1x: Version 3 Frames Weren't Processed ===


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|[http://mantis.innovaphone.com/view.php?id=82482 82482]
|[http://mantis.innovaphone.com/view.php?id=90485 90485]
|}
|}
The PCM mode of the CONF interface is not activated for a call even though it is configured and possible. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>conference_sig.cpp<br/>--><!---->
Now they're read initially. Handshakes remains to be in version 1.<br/><!--<br/>8021x.cpp<br/>--><!---->
      
      
=== IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: T38 trace flag added ===
=== PBX: Forking a call to a Trunk with "Outgoing call restricted", causes the original call to be restricted as well ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=82484 82484]
|[http://mantis.innovaphone.com/view.php?id=90487 90487]
|}
|}
Allows to trace T38 connection on the PCM port and on the DSP host interface.<br/>Use this if fax modem problems are suspected.<br/>Enable at http://addr/debug.xml at trace->T38 trace.<br/><br/><!-- ac_dsp3.cpp ac_dsp.h ac_491.cpp ac_494.cpp ip6010/ip24: trace.xsl --><!---->
The "Outgoing call restricted" flag on the trunk object to which the call was forked caused the call as a whole to be marked as Calling Line Presentation Restricted.<br/><!--<br/>pbx_trunk.cpp<br/>--><!---->
      
      
=== Gateway: Routing problem with blockdial route and following matching non-blockdial incomplete routes ===
=== SIP: Locally configured DNS entries were not used if no DNS server configured ===


{|
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|[http://mantis.innovaphone.com/view.php?id=82486 82486]
|[http://mantis.innovaphone.com/view.php?id=90508 90508]
|}
|}
Example:<br/><br/>Blockdial Route 00-><br/>After this a non-blockdial route with 0...<br/><br/>If now a number of 001 was dialed, the first route should match and after the enbock dialout the call should be sent to the destination of the route. Instead the call was rejected with "no destination found"<!---->
If no DNS server was configured, but DNS names are to be resolved,<br/>local DNS entries can be added (Services/DNS/Hosts).<br/>SIP stack fails with SRV query and does not try A query which would deliver IP address.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== failure of analog ports of ip28 ===
=== IP-DECT: RTP stream ===


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|[http://mantis.innovaphone.com/view.php?id=90539 90539]
|}
|}
ip28 analogue ports do not react to incoming calls and hook-off. Problem could only be solved by reset.<!---->
If a remote hold event is received, no RTP data should be sent by the IP-DECT device.<br/>A CTI initiated call is established with a call transfer and a "No Media data received" error event can occur.<br/>This is fixed now.<br/><br/><!--<br/>Changed files:<br/>h323ch.cpp<br/>dectradio_call.cpp<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp<br/>--><!---->
      
      
=== Debug "HTTP_GET LOG_HTTP.1: retry, authentication failed" removed ===
=== SIP: Trap on IP-DECT ===


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|[http://mantis.innovaphone.com/view.php?id=90569 90569]
|}
|}
<!-- httpclient_i.cpp --><!---->
Trap in GK-CHANNEL when Dectmaster application sends DTMF before call if created.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== Phone: Display text received with BYE ===
=== SIP: Disabled IP-DECT interface tries to register ===


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|[http://mantis.innovaphone.com/view.php?id=90597 90597]
|}
|}
Pass display text to phoneapp.<br/><br/><!--<br/>phonesig.cpp<br/>--><!---->
Disabled IP-DECT interface tries to register.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== SIP: Trap during call handling ===
=== H.323: Timer to monitor response to setup too short for some traffic cases ===


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|[http://mantis.innovaphone.com/view.php?id=82544 82544]
|[http://mantis.innovaphone.com/view.php?id=90696 90696]
|}
|}
Trap during call handling<br/><br/><!--<br/>sip.cpp<br/>--><!---->
This created unnecessary event in IP-DECT systems when calling powered off or out of range handsets<br/><!--<br/>h323sig.cpp<br/>--><!---->
      
      
=== IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: update to DSP code 660.27 ===
=== PBX Waiting: Potential Trap if editing while a call is initiated with SOAP ===


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|[http://mantis.innovaphone.com/view.php?id=90766 90766]
|}
|}
660.27 is the released version on the audiocodes web.<br/>Fax relay and modem bypass improved.<br/><br/><!-- <br/>Laut AC49x Voice over Packet Processor (VoPP) Release Notes Ver 6.6 Rev3 bei den Gateways keine wesentliche Änderung zu 660.19, also nur ein update um auf Stand zu sein.<br/><br/>ac_491.cpp ac_494.cpp ip24.mak ip6010.mak ac_firmware.h --><!---->
The Waiting object can be used as outgoing dialing object with SOAP. If this is done and the configuration is changed while an outgoing call was pending, a trap could happen<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
      
      
=== IP222 IP232 IP241: update to DSP code 660.27 ===
=== H.323: Potential Trap in special case which could only happen in version 10 ===


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660.27 is the released version on the audiocodes web.<br/>Improved acoustic echocanceller.<br/><br/><!-- <br/><br/>ac_494e.cpp ac_codec.cpp phone_orchid.mak  --><!---->
This fixed is merged to version 9 only of consistency reasons<br/><!--<br/>h323_ras.cpp<br/>--><!---->
      
      
=== phone: call forwarding options offered even when call forwarding was not  possible ===
=== SIP: Respect changes in PAI/PPI header when receiving UPDATE with SDP offer ===


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|[http://mantis.innovaphone.com/view.php?id=90778 90778]
|}
|}
Call forwarding is not supported when running SIP. But when the menu key was pressed after entering a number call forwarding options were offered (happened with the primary registration only). <br/><br/><!-- phone_user.cpp --> <!---->
PAI/PPI was processed when receiving UPDATE without SDP offer.<br/>PAI/PPI was ignored when receiving UPDATE with SDP offer.<br/>Now PAI/PPI is processed when receiving UPDATE with SDP offer.<br/><br/><!--<br/>sip.cpp<br/>--><br/><!---->
      
      
=== PBX Mobility: Trap in case of Transfer of a call from a mobile endpoint to another mobile endpoint ===
=== SIP: Possible buffer overrun ===


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|}
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The Trap happens in the following call scenarion<br/>- Mobile endpoint calls in, using mobility two-stage dialing<br/>- call is accepted at local phone<br/>- on local phone a consultation call is initiated to another user with mobility<br/>- when mobile phone rings, a transfer is initiated on local phone<br/>- the called mobile phone accepts the call<br/>- the trap happens when the called mobile phone hangs up<br/><br/>There could be other call scenarions where the trap happens as well<!---->
Fix for possible buffer overrun.<br/><br/><!--<br/>siputil.cpp/h<br/>--><!---->
      
      
=== SIP: SRTP key exchange failed ===
=== SIP: Wrong error log "Timeout during media negotiation for call" ===


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|[http://mantis.innovaphone.com/view.php?id=90821 90821]
|}
|}
Bug in base64 decoding of SRTP key.<br/><br/><!--<br/>sdp.cpp<br/>--><!---->
Error log "Timeout during media negotiation for call" may occur after re-negotioation.<br/>Re-negotioation occurs during hold/retrieve/transfer.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== PBX Trunk: Number to Name Feature did not work for calls to busy endpoints ===
=== PBX Trunk: Name to Number Feature did not work with calls to extern ===


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|}
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Such a call was not rejected with cause 'User Busy', but was just hanging.<!---->
If an endpoint cannot be found by name, the call should be forwarded to extern<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_api.h<br/>pbx_trunk.cpp<br/>pbx_trunk.h<br/>--><!---->
      
      
=== PBX Mobility: Call to mobile phone was sent with invalid diverting information ===
=== PBX CSV Import: Corrupted objects at buffer boundaries ===


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|[http://mantis.innovaphone.com/view.php?id=90942 90942]
|}
|}
A call to a mobile phone is sent with a diverting leg2 info, which means, the call contains the information, that it was diverted by the called user to the mobile phone. So in theory this could be displayed on a mobile phone.<br/><br/>The coding of this information was wrong and created interop problems with some networks.<br/><!--<br/>pbx_mobility.cpp<br/>fty.h<br/>--><!---->
The upload is processed in chunks of 2K. At boundaries of these chunks data could be corrupted. This was fixed and the chunk size increased to 10K<br/><!--<br/>pbx_admin.cpp<br/>pbx_admin.h<br/>--><!---->
      
      
=== SIP: Trap on subscription handling ===
=== Phone: Trap when selecting registration for a directory entry ===


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|[http://mantis.innovaphone.com/view.php?id=90999 90999]
|}
|}
Trap on RAS_REGISTRATION_VERIFY between sending SUBSCRIBE and receiving 200/OK.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
Trap when selecting registration for a directory entry,<br/>but only if the registration has either no name or no number.<br/><br/><!--<br/>phone_dir_ui.cpp<br/>--><!---->
      
      
=== SIP: Removed cisco-special retrieve signaling ===
=== H.323: No event should be generated in State 11 and 25 ===


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|[http://mantis.innovaphone.com/view.php?id=91020 91020]
|}
|}
Removed cisco-special retrieve signaling.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
State 25 is incoming overlap sending. This means a call was received with incomplete dialing information and the caller failed to dial more digits within the timeout of 2min. This is no indication of any malfunction but only a usage problem, so no event should be generated.<br/><br/>State 11 is disconnecting with inband announcement. A timeout happens if a user listens to the announcement for more then 30s. This could be normal.<br/><!--<br/>h323sig.cpp<br/>--><!---->
      
      
=== phone: ip222,ip232:  Plantronics APU70 - Savi 7xx - Radio Link not cleared on release from remote ===
=== SIP: Trap when cancelling call ===


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|}
|}
On a release from remote for a call set up by pressing the Talk button (headset or base) the Radio Link between base and headset was not cleared until the Talk button was pressed again. <br/><br/><!-- keys_ip222.cpp -->     <!---->
Double delete of a call entity.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== SIP: Mobility did not work due to RTP-DTMF ===
=== SIP: Heavy TLS retry load when server certificate was rejected ===


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|[http://mantis.innovaphone.com/view.php?id=91033 91033]
|}
|}
DTMF must be passed through signaling channel to get mobility working.<br/>Suppress RTP-DTMF capability in SDP answer also.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
Collateral damage from earlier fixed for DNS refreshing.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== IP3010 IP810: number of DSP channels and number of conference channels was wrong. ===
=== H.323: Incoming faststart call was sometimes not accepted as faststart ===


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|[http://mantis.innovaphone.com/view.php?id=82675 82675]
|[http://mantis.innovaphone.com/view.php?id=91136 91136]
|}
|}
Now the IP810 shows 20 DSP channels and 30 conference channels.<br/>Now the IP3010 shows 42 DSP channels and 60 conference channels.<br/><br/><!-- ac_491.dsp ac_491.h adsp2191_drv.cpp adsp2191_drv.h ip6010-config.h ip6000 config.h --><!---->
Only happens if non EFC is used on the incoming call, so this only happens in interop cases with other H.323 equipment<br/><!--<br/>h323ch.cpp<br/>--><!---->
      
      
=== SIP: Restart NAT discovery if failed ===
=== SNMP Get-Next Requests Carrying an Octet-String Value Caused Memory Leak ===


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|[http://mantis.innovaphone.com/view.php?id=82676 82676]
|[http://mantis.innovaphone.com/view.php?id=91215 91215]
|}
|}
Restart NAT discovery if failed<br/><br/><!--<br/>sip.cpp<br/>siptrans.cpp<br/>--><!---->
<!--<br/>mib.cpp<br/>--><!---->
      
      
=== IP222 IP232 IP241: Reduce DTMF level ===
=== SIP: Media negotiation for video fails if called through waiting queue or multi reg ===


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|}
|}
During DTMF receive and transmit levels similar as on IP240 are used<br/><br/><!-- ac_phone3.cpp <br/>Beim Inca ist das so definiert:<br/>" /playback_dtmf_tx_coeff_LEV12 28"<br/>" /playback_dtmf_tx_coeff_ATT12 c0"<br/>0x28=40 --> -10db level<br/>0xc0=208--> 22db attenuation<br/>also insgesamt ein level von -32db ( bei Vollauschlag ) oder -29dbm.<br/>Der alte Wert beim ac_phone3.cpp war -9db <br/><br/>Weitere Diskussion:<br/>Es gibt den Fall das inband DTMF zum IP Netz geschickt wird, da gabs in Fall 59846 die Änderung zum IP mit LEV=0x28 -->-10db und Attenuation 0xff-->18db, also mit -28db zu senden. <br/>Da das gut funktioniert und die beiden  Pegel nicht so unterschiedlich sind unde der ac_dsp3 nicht unterschiedliche Pegel zum IP und zum Codec kann  nehmen wir die -28db=-25dbm.<br/><br/><br/>Laut www.itu.int/rec/dologin_pub.asp?lang=e&amp;id=T-REC-Q.24-198811-I!!PDF-E&amp;type=item Tabelle A-1 sind die  -25dbm am unteren Ende, da ist vom Wert A <br/>-22..-30 je nach Administration die Rede. Wahrscheinlich ist das kein Problem,<br/>bei Audiocodes (ac_dsp2) kann man die sensitivity von -28 bis -38db einstellen (DTMF_DETECTION_ENERGY_THRESHOLD__28dBm)<br/><br/><br/>--><!---->
Media negotiation for video fails if called through waiting queue or multi reg.<br/>In this case the PBX has to handle offer/offer-collision.<br/>In this case the PBX must select audio and video codec.<br/>In this case the PBX must send SDP answers to both endpoints.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== IP222,IP232: Going offhook in call list always dials first list entry (not touched list entry) ===
=== SIP: Secondary target (hostname) is not resolved ===


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|}
|}
Going offhook in call list always dials first list entry (not touched list entry)<br/><br/><!--<br/>forms_color.cpp<br/>--><!---->
Usually a response to a SRV query delivers additional records containing the ip address of any target (hostname).<br/>Some DNS servers do not.<br/>Additional A querys are required.<br/>An A query was issued for the primnary target (most preferred hostname).<br/>No A query was issued for the secondary target (less preferred hostname).<br/>Fixed now.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== Memory leak after Firmware or Bootcode download ===
=== IP-DECT: Trap in Radio module (IP1202) ===


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|}
|}
After a Firmware or bootcode download via Maintenance/Download the associated command processor instance was not released.<br/><br/><!-- command.cpp --><!---->
A trap in the IP-DECT Radio module occurs if the Mobility Master is used and a duplicate IPEI command is sent to the Master. The Master handles it with a location cancel and an endpoint delete command sent to the radio. If the two commands arrives with no delay, the Radio module traps. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>dectradio.cpp<br/>dectlocalusers.cpp<br/>--><!---->
      
      
=== IP-DECT: Call transfer compatibility with SIP ===
=== Ldap Replication from NDS ===


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For compatibility reasons with the SIP protocol the call transfer initiate result message should not be sent until the connect message is received. This is changed now again.<br/><br/><!--<br/>Changed files:<br/>dectradio_call.cpp<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp<br/>--><!---->
Skip isDeleted attribute with content others than 'true'. Occurred when replicating from an NDS running in AD compatibility mode.<br/><!--<br/>ldaprep.cpp<br/>--><!---->
      
      
=== SIP: Trap if multiple calls arrive at same time ===
=== SIP: Wrong call was disconnected after successful transfer ===


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|}
Trap if multiple calls arrive at same time.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
Wrong call was disconnected after successful transfer.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== SIP: Bug in digest authorization ===
=== SIP: SDP answer for T.38 switch-over must contain multiple media descriptions ===


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Sometimes wrong method is used in digest calculation.<br/><br/><!--<br/>sip.cpp<br/>siptrans.cpp/h<br/>--><!---->
... if SDP offer contains multiple media descriptions.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== phone: ip222,ip232:  USB headset mute after activation of changes in codec parameters, headset icon cleared on status line ===
=== PBX: Not possible to login as user with non-full admin rights if 'Password protect all Pages' ===


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|}
After for example<br/>  config add AC-DSP0 HEADSET /InputGain 32<br/>  config activate<br/>the headset icon was cleared on status line and the headset was mute although the headset Talk key was handled. <br/><br/><!-- ac_codec3.cpp -->  <br/><br/><!---->
Some pages needed for the UI, which are normally not password protected, could not accessed with the reduced rights<br/><!--<br/>http.cpp<br/>--><!---->
      
      
=== PBX Waiting: Evaluate Busy on ... Calls for calls to an operator ===
=== SIP: Follow offers ptime proposal ===


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|}
|}
If a operator has configured Busy on 1 call, there should be no call from the Waiting Queue if the operator is already busy.<!---->
Better follow offers ptime proposal.<br/>Otherwise SAMwin operator does not stop sending re-INVITE.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== PBX: Avoid signaling loop after call transfer ===
=== IP0010 IP1060 IP3010 IP6010 IP22 IP24 IP28 IP302 IP305: switch from modem bypass to voice did not work ===


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|}
A signaling loop could be created by calling from a phone registered at one PBX to a phone at another PBX, then putting the call on hold and do the same call again, accept on the other side the waiting call. If both parties do then a transfer there is the signaling loop.<br/><br/>Such loop ist now detected and the call is cleared.<br/><!--<br/>pbx.cpp<br/>--><!---->
In some cases announcments with music trigger modembypass, in this case at least the fallback to voice should work.<br/><br/><!-- ac_dsp3.cpp --><!---->
      
      
=== PBX Broadcast: No diverting name sent with broadcasted call ===
=== H.323: Allow update of Registration password ===


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The diverting and original-called name info was missing from the diverting leg2 info generated by the broadcast object<br/><!--<br/>pbx_bc.cpp<br/>pbx_bc.h<br/>--><!---->
needed in case a password is changed, which is replicated to a DECT system<br/><!--<br/>h323sig.cpp<br/>h323.h<br/>voip.cpp<br/>voip.j<br/>--><!---->
      
      
=== Web-UI: Misplaced reset-required indication ===
=== H.323: PROGRESS in connected state was treated as 'unexpected' ===


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Misplaced reset-required indication on ISDN interface config (TEL1,TEL2,...).<br/><br/><!--<br/>ipac_drv.xsl<br/>isdn.xsl<br/>--><!---->
This happened with H.323 connections without registration when disconnecting a call with inband information (e.g. a call to an ISDN interface). Unnecessary events were generated.<br/><!--<br/>h323sig.cpp<br/>h323sig.h<br/>h323_tbl.tbl<br/>--><!---->
   
= V9 Hotfix 21 (9061222) =
Changes included in Version 9 hotfix21
[http://mantis.innovaphone.com/view.php?id=91721 Definition]
 
== New Features ==
 
 
      
      
=== IP0010 IP1060 IP3010 IP6010 IP810 IP302 IP305: ISDN: Enable fax detection only after connect - fixed ===
=== PBX: Allow Name (instead of Long Name) to identify user for mobility data call thru ===


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|[http://mantis.innovaphone.com/view.php?id=82900 82900]
|[http://mantis.innovaphone.com/view.php?id=91660 91660]
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Fax detection was not enabled in all cases.<br/><br/><br/>Happens in test/10.00/box/dsp/ip6010 with<br/> fix: #78316: SIP/H.323: Don't complete media negotiation if no media can be seen<br/><br/><!-- ac_dsp3.cpp --><!---->
This should simplify Opticaller configuration<br/><!--<br/>pbx_admin.cpp<br/>--><!---->
      
      
=== IP1060: Memory size is not correctly shown ===
=== IP-DECT: Static ports between Master and Radio ===


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The memory size is not correctly shown in the IP1060 in diagnostics counter page.<br/><br/><!--<br/>Changed file:<br/>ip6010.cpp<br/>--><!---->
Now the VOIP connections between the Master and the Radio use static ports instead of dynamic ones. This is useful if only a few ports should be opened through a firewall. For calls from the Radio to the Master the ports 1716 and 1717 (TLS) are used. For the default Master connection for calls from the Master to the Radio the ports 1718 and 1719 (TLS) are used. For dynamic Radio-Master connections the ports from 1722 are used. Every connection needs two ports.<br/><br/><!--<br/>Changed files:<br/>dectmaster.cpp<br/>dectradio.h<br/>dectradio.cpp<br/>dectradio_reg.h<br/>dectradio_reg.cpp<br/>h323sig.cpp<br/>--><br/><!---->
      
      
=== phone: a calling party name found by inverse directory lookup  for an external call was sometimes not stored in call list ===
=== IP-DECT: Cipher key index request for security test devices ===


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When the external call setup came in with a name identification provided by the external source and there was another name found by inverse directory lookup the name from directory was displayed on the call screen but the name identificication was stored in the call list. Now the name found by inverse directory lookup will be stored.<br/><br/><!-- app_call.cpp --><!---->
Cipher key index request procedure is changed to pass the test with security test devices. The cipher key index is used for DECT "Early Encryption"(EE).<br/><br/><!--<br/>Changed files:<br/>dectusers_if.h<br/>dectmaster.cpp<br/>--><!---->
      
      
=== PBX Routing: Node extern did not work for calls from a trunk marked as local object ===
=== Voicemail: &lt;pbx-getcallinfo out-calling-name="..."/&gt; ===


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The call was not routed back to the originating slave<br/><!--<br/>pbx.cpp<br/>--><!---->
Pass H.450 callingName into the script<br/><!--<br/>pbx/vm/*<br/>--><!---->
      
      
=== IP222 IP232 IP241: New equalizer and volume setting ===
=== PHONE_SIG_MODE_KEEP_NUMBER_TYPE can be set at phonesig startup to keep type of number in all q931 numbers ===


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- use equalizer up to approx 6db<br/>- use digital volume to adjust volume<br/> <br/><!-- ac_codec3.cpp filter/* --><br/><!---->
<!-- phonesig_if.h phonesig.cpp --><!---->
      
      
=== SIP: Trap on out of memory ===
=== PBX Trunk: Flag to block presence/dialog-info subscriptions ===


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SIP-Client allocations not deleted.<br/>Outbound control calls without facility interworking.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
Some networks e.g. sip carriers behave badly when receiving subscribes for presence/dialog-info, which cannot be handled, so there is an option added to block these.<br/><!--<br/>pbx_api.h<br/>pbx.cpp<br/>pbx_trunk.cpp<br/>pbx_trunk.h<br/>pbx_edit_trunk.xsl<br/>--><!---->
      
      
=== PBX Routing: A CFNR at a PBX object for WAN re-routing did not work if it contained escapes ===
=== SIP: Debug information for problems with STUN ===


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The number configured at the PBX object is interpreted in the context of the node of this PBX object. If escapes were needed to dial the WAN trunk, it did not work.<br/><!--<br/>pbx.cpp<br/>--><!---->
Added debug output to trace problems with STUN.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== External-UC: Presence info assigned to wrong PBX object ===
=== Gateway: FAX interface User-User-Info error response in disconnect event ===


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When presence eventlist is received from External-UC<br/>all presence info was assigned to same PBX object.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
User-User-Info response of the FAX interface is not forwarded in the alerting state. The problem exists when call to Fax interface was routed through multiple PBX. Now the UUI response is sent in the disconnect event if the response is a error notification.<br/><br/><!--<br/>Changed files:<br/>dummysig.cpp<br/>webfax.h<br/>webfax.cpp<br/>--><br/><!---->
      
      
=== PBX: Editing Config Templates impossible, if by some old firmware a strange config ended up in a User object ===
=== IP222 IP232 IP241: DSP code update ===


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The problem was a empty hardware id (hw=""). This is now ignored.<br/><!--<br/>pbx.cpp<br/>--><!---->
DSP code update to version 680<br/><br/><br/><!-- ac_494e.cpp phone_orchid.mak  ac_codec3.cpp phone-dsp.xsl new dsp file <br/><br/><br/><br/>Dear Carsten,<br/> <br/>We found the bug in our FW.<br/>You can try using the attached patch or, if you prefer, we should have a new official 680 release within two weeks.<br/>Anyway, I’ll update you once the new release is available.<br/> <br/>Regards,<br/>Eliezer <br/> <br/>--><!---->
      
      
=== PBX mobility: The forking destination put in for mobility at a user should be dialed from the node of the user ===
=== SIP: New interop tweak "No Remote Hold Signaling" ===


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This number was dialed from the node of the Mobility object. This was confusing, because this number was configured at the user and it was also different behaviour as with forking without mobility<br/><!--<br/>pbx_mobility.cpp<br/>--><!---->
New config option on gateway interfaces "No Remote Hold Signaling".<br/>Disables interworking of "inactive" into remoteHold.<br/><br/><!--<br/>voip.h<br/>sip.cpp<br/>gk.cpp/h<br/>relay_edit_voip.xsl<br/>--><!---->
      
      
=== PBX: Standyby PBX generated alarms for missing slave registrations, even if active PBX up ===
=== IP6010: DSP trace options improved ===


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There should be only an alarm, if the standby PBX is active<br/><!--<br/>pbx.cpp<br/>--><!---->
T38 trace flag worked only in one direction.<br/>Changing trace options needed a reboot.<br/><br/><!-- ac_491.cpp ac_dsp3.cpp --><br/><!---->
      
      
=== SIP: One-way audio after mutual hold on dect systems ===
=== IPv6: Disable checkmark on ethernet interfaces added ===


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Dect ep gets FTY_HOLD_NOTIFY but no FTY_RETRIEVE_NOTIFY.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
Allows to disable sending/receiving of IPv6 packets on this interface. This may be desireable for security reasons<br/><!--<br/>ip6_ethif.cpp<br/>ip6_ethif.h<br/>ip6_ethif_ip.xsl<br/>--><!---->
      
      
=== X.509: Avoid alarms on missing system time after reboot ===
=== IP222 IP232 IP241: DSP code update ===


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Allow 60 seconds for setting the system time before an alarm is set. Also certificates are now rejected silently, in that time.<br/><!--<br/>x509.cpp<br/>x509.h<br/>--><!---->
Channel was muted during silent periods ( no CNG during NLP operation ).<br/><br/><!-- ac_494e.cpp ac494e002ce4.680.10.pf.01. --><!---->
   
== Bug Fixes ==
 
 
      
      
=== SIP: Problems with presence signaling on External-UC link ===
=== PBX Mobility: Connected number from mobile phone was forwarded to caller ===


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Missing parameters on Contact-URI in 200/OK for SUBSCRIBE(presence):<br/>  maddr<br/>  transport<br/><br/>Wrong Contact-URI in presence XML in PUBLISH.<br/><br/><!--<br/>sip.cpp<br/>sip_presence.cpp<br/>--><br/><!---->
If a call is answered on the mobile phone, it should look identical to the caller to the case that the call was answered locally. This means a connected number from the mobile phone must not be forwarded.<br/><!--<br/>pbx_mobility.cpp<br/>--><!---->
      
      
=== IP-DECT: Support for more than 32 LDAP attributes ===
=== Potential Trap when rapidly switching local Media connections (Conferencing) ===


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The IP-DECT devices support up to 256 LDAP attributes now.<br/><br/><!--<br/>Changed file:<br/>dectusers.h<br/>--><!---->
There was a race condition when switching local media channels (e.g. ISDN channels to conference interfaces), which could cause media not functioning or even a trap<br/><!--<br/>media.cpp<br/>medialib.h<br/>--><!---->
      
      
=== PBX: Support for more than 64 LDAP attributes ===
=== SIP: Better handling of incoming calls ===


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The PBX supports up to 256 LDAP attributes now.<br/><br/><!--<br/>Changed file:<br/>pbx.h<br/>--> <!---->
Msg sequence INVITE,CANCEL,INVITE may result into second call rejected.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== IP-DECT: LDAP replication alarm loop ===
=== IP0010,3010,6010,1060,810: ethernet link down not detected when a cable was unplugged after boot ===


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If the LDAP replication is configured, but it can not be synchronized, there is an alarm loop. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dectusers.cpp<br/>--><!---->
The link state interrupt was triggered only once after boot, further link state changes were not indicated.<br/><br/><!-- mv78x00_drv.cpp --><!---->
      
      
=== Gateway: On IP24, IP302, IP305 it could happen that multiple routes disappeared if a route was deleted ===
=== myPBX: Rejecting an incoming broadcast call disconnected the call for all alerting endpoints ===


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Hard to predict when this happened.<br/><!--<br/>gk.cpp<br/>--><!---->
The call sould continue alerting on the other endpoints.<br/><!--<br/>pbx_client.cpp<br/>--><!---->
      
      
=== Gateway: No Media received event was generated for T.38 calls, which started without actual RTP Traffic ===
=== phone: ip222, ip232:  reset/restart USB headset when a hang condition or a port disconnect condition is detected ===


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This happend for calls to/from Fax servers, which switch to T.38 without sending any RTP packets first.<br/><!--<br/>media.cpp<br/>--><!---->
For unknown reasons some types of wireless headsets stop working after some hours or days. Either the port state changes to disabled or the device rejects control commands with a stall response. In both cases the device is reset and restarted now. If even this fails the complete USB host controller is reset and in most cases the device returns to operational state thereafter<br/><br/><!-- ohci.cpp ohci.h usb_hid.cpp --><br/><!---->
      
      
=== SIP: Multiple subscriptions for 'message-summary' ===
=== phone: ip222, ip232: Some USB headsets were not detected after a soft reset ===


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After every RAS_START another subscription for 'message-summary' is established.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
This was observed with Jabra BIZ 2400 USB at IP222/232 with hardware build 800 and newer.     <br/><br/><!-- ohci.cpp --><!---->
      
      
= V9 Hotfix 15 (9061078) =
=== IP-DECT: Trap with data calls ===
Changes included in Version 9 hotfix15
[http://mantis.innovaphone.com/view.php?id=83650 Definition]
 
== New Features ==
 
 
   
=== New remote control codes to be used for phone tests by soap applications ===


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The new UserRc codes are executed only when the addressed phone is either in handset, headset or handsfree mode, i.e when calling, connected or disconnected but not when alerting:<br/>16 - change to handset mode<br/>17 - change to headset mode<br/>18 - change to handsfree mode<br/>19 - monitor mode on (add speaker to handset or headset mode)<br/>20 - monitor mode off (back to plain handset or headset mode)<br/><br/><!-- fty.h phonesig_if.cpp phonesig.cpp app_call.cpp --><!---->
The IP-DECT Radio traps if a data call is released and the release includes a facility.<br/><br/><!--<br/>Changed file:<br/>dectradio_fty_call.cpp<br/>--><!---->
      
      
=== Alarm/Event handling: Authentication for received remote Alarms/Events ===
=== Kerberos administration: Increase maximum number of Kerberos users from 20 to 50 ===


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Allow by configuration to only accept authenticated alarms or events<br/><!--<br/>logging.cpp<br/>logging.h<br/>fault_handler.cpp<br/>fault_handler.h<br/>logging.xsl<br/>--><!---->
Increase limit on page General/Kerberos.<br/>Use POST for submitting form, instead of GET.<br/><!--<br/>kerberos_db.xsl<br/>--><!---->
      
      
=== PBX: Preparations for objects visible only if appropriate license installed ===
=== Voicemail: &lt;pbx-getcallinfo out-leg2-name=".." out-leg2-orig-name=".."&gt; ===


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for v10<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_api.h<br/>--><!---->
Passing name info of divertingLegInformation2 facility into the script.<br/><!--<br/>pbx/vm/*<br/>--><!---->
      
      
=== PBX: New Feature to allow registration with password to devices regardless of address filter ===
=== SRTP: Avoid one-way media with high start sequence numbers ===


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It is now possible to configure a flag at a device to allow a registration for this device even if there is an IP Filter which does not match. This is useful if registrations from the public internet to the PBX shall be possible. Without this feature this could be opened only for the complete PBX. Now it can be restricted to a few devices.<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_admin.cpp<br/>pbx_edit_object.xsl<br/>--><!---->
In some cases SRTP calls had one-way media because the RTP sequence number wrapped from 65535 to 0 at be beginning of the call before the receiver started receiving and processing packets.<br/><br/>The scope of start sequence numbers for RTP streams is changed from [0;65535] to [0;32767] to make sure that the receiver can always receive packets before the overflow happens.<br/><br/>The calculation of the roll-over counter (ROC) is also improved to be more reliable.<br/><br/><!--<br/>media.cpp<br/>cipher_api.h<br/>cipher_api.cpp<br/>--><!---->
      
      
=== SoftwarePhone: Signature added ===
=== H.323: Unnecessary re-initializing of rtp-channel on incoming calls to phone ===


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The SoftwarePhone installer and the install package is signed now.<br/><br/><!--<br/>Changed file:<br/>Makefile<br/>--><!---->
This did not create any problems except CPU load and together with another problem in RTP it caused no media on incoming SRTP calls approximately every 1000th call.<br/><!--<br/>h323ch.cpp<br/>phonesig.cpp<br/>phonesig.h<br/>--><!---->
      
      
=== IP-DECT: Phone book 'Use TLS' option ===
=== IP22,IP24,IP28,IP302,IP305: RTP-DTMF not offered when using a/b interface ===


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'Use TLS' option added for the central phone book search. This changes the standard port from 389 to 636 if no port is configured. The central phone book search is only available with the IP1202.<br/><br/><!--<br/>Changed files:<br/>ascom/phonebook_ldap/config_ldap.h<br/>ascom/phonebook_ldap/config_ldap.cpp<br/>ascom/phonebook_ldap/phonebook_ldap_search.cpp<br/>ascom/phonebook_ldap/phonebook_ldap.xsl<br/>--><br/><!---->
For example:<br/><br/>\tv=0<br/>\to=- 14 1 IN IP4 10.17.1.91<br/>\ts=-<br/>\tt=0 0<br/>\tm=audio 16414 RTP/AVP 8 0 18 4 97<br/>\tc=IN IP4 10.17.1.91<br/>\ta=rtpmap:97 CLEARMODE/8000<br/>\ta=fmtp:18 annexa=yes<br/>\ta=fmtp:18 annexb=yes<br/>\ta=ptime:20<br/>\ta=silenceSupp:off - - - -<br/>\ta=sendrecv<br/><br/><!--<br/>ab_sig.cpp<br/>--><!---->
      
      
=== simple static logging interface ===
=== PBX: MOH URL parameters for parked calls should refer to the parked-to and not parking user ===


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to simplify sending of log messages, alarms and errors a simple static interface to the logging module was added. log_if::log(class serial src, const class event &amp; event) passes the given event to the primary logging module (aka LOG0). This works also with 'src' = 0.<br/><br/><!-- log_if.cpp log_if.h logging.cpp http.cpp --> <!---->
The MOH URL Paramter (%l, %h, %n, ...) can be used to use different MOH Files based on the User who is holding the call. In case of a parked call this should refer to the object where the call is parked, not to the user who has initiated the parking.<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== myPBX: Pass selected device to launcher ===
=== IP-DECT: Wrong XML data in Radio endpoint ===


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Needed for version 10 remote video<br/><!--<br/>client.js<br/>client_util.js<br/>--><!---->
The XML data in the Radio endpoint is wrong and fixed now. The data is only used for the command show-endpoints, not for the GUI.<br/><br/><!--<br/>Changed file:<br/>dectlocalusers.cpp<br/>--><!---->
      
      
=== SIP: Announcing  "a=T38MaxBitRate:14400" in T.38 offer ===
=== PBX Mobility: Call from mobile endpoint to a user with mobility, but without PBX phone was disconnected ===


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Announcing  "a=T38MaxBitRate:14400" in SDP offer for T.38<br/>since all current devices support that modem speed.<!---->
This was a collateral damage of<br/><br/>fix: #86813: PBX Mobility: Take precaution against hanging calls because of misbehaving mobile phone or client<br/><br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== SIP: Set numbering plan to "ISDN/telephony" in case of "Bellcore-dr2" as Alert-Info in INVITE ===
=== SIP/DNS: Wrong port used on secondary SIP server ===


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|}
Set numbering plan to "ISDN/telephony" in case of "Bellcore-dr2" as Alert-Info in INVITE.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
If SRV query returns 2 hosts with different port, but no IP address in additional records,<br/>SIP starts two A queries for the two host names.<br/>Both resolved IP addresses are combined with the port of the most preferred host of the SRV answer.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== IP-DECT: Phone book error events ===
=== IP-DECT: Hold/Retrieve could result in no media for incoming SIP calls with SRTP ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=85001 85001]
|[http://mantis.innovaphone.com/view.php?id=92014 92014]
|}
|}
The phone book module sends an error event now if the LDAP directory search fails.<br/><br/><!--<br/>Changed files:<br/>platform/config.h<br/>dect/dectusers_if.h<br/>ascom/phonebook_ldap/phonebook_ldap.h<br/>ascom/phonebook_ldap/phonebook_ldap.cpp<br/>ascom/phonebook_ldap/phonebook_ldap_request.cpp<br/>ascom/phonebook_ldap/phonebook_ldap_search.h<br/>ascom/phonebook_ldap/phonebook_ldap_search.cpp<br/>--><!---->
This was in fact a H.323 media negotiation problem between master and slave<br/><!--<br/>h323ch.cpp<br/>--><!---->
      
      
== Bug Fixes ==
=== SIP/WLAN: Keep local Contact-URI up-to-date on subscriptions ===
 
 
   
=== IP-DECT: Trap during subscribing handsets ===


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|[http://mantis.innovaphone.com/view.php?id=92134 92134]
|}
|}
A trap occurs during subscribing handsets on the IP1200. This is a fix for the previous fix #80424 and fixed again.<br/><br/>Trap identification: <br/>XCPT: no 2 (TLB load) pc 942e23d8  ra 942e23cc  va 00000000<br/><br/><!--<br/>Changed file:<br/>dectusers.cpp<br/>--><!---->
Keep local Contact-URI up-to-date on subscriptions (e.g. message summary) when local ip address changes at runtime.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== PBX Waiting: Trap on leak-check if dtmf maps are configured ===
=== IP-DECT: Hold and Retrieve with SIP and SRTP could result in no media ===


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|[http://mantis.innovaphone.com/view.php?id=92189 92189]
|}
|}
Only happened if debug.xml leak check was used<br/><!--<br/>pbx_wait.h<br/>--><!---->
Problem with forwarding changed SRTP Keys from master to radio<br/><!--<br/>h323ch.cpp<br/>--><!---->
      
      
=== IP222 IP232 IP241: Adjust gains for better echo canceller performance ===
=== phone: ip241: do headset volume control in digital domain ===


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|[http://mantis.innovaphone.com/view.php?id=83703 83703]
|[http://mantis.innovaphone.com/view.php?id=92203 92203]
|}
|}
..<br/><!-- <br/>Use original EC mode by selecting mode 2.<br/>Max output gain reduced to avoid strong echo<br/>ADC input gain max<br/>Digital input gain set to 0db.<br/><br/>ac_phone3.cpp ac_codec.cpp --><!---->
<!-- ac_codec3.cpp --><!---->
      
      
=== myPBX: Remove unimportant notifications ===
=== IP232,IP222,IP241: Reduce flicker when opening details of call list entry ===


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|[http://mantis.innovaphone.com/view.php?id=83707 83707]
|[http://mantis.innovaphone.com/view.php?id=92242 92242]
|}
|}
The following events are not so important that the user has to be notified using a windows bubble notification.<br/>- Visibility requests<br/>- Missed calls<br/><!--<br/>client.js<br/>--><br/><!---->
Reduce flicker when opening details of call list entry.<br/><br/><!--<br/>phone_list_ui.cpp<br/>--><!---->
      
      
=== IP222 IP232 IP241: Enable noise reduction ===
=== IP-DECT: Potential trap when switching to an from 3pty conference ===


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|[http://mantis.innovaphone.com/view.php?id=83715 83715]
|[http://mantis.innovaphone.com/view.php?id=92262 92262]
|}
|}
Enable noise reduction on all handset/handsfree/DHSG headset micro.<br/>Use  DSP code 660.27.pa.03<br/><br/>Noise reduction parameter can be tuned at<br/>http://addr/AC-DSP0/mod_cmd.xml?xsl=phone-dsp.xsl<br/><br/><!-- <br/>ac494e002ce4.660.27.pa.03.h ac494eac_494e.cpp ac_phone3.cpp ac_phone3.h phone-dsp.xsl <br/>--><!---->
In case of a very unlikely collision of events, a message was sent to an already deleted object, which cause a restart<br/><!--<br/>comcertp_msp_drv.cpp<br/>+ many more dsp drivers<br/>--><!---->
      
      
=== Voicemail: Prevent &lt;prompt&gt;, &lt;record&gt; on incoming control calls ===
=== IP232,IP222,IP241: Fix for display of international numbers on phone UI ===


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|[http://mantis.innovaphone.com/view.php?id=92335 92335]
|}
|}
Turned out to cause idle-reset requests never being processed.<br/><br/>An administrative solution also exists: Watch out for the URL variable "$_noctl" in this article:<br/>http://wiki.innovaphone.com/index.php?title=Howto:Configure_the_innovaphone_Voicemail#URL_Query_String_Variables<br/><br/><!--<br/>pbx_vm.cpp<br/>--><!---->
Show international numbers as +4930123456 instead of I4930123456.<br/><br/><!--<br/>forms_color.cpp<br/>--><!---->
      
      
=== HTTPCLIENT: Allow configuration of a http authenticated URL even if the server needs no authentication ===
=== PBX: Call Completion to a user without registration, but forking, caused immediate Callback ===


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|[http://mantis.innovaphone.com/view.php?id=92341 92341]
|}
|}
If a password was configured and the server didnt need it, a HTTP put created a file of zero length.<br/><br/><!-- httpclient_i.cpp httpclient_.h --><br/><br/><!---->
Instead to call completion request should be rejected in this case<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== PBX Waiting: Trap on collision of operator connect and two-stage dialing ===
=== SIP: Huge SIP messages causes out-of-sync on TCP stream ===


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|[http://mantis.innovaphone.com/view.php?id=92373 92373]
|}
|}
If two-stage dialing (Maps) is used to call a Trunk or Gateway object, the call is sent after a blockdial timeout. If an operator connected the call before this timeout happened, a trap occured.<br/><br/>For this to happen DTMF maps and operators have to be configured on the same Waiting Queue object, with is kind of unusual<br/><!--<br/>pbx_wait.cpp<br/>pbx_wait.h<br/>--><!---->
Reading SIP messages from TCP stream gets confused by huge SIP messages.<br/>Presence exchange with external UC was disordered.<br/>Increased size limit from 100KByte to 200KByte.<br/><br/><!--<br/>siptrans.cpp<br/>--><!---->
      
      
=== PBX: Call to a Trunk/Gateway was not marked correctly as external, if no connected number was received ===
=== Gateway: Transmitting FAX documents to receiver with polling mode ===


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|[http://mantis.innovaphone.com/view.php?id=92388 92388]
|}
|}
Calls from a Trunk must be explicitly marked as internal with respective connected number, otherwise they should be treated as external<br/><!--<br/>pbx.cpp<br/>pbx_mobility.cpp<br/>--><!---->
It is not possible to send fax documents to a receiver with polling capability. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>t30.cpp<br/>--><!---->
      
      
=== HTTP: Chunked transfer fails if the last 2 bytes of the chunk header are in the next tcp packet ===
=== IP-DECT: Cipher key index table update function ===


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|[http://mantis.innovaphone.com/view.php?id=92467 92467]
|}
|}
Seen with the application platform as broken pipe<br/><br/><!-- http.cpp --><!---->
The cipher key index table is wrongly updated in the Crypto Master if a entry line yet exists. This is fixed now.<br/>The Crypto Master is needed for DECT Security Early Encryption.<br/><br/><!--<br/>Changed file:<br/>crypto_key_index_table.cpp<br/>--><!---->
      
      
=== PBX: Registration with Name/Number did not work correctly if default device not first ===
=== IP222 IP232: Handset gains changed to avoid low microphone volume ===


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|}
|}
For a registration with name or number, the information if the PBX password shall be used was always taken from the first device regardless if this was the default device (hw-id identical to name) or not<br/><!--<br/>pbx.cpp<br/>--><!---->
Low microphones levels were squelched.<br/><!-- ac_codec3.cpp <br/>Gain update nach Empfehlung von Audiocodes:<br/><br/>Dear Carsten,<br/> <br/>First, I can’t recall if there is a reason for you to use the 660.27 patch? Please remind me if there is a reason for not upgrading to version 680.<br/>In this patch version there is a bug in the EC debug mechanism (The Seq. # is not initialized properly). It would be better to upgrade.<br/> <br/>We’re suspecting the cutoffs are due to the new noise-tail remover, which you complained about (the dispersion phenomenon). It can be harm in low gains, so we would like the gain the the EC input to be higher.<br/>Please try another set of gains (In barracks – Current gains):<br/> <br/>Dig. Output gain = -5 dB (-8 dB)<br/>Anl. Output gain = -3 dB (0 dB)<br/> <br/>Anl. Input gain  = +37.5 dB (+31.5 dB)<br/>Dig. Input gain = +2 dB (+8 dB)<br/> <br/>Please let me know if you get any improvement from this above test.<br/> <br/>Thanks,<br/><br/>--><!---->
      
      
=== PBX: Partial Rerouting was prohibited in Alerting State (CFNR) ===
=== Voicemail: Emailing file names with '#'-characters failed ===


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|[http://mantis.innovaphone.com/view.php?id=83993 83993]
|[http://mantis.innovaphone.com/view.php?id=92757 92757]
|}
|}
This was done under the assumption the partial rerouting is not supported as CFNR by public networks.<br/><!--<br/>pbx.cpp<br/>--><!---->
An Escape mechanism wasn't applied<br/><!--<br/>pbx/vm_script_if.cpp<br/>--><!---->
      
      
=== PBX Waiting: Call to operator with Twin Phone Checkmark did not work anymore ===
=== Trap On Ldap DOS Attack ===


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|[http://mantis.innovaphone.com/view.php?id=92978 92978]
|}
|}
Collateral damage of<br/><br/>fix: #82858: PBX Waiting: Evaluate Busy on ... Calls for calls to an operator<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
Malign asn.1 content wasn't rejected<br/><!--<br/>ldapasn1.cpp<br/>--><!---->
      
      
=== SoftwarePhone: Product string ===
=== IP800 IP6000 IP2000: duplicated DTMF digits in transcoding scenarios ===


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|[http://mantis.innovaphone.com/view.php?id=84119 84119]
|[http://mantis.innovaphone.com/view.php?id=92981 92981]
|}
|}
The product string is changed, used e.g with the PBX registration.<br/><br/><!--<br/>Changed file:<br/>portkernel.cpp<br/>--><!---->
When tandeming VOIP links for trancoding or other purposes DTFM digits were sometimes duplicated.<br/>The RTP carried up to 25ms DTMF remaining DTMF, now its only 16ms.<br/><br/><!-- AC48xLo.c --><!---->
      
      
=== phone: provide complete dialog info to a phoneapp ===
=== PBX: Trap if user object is deleted, which is used by other applications (e.g. myPBX) ===


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|}
|}
Both group indications and dialog infos are signaled via a group indication facility. For dialog infos the parked_to_alerting member was overloaded to provide the info as expected by the existing phoneapp.<br/>Now the parked_to_alerting member is passed to a phoneapp as received.<br/><br/><!--<br/>phonesig.cpp phonesig_if.h app_reg.cpp app_gui.cpp app_fkey.cpp app_disp.cpp<br/>--><!---->
The applications need to cleanup in this case<br/><!--<br/>pbx_api.h<br/>pbx.cpp<br/>pbx.h<br/>pbx_client.h<br/>pbx_client.cpp<br/>--><!---->
      
      
=== IP222 IP232 IP241: Force same speed of the switch ports for 1000M/100M scenarios (configuration option added) ===
=== IP-DECT: CSV user export ===


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|[http://mantis.innovaphone.com/view.php?id=92991 92991]
|}
|}
In scenarios with frequent transistions of the attached PC to sleep  renegotiating the link speed may be undesired. For this case the force same speed mechanism can be disabled.<br/><br/>Other changes:<br/>1000M is only changed to 100M if the other port runs at 100M. The previous version changed from 1000M to 100M if the other port runs at 100M or 10M.<br/><br/>The statistics can be collected from the PC port or from the LAN prot or from both.<br/><br/>Packet forwarding on the PC port is disabled if the port is down to avoid misleading collision counter behaviour.<br/><br/><!-- orchid_drv.cpp orchid_drv.h orchid_drv.xsl orchid_drv_s.xsl --><br/><br/><!---->
There are some missed users in the CSV user export file. It occurs if there are users with login rights. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dectusers.cpp<br/>-><!---->
      
      
=== CDR fixes for external call detection ===
=== Voicemail: Url Parameter To Control Number Processing ===


{|
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|[http://mantis.innovaphone.com/view.php?id=93043 93043]
|}
|}
The type="ext" attribute was not set reliably. Additionaly an attribute pseudo was added to the <user/> tag to indicate the type of object the CDRis created for.<br/><!--<br/>pbx_api.h<br/>pbx.cpp<br/>pbx_exec.cpp<br/>pbx_wait.cpp<br/>pbx_bc.cpp<br/>pbx_mobility.cpp<br/>--><!---->
$_leg2tweak<br/>--<br/>Controls <pbx-getcallinfo out-leg2=".."/><br/>true(default): set leg2 to <ext-nr> from <vm-nr>+<ext-nr><br/>false: set leg2 according to received divertingInfoLeg2 facility<br/><br/>$_trailhash<br/>--<br/>Controls <pbx-getcallinfo out-cdpn=".."/><br/>true: pass trailing (en-bloc) '#' into cdpn<br/>false(default): don't pass trailing (en-bloc) '#' into cdpn<br/><br/>http://wiki.innovaphone.com/index.php?title=Howto:Configure_the_innovaphone_Voicemail#URL_Query_String_Variables<br/><br/><!--<br/>pbx_vm.cpp/.h<br/>--><!---->
      
      
=== http client : authentication was not retried after a failure when the offending request was repeated in the same session ===
=== IP0010,3010,6010,1060,810: limit number of ethernet packets processed per receive interrupt ===


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|[http://mantis.innovaphone.com/view.php?id=93130 93130]
|}
|}
When a httpclient user repeated a failing request in the same session the authentication was not tried again. Thus a change of the client side URL password or a change of the server side password had no effect until a new session was started.<br/><br/><!-- httpclient_i.cpp --> <!---->
this prevents that too much time is spent in ethernet driver in case of broadcast storms or DOS attacks.<br/><br/><!-- mv78x00_drv.cpp mv78x00_drv.h mv78x00_drv_s.xsl --> <!---->
      
      
=== IP6000: Prevent blinking error LED on old IP6000 with HW-Build &lt;110 ===
=== IP-DECT: Call transfer with enbloc dailing ===


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|[http://mantis.innovaphone.com/view.php?id=93185 93185]
|}
|}
Conference DSP driver was started on old hardware that doesnt support the conference DSP<br/><!-- adsp2191_drv.cpp --><!---->
Call transfer with enbloc dailing fails. This is fixed now.<br/>This changes also the R-key handling: after dialling a digit for a consultation call the call must disconnect with R-1 like in ring-back state.<br/><br/><!--<br/>Changed files:<br/>dectradio_call.h<br/>dectradio_call.cpp<br/>dectradio_mastercall.cpp<br/>-><!---->
      
      
=== IP222 IP232 IP241: pressing speaker key when phone is in handset mode switches to handsfree mode instead to toggle monitor mode ===
=== SRTP: Remove traces when packet authentication failed ===


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|[http://mantis.innovaphone.com/view.php?id=93203 93203]
|}
|}
handset/headset plus speaker is not supported<br/><br/><!-- codec.h ac_codec3.h ac_codec3.cpp phonesig_if.h phonesig.h phonesig.cpp app_ctl.cpp --> <!---->
SRTP and SRTCP software encryption produced traces when packet authentication failed. This is not needed, because an event is created anyway, when this happens frequently.<br/><!--<br/>cipher_api.cpp<br/>--><!---->
      
      
=== myPBX: Inconsistent display of group monitoring rights ===
=== SNMP Walk udpTable, tcpConnTable Could fail ===


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|[http://mantis.innovaphone.com/view.php?id=84350 84350]
|[http://mantis.innovaphone.com/view.php?id=93225 93225]
|}
|}
In the visibility settings groups were not displayed if the membership is "dynamic out". But members of that group still have monitoring rights.<br/><!--<br/>pbx_client.cpp<br/>--><!---->
The table index returned wasn't always lexically ascending, causing a walk to stop.<br/><!--<br/>udp4.cpp<br/>tcp4.cpp<br/>--><!---->
      
      
=== phone: when scrolling directory search results sometimes one of  the numbers of a contact was not displayed ===
=== phone: ip222,ip232: sporadic boot time trap when a Jabra LINK 280 adapter is plugged ===


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|}
|}
the tag characters assigned to the different numbers were not included in sort order. <!---->
the initialisation fails with CC=5 on first device descriptor read. after restart of host controller serial_irq() traps in reading the done list.<br/>Happens mostly with upload DRAM.<br/><!--<br/>ohci.h<br/>ohci.cpp<br/>--><!---->
      
      
=== phone: status messages for outbound external calls were sometimes garbled on display ===
=== IP-DECT/Relay: Blocked calls by hidden feature code *5/*7 ===


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|}
|}
happened when for a preceeding outbound internal call in alerting state presence info was displayed<br/><br/><!-- forms.h forms_gen.cpp forms_color.cpp app_ctl.h app_call.cpp app_disp.cpp --><!---->
Outgoing calls with beginning number *5 or *7 are blocked by the feature codes module because of hidden new service codes for an OEM device (#79028). This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dtmffty.cpp<br/>--><!---->
      
      
=== IP0010: DSP didnt start with build 9061044 ===
=== PBX: Master Slave license update period 10s instead of 10min ===


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|[http://mantis.innovaphone.com/view.php?id=93330 93330]
|}
|}
<!--ac_491.cpp adsp2191:drv.cpp--><!---->
For test purposes the period was reduced to 10s, but by accident this change ended up in version 9 as well<br/><!--<br/>pbx.h<br/>--><!---->
      
      
=== Page Ldap/Replicator/Status didn't display in WebKit ===
=== Gateway: CGPN-Maps executed even if the Route did not match in case of enbloc calls ===


{|
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|[http://mantis.innovaphone.com/view.php?id=84400 84400]
|[http://mantis.innovaphone.com/view.php?id=93400 93400]
|}
|}
Page Ldap/Replicator/Status didn't display in WebKit<br/><!--<br/>ldaprep_status.xsl<br/>--><!---->
For enbloc calls the CGPN of routes were executed even if the dialed number was incomplete.<br/><!--<br/>gk.cpp<br/>--><!---->
      
      
=== phone_orchid: displaying both  dialed and connected number may be misleading, it's better to omit dialed number ===
=== PBX: Port License counting wrong, when moving users ===


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|}
|}
when for example 022222222 was dialed and the network reported a connected number 03022222222 the display info "022222222 -> 03022222222" looked like a transfer.<br/><br/><!-- forms_color.cpp --><br/> <!---->
When moving a user from a master to a slave, the license count on the master was reduced only after reboot.<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>--><!---->
      
      
=== Hide LDAP Server Password For Viewer Accounts ===
=== PBX: XML error on User Interface if slave name with non-ascii characters used and registration was redirected from such a slave ===


{|
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|[http://mantis.innovaphone.com/view.php?id=84557 84557]
|[http://mantis.innovaphone.com/view.php?id=93483 93483]
|}
|}
Was accessible for viewers<br/><!--ldapsrv.cpp<br/>--><!---->
Conversion of UCS2 as received from the registration to UTF-8 on Web UI was wrong<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== phone: enable directory search function key also in connected state ===
=== PBX: Hide connected Endpoint did nor work for forward to other PBX or Gateway object ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84559 84559]
|[http://mantis.innovaphone.com/view.php?id=93513 93513]
|}
|}
it's sometimes useful to browse the directories for a number while talking.<br/><br/><!-- app_fkey.cpp --> <!---->
Diverting Leg Information facilities or Name Identification facilities were still forwarded in this case.<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== myPBX: Only send one command at a time ===
=== LDAP Client: SearchRequest.derefAliases Changed To neverDerefAliases(0) ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84568 84568]
|[http://mantis.innovaphone.com/view.php?id=94812 94812]
|}
|}
Queue commands instead of sending overlapping commands. This limits the number of open AJAX connections to two.<br/><!--<br/>client.js<br/>--><!---->
Was derefInSearching(1)<br/><!--<br/>ldapdir.cpp<br/>--><!---->
      
      
=== H.323: Accidential fallback to slowstart if faststart response received in PROGRESS ===
=== Gateway: Header line of FAX documents with big endian devices ===


{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84601 84601]
|[http://mantis.innovaphone.com/view.php?id=94824 94824]
|}
|}
Only happened in some H.323/SIP interop scenarios<br/><!--<br/>h323sig.cpp<br/>--><!---->
The header line of FAX documents with big endian devices was not correctly printed. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>t4.h<br/>t4.cpp<br/>--><br/><!---->
      
      
=== IP222 IP232 IP241: Updated gain and equalizer setting ===
=== SIP: Port mapping with STUN failed since DNS resolvin of STUN server failed ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84605 84605]
|[http://mantis.innovaphone.com/view.php?id=94876 94876]
|}
|}
Tuned for high MOS values according to ETSI ES202737 ES202738 ES202739 ES202740<br/><!-- ac_codec3.* platform/filter/* --><!---->
Port mapping with STUN failed since DNS resolvin of STUN server failed.<br/>Wrong STUN server port was used (5060 instead of 3478).<br/><br/><!--<br/>sip.cpp/h<br/>--><br/><!---->
      
      
=== IP0010 IP1060 IP3010 IP6010 IP22 IP24 IP28 IP302 IP305 IP800 IP6000: Minifirmware not shown on LED ===
=== H.323: No media after blind transfer to Waiting Queue on other PBX, when operator connects ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84616 84616]
|[http://mantis.innovaphone.com/view.php?id=94961 94961]
|}
|}
Minifirmware should blink long green short red. This happened only after a firmware/bootcode update.<br/><br/><!-- ip6010.cpp ip24.cpp ip800.cpp ip6000.cpp--><!---->
Happened when different coders where used on caller and called side.<br/><!--<br/>h323ch.cpp<br/>--><!---->
      
      
=== phone: ip222, ip232: status stage added to all USB control transfers ===
=== SIP: Display names need escaping of " and \\ according to RFC-3261 ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84617 84617]
|[http://mantis.innovaphone.com/view.php?id=94966 94966]
|}
|}
the status stage is mandatory as well for IN as for OUT control tranfers<br/><br/><!-- ohci.cpp ohci.h ohci_defs.h --><!---->
Acc. to RFC-3261 characters " and \\ (%x22 and %x5C) are to be escaped as "quoted-pair".<br/><br/><!--<br/>sipmsg.cpp<br/>--><!---->
      
      
=== H.323: More information on "Unexpected Message" event ===
=== PBX Waiting: Presence set for operator was not cleared, on delete or editing of Waiting Queue object ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84699 84699]
|[http://mantis.innovaphone.com/view.php?id=95016 95016]
|}
|}
The message type and the state for which this message was unexpected is needed to find out what the problem is.<br/><!--<br/>h323sig.cpp<br/>--><!---->
When configuration of the Waitinng object is now changed, any presence set by the Waiting object is cleared.<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
      
      
=== SIP: Trap during channel handling ===
=== Gateway: FAX interface interop with non-conforming Fax devices improved ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84800 84800]
|[http://mantis.innovaphone.com/view.php?id=95029 95029]
|}
|}
Rare trap when re-assigning channels.<br/><br/><!--<br/>sip.cpp/h<br/>medialib.h<br/>--><!---->
There are Fax devices sending wrong (too long) message initially after being called. In this case it is best handled by ignoring these message and wait for the retry instead of disconnecting the call.<br/><!--<br/>Changed file:<br/>t30.cpp<br/>--><!---->
      
      
=== IP232,IP222,IP241: Rendering errors when trying to use backround image with indexed colors ===
=== SIP/UDP: Sending response to wrong address and port ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84814 84814]
|[http://mantis.innovaphone.com/view.php?id=95065 95065]
|}
|}
PNG mode "indexed colors" is not supported.<br/>Trying to use an indexed color png leaves phone screen in bad shape.<br/><br/><!--<br/>read_png.cpp<br/>--><!---->
Sending response to wrong address and port.<br/>But only if Via header of incoming request contains domain name.<br/><br/><!--<br/>siptrans.cpp<br/>--><!---->
      
      
=== NAT: Don't forward DNS requests from public network ===
=== IP241: Headset receiver muffled sound ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84842 84842]
|[http://mantis.innovaphone.com/view.php?id=95164 95164]
|}
|}
As kind of denial of service attack, bursts of incoming DNS requests were seen. The nat process was forwarding these requests to the public DNS. This is a useful function for DNS requests from the private network, but not for requests from the public network.<br/><br/>These DNS requests are now discarded<br/><!--<br/>socket.h<br/>udp4.cpp<br/>udp.cpp<br/>nat.cpp<br/>--><!---->
..<br/><!-- ac_codec3.cpp Headsetvolume wieder analog geregelt, mit der Digitaleinstellugng war der klang dumpf --><!---->
      
      
=== H.323: Media Negotiation did not work for call with reverse media and media response in CALL-PROC ===
=== PBX Exec: Secretary availability monitoring did not work with multiple scretaries, with names starting identically ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84848 84848]
|[http://mantis.innovaphone.com/view.php?id=95197 95197]
|}
|}
Caused SIP interop problems with CUCM<br/><!--<br/>h323sig.cpp<br/>--><!---->
The availability state (secretary booked into the exec primary group) was not associated with the correct secretary. A compare of the names only covered the first half of the name.<br/><!--<br/>pbx_exec.cpp<br/>--><!---->
      
      
=== Maximum LDAP PDU Size Too Small ===
=== Click sounds at caller side when calling another port of same gateway ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84851 84851]
|[http://mantis.innovaphone.com/view.php?id=95436 95436]
|}
|}
Maximum LDAP PDU Size Too Small. Now internal maximum allocation unit plus a bit for encoding overhead.<br/><br/><!--<br/>ldapasn1.cpp<br/>--><!---->
occured since V9hotfix5<!---->
      
      
=== PBX Broadcast: Call to group members was not cleared when CFNR was executed ===
=== PBX Broadcast: Memory leak when calling busy broadcast object with round-robin config ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84857 84857]
|[http://mantis.innovaphone.com/view.php?id=95439 95439]
|}
|}
This was a collateral damage from<br/>fix: #79549: PBX Waiting/Broadcast: Incomplete CDRs if CFNR configured on object<br/><!--<br/>pbx.cpp<br/>--><!---->
If all destinations of a broadcast object are busy a name-id facility generates a memory leak.<br/><!--<br/>pbx_bc.cpp<br/>--><!---->
      
      
=== IPVA: V10 Code Merge Aiming To Stabilize FW Upload To DRAM ===
=== IP222 IP232 IP241: Use fifo for DSP control channel ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84862 84862]
|[http://mantis.innovaphone.com/view.php?id=95763 95763]
|}
|}
Addresses a problem in conjunction with innovaphone's automated test environment<br/><!--<br/>ipva.cpp<br/>vlance_drv.cpp<br/>ide_drv.cpp<br/>--><!---->
try to fix a trap with USB headset <br/><!-- ac_494e.cpp  <br/>The fifo was implemented in fix 63648.<br/>The fifo usage was replaced with a ASSSERT at 20.10.2011.<br/>Now it is enabled again and a DSP error event "Host transmit queue congestion" is logged when a packet cannot be sent immediately. <br/>--><!---->
      
      
=== PBX: Filters should only be applied to calls with media channels ===
=== IP232,IP222,IP241: Fix for call status display ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84892 84892]
|[http://mantis.innovaphone.com/view.php?id=95769 95769]
|}
|}
A join group operation could be prohibited with filters. This was unexpected.<br/><!--<br/>pbx.cpp<br/>--><!---->
If a call was remotely disconneced during HOLD,<br/>the held phone was constantly displaying "held" instead of "disconnected".<br/><br/><!--<br/>forms_color.cpp<br/>--><!---->
      
      
=== phone: ip222, ip232: raise alarm if an USB Headset does not respond on USB bus anymore ===
=== SIP: Interworking issue with "LifeSize Passport/LS_PP1_4.11.9 (8)" ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84893 84893]
|[http://mantis.innovaphone.com/view.php?id=95785 95785]
|}
|}
<!-- keys_ip222.cpp, ohci.cpp, ohci.h --><br/><!---->
Problems decoding large and complex SDP offer from LifeSize.<br/><br/><!--<br/>sdpmsg.cpp/h<br/>--><!---->
      
      
=== SIP: Fix for media negotiation on calls re-routed from TONE interface to outbound SIP ===
=== 802.1x (EAPOL) did not work on interfaces configured for VLAN ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84932 84932]
|[http://mantis.innovaphone.com/view.php?id=95931 95931]
|}
|}
Helps on gateway interfaces configured for media-relay with exclusive codec.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
802.1x (EAPOL) frames received without a VLAN tag must always be passed to the protocol module, even if the interface is confiured to use VLAN<!---->
   
= V9 Hotfix 22 (9061240) =
Changes included in Version 9 hotfix22
[http://mantis.innovaphone.com/view.php?id=96108 Definition]
 
== New Features ==
 
 
      
      
=== IP-DECT: Call transfer timer ===
=== PBX: New option for RTP Proxy - proxy only if different registration address ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84951 84951]
|[http://mantis.innovaphone.com/view.php?id=88439 88439]
|}
|}
The call transfer timer is stopped with the call proceeding event now. This fixes a call transfer to e.g. a mobile user with a delayed alert.<br/><br/><!--<br/>Changed files:<br/>dectradio_call.cpp<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp<br/>--><!---->
To avaoid RTP Proxy for two endpoints located within same private network behind NAT<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_general.xsl<br/>--><!---->
      
      
=== IP-DECT: Reverse phone book ===
=== PBX Executive: Allow monitoring of availability of secondary secretary, don't  treat Exec as secretary ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84953 84953]
|[http://mantis.innovaphone.com/view.php?id=95497 95497]
|}
|}
The reverse phone book does not work till hotfix 11. This is fixed again now.<br/><br/><!-- Changed files:<br/>dectradio.cpp<br/>dectradio_mastercall.cpp<br/>--><!---->
With these two additions a configuration with two executives and two secretaries, each secretary being primary to one executive an secondary to other can be configured with a single group for each secretary and both executives can monitor the availability of both secretaries.<br/><!--<br/>pbx_exec.cpp<br/>---><!---->
      
      
=== SIP: Loosing remote IP addresses when DNS becomes temorarily unavailable ===
=== SIP: Workaround for buggy registrar ===


{|
{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=84991 84991]
|[http://mantis.innovaphone.com/view.php?id=96313 96313]
|}
|}
Loosing remote IP addresses when DNS becomes temorarily unavailable.<br/>Results in interfaces without remote ip addresses.<br/><br/><!--<br/>sip.cpp/h<br/>--><br/><!---->
Workaround for buggy registrar.<br/>Different expirtes values in Contact header and Expires header.<br/><br/>\tSIP/2.0 200 OK<br/>\tVia: SIP/2.0/UDP x.x.x.x:2069;rport=2069;branch=z9hG4bK-CDDD130C<br/>\tTo: ;tag=5b8729d5-6f6353c4-cbsxz<br/>\tFrom: <sip:38795988@193.90.37.3>;tag=2109370043;epid=0090331e0bef<br/>\tCall-ID: 7ef63c56e909d311b3890090331e0bef@95.130.221.205<br/>\tCSeq: 1004 REGISTER<br/>\tExpires: 60<br/>\tContact: <sip:38795988@x.x.x.x:2069;transport=UDP>;expires=300<br/>\tUser-Agent: ZTE-SBC<br/>\tX-ZTE-Cause: "SBC-4721-2002"<br/>\tContent-Length: 0<br/><br/>Better apply the smaller expires value.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== PBX Routing: A CFNR on a slave gateway object redirecting the call to the master failed ===
=== PBX: Send forking calls as diverted calls ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=85017 85017]
|[http://mantis.innovaphone.com/view.php?id=96370 96370]
|}
|}
This was a collateral damage of<br/><br/>fix: #77874: PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX<br/><!--<br/>pbx.cpp<br/>--><!---->
So that on the called side, it will be displayed who forked the call the same way as a diverted call is displayed.<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== SIP: Switch to T.38 did not work when interworking with H.323 slowstart (XCAPI) ===
=== PBX: pbx_makecall.txt not only for mobility, but for Waiting Queue also ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=85047 85047]
|[http://mantis.innovaphone.com/view.php?id=96384 96384]
|}
|}
Switch to T.38 did not work when interworking with H.323 slowstart (XCAPI).<br/><br/><!--<br/>sip.cpp<br/>--><!---->
Allows to initiate call with simple web request from a Waiting Queue<br/><!--<br/>pbx_admin.cpp<br/>--><!---->
      
      
=== IP222 IP232 IP241: Headet volume adjust added, re-enable noise reduction in headset mode ===
=== IP-DECT: Physical location was wrong after logout/login on handset ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=85058 85058]
|[http://mantis.innovaphone.com/view.php?id=96392 96392]
|}
|}
Headset volume adjust can be done with the webinterface at<br/> http://addr/dsp.xml<br/><br/>This feature can be used to adapt to different DHSG headsets.<br/><br/>The noise reduction in headset mode is re-enabled to avoid sporadic noise with <br/><br/><!-- ac_phone3.* ac_codec3.* ac_dsp3.mak --><!---->
The physical location information is based on the redirection of the registration from the PBX at the physical location to the registration PBX. Some information was not cleared with the logout, so re-registration startet with the registration PBX right away.<br/><!--<br/>h323sig.cpp<br/>--><!---->
      
      
=== SIP: Memory leak ===
=== phone: if a number to be dialled contains a comma, the digits following the comma are sent as DTMF tones after connect ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=85083 85083]
|[http://mantis.innovaphone.com/view.php?id=96402 96402]
|}
|}
Memory leak.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
This applies to all numbers dialed en bloc, i.e. numbers dialed via indirect dialing, a phone directory or a function key. The comma must not be the first character of the number. <br/><br/><!-- phonesig.cpp --><!---->
      
      
=== SIP: Fix for STUN problems ===
=== IP-DECT: OEM PBX type info in GUI data ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=85118 85118]
|[http://mantis.innovaphone.com/view.php?id=96723 96723]
|}
|}
Try alternative STUN server address if first fails and another was gathered by DNS.<br/><br/><!--<br/>sip.cpp/h<br/>siptrans.cpp/h<br/>--><!---->
Now the GUI data includes the type info of an OEM PBX.<br/><br/><!--<br/>Changed file:<br/>dectusers.cpp<br/>--><!---->
      
      
=== IP232,IP222,IP241: Do not hide configured fkey if neither text nor icon is displayed ===
=== IP22 IP24 Ip28 IP302: Don't complete media negotiation for ab-interfaces if no media can be sent ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=85211 85211]
|[http://mantis.innovaphone.com/view.php?id=96773 96773]
|}
|}
Do not hide configured fkey if neither text nor icon is displayed.<br/>E.g. Call Forwarding fkey with CF destination but no label text.<br/><br/><!--<br/>forms_color.cpp<br/>--><!---->
For incoming calls to a phone media negotiation was already completed during ringing, so that when going off hook the media channel was already established. This causes interop problems, because there are endpoints which asssume there is inband info (e.g. ringback) if media negotiation is complete so local tones (e.g. ringback) were turned off.<br/><br/>In the past with slowstart this premature media negotiation was usefull to avoid delayed media after off-hook. With SIP or H.323 faststart there is no use anymore.<br/><br/>Needed to avoid that the DSP send CLIP and tones at the same time, which can cause sporadic DSP failures.<br/><br/><!-- ab_sig.cpp --><!---->
      
      
=== PBX: Slave with non-ASCII PBX name did not register at master ===
=== SIP: New interop tweak /register-interval ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=85235 85235]
|[http://mantis.innovaphone.com/view.php?id=97834 97834]
|}
|}
Error in utf-8 to unicode convertion in this case<br/><!--<br/>pbx.cpp<br/>--><!---->
New config file option /register-interval 60<br/>Problem is too weired to explain.<br/>This option can be used to set the REGISTER interval to a fixed value regardless of the negotiation.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
= V9 Hotfix 16 (9.061101) =
== Bug Fixes ==
Changes included in Version 9 hotfix16
[http://mantis.innovaphone.com/view.php?id=85032 Definition]
 
== New Features ==




      
      
=== Media: Preparing for G.722.1 ===
=== PBX: URI dialing, should not be case sensitive and numbers should be possible ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=85316 85316]
|[http://mantis.innovaphone.com/view.php?id=89326 89326]
|}
|}
Decoding SDP containing G.722.1<br/><br/>\tv=0<br/>\to=- 5140 5141 IN IP4 10.138.6.91<br/>\ts=-<br/>\tc=IN IP4 10.138.6.91<br/>\tt=0 0<br/>\tm=audio 50000 RTP/AVP 122 8 0 18 121 101<br/>\ta=rtpmap:122 G7221/16000<br/>\ta=fmtp:122 bitrate=32000<br/>\ta=rtpmap:121 L16/16000<br/>\ta=rtpmap:18 G729/8000<br/>\ta=fmtp:18 annexb=no<br/>\ta=rtpmap:101 telephone-event/8000<br/>\ta=fmtp:101 0-15<br/><br/><!--<br/>channel.cpp/h<br/>sdpmsg.cpp<br/>--><!---->
needed for federation<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== SIP: New config file option /tls-unchecked ===
=== IP222 IP232 IP241: LCD display is after softreset sometimes out of sync ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=85449 85449]
|[http://mantis.innovaphone.com/view.php?id=95866 95866]
|}
|}
New config file option /tls-unchecked disables validation of remote server certificate.<br/><br/><!--<br/>sip.cpp/h<br/>siptrans.cpp<br/>--><!---->
Displaycontroller needs to be stopped before restart.<br/><br/><!-- orchid_lcd.cpp --><!---->
      
      
=== IP-DECT: Configuration changes for OEM device ===
=== Httpclient: Problems with HTTPS URLs ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=85522 85522]
|[http://mantis.innovaphone.com/view.php?id=96099 96099]
|}
|}
For an OEM device the configuration update behaviour is changed.<br/><br/><!--<br/>Changed files:<br/>dectmaster.h<br/>dectmaster.cpp<br/>--><br/><!---->
Shortcut to local file I/O did not work fot HTTPS URLs.<br/><br/><!--<br/>httpclient_i.cpp<br/>--><!---->
      
      
=== SIP: New config file option /fixed-contact-addr ===
=== phone: do not report "No Media Data received" errors for connections to a recording device ===


{|
{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=85529 85529]
|[http://mantis.innovaphone.com/view.php?id=96102 96102]
|}
|}
New config file option /fixed-contact-addr to keep SIP client from changing it's Contact address into public address of NAT mapping after registration.<br/>(RFC-3581 Symmetric Response Routing)<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
some recoding devices, for example ASC never send data on a recording connection<br/><br/><!-- phonesig.cpp,sip.cpp --><!---->
      
      
=== IP-DECT: DECT security (3) ===
=== IP-DECT: Possible no media in case of media renegotiation after handover ===


{|
{|
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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=85554 85554]
|[http://mantis.innovaphone.com/view.php?id=96124 96124]
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Some changes in preparation for the new feature "DECT security". This feature will be available later (IP1202).<br/><br/><!--<br/>Changed files:<br/>common/interface/signal.h<br/>common/interface/signal.cpp<br/>ascom/mobmaster/cryptomaster.h<br/>ascom/mobmaster/cryptomaster.cpp<br/>ascom/mobmaster/dectmobmaster.h<br/>ascom/mobmaster/dectmobmaster.cpp<br/>dect/master/dectmaster.h<br/>dect/master/dectmaster.cpp<br/>dect/radio/dectradio.h<br/>dect/radio/dectradio.cpp<br/>dect/localusers/dectlocalusers.h<br/>dect/localusers/dectlocalusers.cpp<br/>ascom/httpfiles/asc_dectusers.xsl<br/>ascom/httpfiles/_inn_asc_dectusers.xsl<br/>--><br/><!---->
The problem happened in about 50% of the cases of a media renegotiation which results in a different coder after handover.<br/><!--<br/>dectradio_dectcall.cpp<br/>--><!---->
      
      
=== Phones: New config file option /recording-without-remote-party-info ===
=== IP-DECT: Show release state in IP1202 ===


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For interop of recording feature with VOXTRON application.<br/>VOXTRON application gets confused by Diversion header in INVITE.<br/><br/>    config change PHONE SIG /recording-without-remote-party-info<br/><br/><!--<br/>phonesig.cpp/h<br/>--><!---->
The release state is not shown in the IP1202. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>config.h<br/>--><!---->
      
      
=== phone_orchid: configuration prameters to adjust microphone and speaker volume ===
=== SIP: Must not answer "refresher=uac" if request contains "refresher=uas" ===


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These parameters set a volume correction factor which is applied at any volume level. Parameter changes are applied immediately even in an active call. <br/> <br/>config add AC-DSP0 HEADSET /mic-volume <mic-adjust> /spk-volume <spk-adjust><br/>config write<br/>config activate<br/><br/>   -20 <= <*-adjust> <= +20<br/><br/> <br/><!---->
Interop issue with Genband C20 PBX.<br/><br/>Must not answer<br/>   Session-Expires: 400;refresher=uac<br/>in 200/OK, if INVITE contains<br/>   Session-Expires: 400;refresher=uas<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== IP-DECT: Radio reconnect handling with OEM PBX ===
=== PBX Trunk: "Outgoing Calls restricted" did not work correctly, Presentation restricted was set, but number could be wrong ===


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The handling for the endpoint location update in combination of a radio reconnect is changed. This fixes the base station behaviour with an OEM PBX.<br/><br/><!--<br/>Changed file:<br/>dectmaster.cpp<br/>--><!---->
For example if an analog Gateway was registered to a PBX user, and this Gateway did not send a Calling Party Number with the call, the call was sent with Presentation restricted, but without digits. This could affect Billing Applications which are based on CDRs from the Gateway.<br/><!--<br/>pbx_trunk.cpp<br/>--><!---->
      
      
=== ip1202: support capture of raw ethernet packets exchanged between MSP (aka DSP) and the  firmware ===
=== phone: ip222,ip232: inbound calls automatically connected to Plantronics Savi W440/740/745 headsets with new firmware Versions ===


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All exchange between the firmware running on the ACP (Application Command Processor) and the firmware running on the MSP (Media Strem Processor) is in ethernet packet format. The Mindspeed support prefers this trace format.<br/>The capture is enabled via <br/>   config add MSP0 /mtrace<br/>It includes as well command and RTP data packets and thus duplicates the RTP packets traced by the general "All TCP/UDP Traffic" option.  <!---->
reported for:<br/>- Savi W440 with firmware 0118 on USB/DECT Dongle D100<br/>- Savi W740/745 with firmware 0115<br/><br/>reason:<br/>the newer firmware versions reject truncated output reports (no trailing 0 bytes) with STALL. The error handling for this case was wrong and caused an autoconnect.   <br/><br/><!-- keys_ip222.cpp, usb_host_if.h, ohci.cpp, ohci.h, usb_hid.cpp, usb_hid.h, usb_audio.cpp, usb_audio.h usb_msd.cpp, usb_msd.h, ip233_host.cpp, ip233_host.h --><!---->
      
      
=== HTTP-Client: Requests with specified credentials ===
=== IP-DECT: Wrong GK id of standby Master to Mobility Master ===


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Additional function for OEM httpclient::auth_request<br/>It uses the specified credentials and ignores the username and password from the configuration.<br/><!--<br/>Files:<br/>common/interface/httpclient.h<br/>common/interface/httpclient.cpp<br/>common/protocol/httpclient_i.h<br/>common/protocol/httpclient_i.cpp<br/>common/protocol/httpclient_session.cpp<br/>test/os/test_httpclient.cpp<br/><br/>Tests:<br/>test/10.00/unit-tests/httpclient/authentication/300-get-specified-basicauth.xml<br/>--><!---->
The standby Master uses a wrong gatekeeper id to register to the Mobility Master. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>dectmaster.h<br/>dectmaster.cpp<br/>--><!---->
      
      
=== Phones: New config option "Allow User Settings at Phone" ===
=== PBX-CDRs: Conference ID missing in CDRs created by Mobility data callback/callthru ===


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Allow changing of User Settings even if 'Protect Configuration at Phone' is activated.<br/><br/><!--<br/>phone_protect.xsl<br/>phone_admin_if.h<br/>phone_admin_ui.h<br/>phone_admin_ui.cpp<br/>app_gui.cpp<br/>phone_user_config.cpp<br/>--><br/><br/><!--<br/>See also: http://class.innovaphone.com/moodle2/mod/forum/discuss.php?d=1795<br/>--><!---->
This caused these calls not to show up in the reporting<br/><!--<br/>pbx_mobility.cpp<br/>--><!---->
      
      
=== IP-DECT: New hidden Radio option 'Max RTP streams' ===
=== PBX: Unexpected behaviour if too many filter were configured ===


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The new hidden option 'Max RTP streams' is added to the IP-DECT Radio module. The option is only visible for an OEM device, but can be used with config change command ("/max-rtp-streams <count>"). The feature is useful to limit the RTP streams for radios connected to the IP-DECT Master with a low data bandwidth. Conference calls are not limited with this feature.<!---->
No new filters were accepted without error message<br/><!--<br/>pbx_admin.cpp<br/>servlet.h<br/>--><!---->
      
      
=== Gateway: New config option "No blind transfer" ===
=== IP22 IP24 IP28 IP302 IP305: Sporadic DSP host interface overruns ===


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New config option "No blind transfer" to keep Gateway from handling blind transfer requests.<br/>If set blind transfer requests are passed through.<br/>Handling is performed at the next signaling hop.<br/><br/><!--<br/>relay.cpp/h<br/>relay_general.xsl<br/>--><!---->
On the small gateways the DSP hangs if control packet on host interface arrive too fast. Now the rate is limited.<br/><br/><!-- ac_494.cpp ac_494.h Queue ähnlich wie bei AC494E --><!---->
      
      
== Bug Fixes ==
=== License download not working ===
 
 
   
=== phone: ip222, ip232: USB Headset could not be disabled via Menu or by Headset Function in Enable mode ===


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sometimes the user want's to use the phone as if no headset is connected, i.e. when for example redial key is pressed after a number has been entered or a list entry has been selected the call should be started in handsfree mode and not in headset mode.<br/>If now the headset is disabled via menu or the headset(enable) function key all headset functions are completely disabled and no calls are directed to the headset, the status bar displays an icon indicating the disabled state.   <br/><br/><!--<br/>phone_user.h<br/>phone_user.cpp<br/>app_ctl.h<br/>app_ctl.cpp<br/>app_disp.cpp<br/>app_form.cpp<br/>app_fkey.cpp<br/>app_gui.cpp<br/>phone_user_if.h <br/>forms.h<br/>forms_color.cpp<br/>icons.mak<br/>headset_disabled.png<br/>ac_codec_3.cpp<br/>--><br/><!---->
License download stops.<br/>Licenses are not downloaded.<br/><br/><!--<br/>license.cpp<br/>--><!---->
      
      
=== H.323: Interop Problem with CUCM SIP for Transfer ===
=== PBX: Changing of PBX Object Name did not change the name to be used for registration ===


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For the transfer the CUCM first sets the call on hold and then requests a new media proposal from this call, which we cannot deliver. The request is just ignored, there should be an answer.<br/><!--<br/>h323ch.cpp<br/>--><!---->
So if the name of the PBX registering as slave was changed as well, it did not register anymore. The PBX object had to be deleted and created with new name.<br/><!--<br/>pbx_admin.cpp<br/>pbx_edit_loc.xsl<br/>--><!---->
      
      
=== IP152: Call replacement (blind transfer) did not work ===
=== Gateway: FAX interface on IPVA ===


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Call replacement (blind transfer) ends up in no audio.<br/><br/><!--<br/>coldfire_afe.cpp<br/>--><!---->
The FAX interface on IPVA can not connect to a remote device because of wrong protocol events. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>facsimile_def.h<br/>--><!---->
      
      
=== Trap when disabling Gateway interface ===
=== SRTP: One way audio after some minutes on IP6000 IP2000 IP6010 IP0010 ===


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0:0204:814:3 - SIP_UDP.17 -> SIP_TRANSPORT.1 : SOCKET_RECVFROM_RESULT(87.173.157.2:5060,10.2.2.1:5060;95fd:75c0:9485:bc94:9485:bc30:9406:630c:5060,948a:2198:9476:5924:19:72:0:1:5060)<br/>      data(913),SIP_TRANSPORT<br/>      <br/>\tREGISTER sip:gwdl_ip800 SIP/2.0<br/>\tVia: SIP/2.0/UDP 87.173.157.2:5060;branch=z9hG4bKA072411261DD0135<br/>\tRoute: <sip:87.139.89.223;lr><br/>\tFrom: <sip:93@gwdl_ip800>;tag=4110839322<br/>\tTo: <sip:93@gwdl_ip800><br/>\tCall-ID: DECE171E57A1898F@192.168.178.1<br/>\tCSeq: 7998 REGISTER<br/>\tContact: <sip:93@87.173.157.2;uniq=D526C620EFAB7D1ACBBC469D33242><br/>\tAuthorization: Digest username="_!(schuetz)", realm="gwdl_ip800", nonce="7977d820e909d311", uri="sip:gwdl_ip800", response="6999f52d0e7d4ddbd18b23b59936aaa6", algorithm=MD5, cnonce="F56053DD72D3E1B3", qop=auth, nc=00000401<br/>\tExpires: 1800<br/>\tMax-Forwards: 70<br/>\tUser-Agent: AVM FRITZ!Box Fon WLAN 7112 (UI) 87.04.87 (Jun  7 2011)<br/>\tSupported: 100rel,replaces,timer<br/>\tAllow-Events: telephone-event,refer,reg<br/>\tAllow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH<br/>\tAccept: application/sdp, multipart/mixed<br/>\tAccept-Encoding: identity<br/>\tContent-Length: 0<br/><br/><br/>0:0204:815:5 - SIP-Client(SIP-CLIENT.0) <sip:93@gwdl_ip800;user=phone> ...<br/>0:0204:816:5 - SIP-Client(SIP-CLIENT.0) Idle->Registering<br/>0:0204:816:6 - SIP_RAS_APP.3 -> GK.0 : RAS_DISCOVERY, ip=87.173.157.2<br/>      data(6),GK<br/>      00 00 00 02  39 33                                  ....93          <br/>0:0204:816:7 - GK.0 -> SIP_RAS_APP.3 : RAS_DISCOVERY_CONFIRM<br/>0:0204:817:0 - SIP_RAS_APP.3 -> GK.0 : RAS_REGISTRATION(87.173.157.2:5060)<br/>      data(6),GK<br/>      00 00 00 02  39 33                                  ....93          <br/>0:0204:817:1 - GK.0 -> SIP_RAS_APP.3 : RAS_REGISTRATION_CONFIRM <br/>      data(32),SIP_RAS_APP<br/>      00 00 00 02  39 33 00 01  00 16 00 5f  00 21 00 28  ....93....._.!.(<br/>      00 73 00 63  00 68 00 75  00 65 00 74  00 7a 00 29  .s.c.h.u.e.t.z.)<br/>0:0204:817:1 - DEBUG this=948b325c event.sig=0<br/>0:0204:817:2 - DEBUG this->reg_reference=0 event.reference=948b3ebc<br/><!---->
Applies to v9hotfix21 on IP6000, IP2000, IP6010, and IP0010.<br/><br/>Collateral damage from fix #91892: SRTP: Avoid one-way media with high start sequence numbers.<br/><br/><!--<br/>cipher_api.cpp<br/>mv78x00_crypt.cpp<br/>--><!---->
      
      
=== Presence note got lost ===
=== Gateway: CGPN-Maps executed even if the Routing was already completed ===


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When configuring a presence on the phone (Main menu/User setup/Presence)<br/>the note is not save along with the selected activity<br/><br/><!--<br/>app_gui.cpp<br/>--><!---->
A CGPN map, which was configured in a route following the route, which was actually executed was executed as well.<br/><br/>This was a collateral damage from fix: #93400: Gateway: CGPN-Maps executed even if the Route did not match in case of enbloc calls<br/><!--<br/>gk.cpp<br/>--><!---->
      
      
=== PBX: Potential trap on registration of a slave PBX, with Master GK-ID configuration changes ===
=== IP22 IP24 IP28 IP302 IP305: Sporadic DSP host interface overruns - CLIP disabled, new trace option ===


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The trap happened if on the Slave a Master GK-ID was configured, then the slave registered, and afterwards the Master GK-ID was removed again and the slave registered again and then was restarted once more.<br/><!--<br/>pbx.cpp<br/>--><!---->
On the small gateways the DSP hangs in some conditions.<br/>Now the trace-stop is replaced with an Assert to recover from this situation.<br/><br/>To get a trace of this condition a new trace option is added at dsp.xsl, called txt-trace. This traces the DSP message as text, so that they can be read out after a trap.<br/>Typical usage is to enable DSP-trace, DSP control messages DSP data messages and DSP txt trace.<br/>DSP pcm trace and DSP T38 trace should be off to avoid excessive debug load.<br/><br/><br/>Also, the CLIP messages are disabled since they caused problems in the past.<br/><br/><br/><!-- ac_494.cpp ac_494.h ac_dsp3.cpp trace.xsl gk.cpp --><!---->
      
      
=== Presence: Note moves into 'tel' presence ===
=== H.323: One-way-voice if SRTP call to a Waiting queue is forwarded via Waiting Queue Maps to a phone ===


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Presence note moves into 'tel' presence when changing activity by use of 'presence' fkey.<br/><br/><!--<br/>app_disp.cpp<br/>--><!---->
This is a problem with SRTP key exchange, which could happen in other traffic scenarios as well.<br/><!--<br/>h323ch.cpp<br/>--><!---->
      
      
=== IP232,IP222: Automatically enter input mode 'alpha' when entering directory search screen through fkey ===
=== IP-DECT: Web UI administrator user list removed on IP1202 ===


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Automatically enter input mode 'alpha' when entering directory search screen through fkey.<br/><br/><!--<br/>phone_dir_ui.cpp<br/>forms_color.cpp<br/>--><!---->
The Web UI administrator user list is removed on the IP1202 now.<br/><br/><!--<br/>Changed files:<br/>httpfiles.mak<br/>_inn_asc_dect_users_right.xsl<br/>--><!---->
      
      
=== SIP: Interop with Genband SBC ===
=== IP28: Click sounds at caller side when calling ip28 gateway ===


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Handling this kind of SDP offer:<br/><br/>\tv=0<br/>\to=IOTMSX1-0 17 2 IN IP4 206.165.51.38<br/>\ts=sip call<br/>\tc=IN IP4 0.0.0.0<br/>\tt=0 0<br/>\tm=audio 42076 RTP/AVP 0 8 4 18 101 13<br/>\ta=sendonly<br/>\ta=rtpmap:101 telephone-event/8000<br/>\ta=fmtp:18 annexa=yes<br/>\ta=fmtp:18 annexb=no<br/>\ta=fmtp:101 0-15<br/>\ta=ptime:20<br/>\ta=silenceSupp:off - - - -<br/><br/><!--<br/>sip.cpp<br/>--><br/><!---->
<!--<br/>si3241_drv.cpp<br/>si3241_drv.h<br/>--><!---->
      
      
=== PBX-Exec: Call at secretary for executive had wrong destination name/number ===
=== IP28: Sometime Ringing stopped working on an anlog port. Worked again only after reset. ===


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Call should be displayed as call to the scretary, diverted by the executive<br/><!--<br/>pbxb_exec.cpp<br/>--><!---->
This happened due to low ringing volatge, the default value of "low" on the interface configuration for the ringing voltage caused this to happen.<br/><!-- si3241_drv.cpp --><!---->
      
      
=== Media: RTP did not work on switch from slowstart T.38 to EFC G.711 ===
=== IP22 IP24 IP28 IP302 IP305: Sporadic DSP host interface overruns - Updated DSP code ===


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T.38 retransmission packets were sent, which caused problems with the RTP<br/><!--<br/>media.cpp<br/>--><!---->
Clip enabled on all channel.<br/><br/><!-- ac_dsp3.cpp ac_dsp3.h ac494.cpp ip24.mak --><!---->
      
      
=== IP222 IP232 IP241: DSP Update to Version 680.05 ===
=== phone: ip222,ip232,ip241: Local Network Coder default for User-2..6 was G711 instead of G722 as for User-1 ===


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Sporadic problems with noise reduction fixed.<br/>Noise reduction can be enabled for Handset/Headset/Handsfree mode<br/><br/><!-- <br/>DSP code details at \\\\innovaphone.sifi\\dfs\\Dokumente\\Hardware\\G7xx\\Audiocodes\\firmware\\firmware_ac49x\\680.05<br/><br/>ac49x_680/* ac_phone3.cpp ac_phone3.h ac_494.cpp ac_494e.cpp ac491.cpp ac_firmware.h phone_orchid.mak phone_dsp.xsl --><!---->
WEB-Interface "Phone/User-2..6/General/Options/Local Network Coder" was preset to G711 instead to G722 as for User-1. <br/><br/><!-- phone_user_ui.cpp, phone_user_ui.h --><!---->
   
= V9 Hotfix 23 (9061252) =
Changes included in Version 9 hotfix23
[http://mantis.innovaphone.com/view.php?id=96632 Definition]
 
== New Features ==
 
 
      
      
=== SIP: Providing private RTP address although public RTP address is available ===
=== PBX Waiting: timeout argument for pbx_makecall.txt URL ===


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During re-negotiation:<br/>Providing private RTP address to external endpoint although public RTP address is available.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
A timeout argument was added to the PBX0/ADMIN/pbx_makecall.txt URL to cancel an outgoing call initiated with this URL.<br/><!--<br/>pbx_admin.cpp<br/>pbx_api.h<br/>pbx_wait.cpp<br/>pbx_wait.h<br/>pbx_xml.cpp<br/>pbx_xml.h<br/>pbx_bc_conf.cpp<br/>pbx_bc_conf.h<br/>--><!---->
      
      
=== SIP: Do not send "504 Server Time-out" after "200 OK" ===
=== ISDN: Send legacy Redirecting Number for 'old' Fax Servers ===


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Do not send "504 Server Time-out" after "200 OK".<br/><br/><!--<br/>siptrans.cpp<br/>--><!---->
The redirecting number is an old style information element, which contains part of the information as the diverting leg2 facility. Some Fax Servers do not understand the leg2 facility.<br/><!--<br/>q931.h<br/>q931.cpp<br/>nt_tbl.h<br/>signal.cpp<br/>signal.h<br/>relay.cpp<br/>--><!---->
      
      
=== IP-DECT: Call transfer compatibility with SIP ===
=== PBX-CDRs: Better CDR for pickup ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=85795 85795]
|[http://mantis.innovaphone.com/view.php?id=97681 97681]
|}
|}
For compatibility reasons with the SIP protocol no hold signal should be sent to the call transfer destination. This is fixed again.<br/><br/><!--<br/>Changed file:<br/>dectradio_call.cpp<br/>--><!---->
A pickup was not indicated in the CDRs<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== SIP: Handling of "sendonly" offer after "inactive" offer ===
=== PBX-SOAP: Status of Boolean object indicated as local number ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=85857 85857]
|[http://mantis.innovaphone.com/view.php?id=97738 97738]
|}
|}
Switching from "inactive" into "recvonly".<br/>Giving REMOTE_RETRIEVE and HOLD_NOTIFY to app.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
When monitoring a Boolean object with SOAP a call is indicated. The local number of this call is set based on the status of the boolean object (00 automatic-off, 01, automatic-on, 10 - manual-off, 11 - manual-on)<br/><!--<br/>pbx_bool.cpp<br/>--><!---->
      
      
=== SIP: Ignore From-URI in re-INVITE and UPDATE when "from-change" not supported ===
=== PBX-SOAP: UserPark allows to park to another object ===


{|
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|Id
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|[http://mantis.innovaphone.com/view.php?id=85873 85873]
|[http://mantis.innovaphone.com/view.php?id=97741 97741]
|}
|}
Ignore From-URI in re-INVITE and UPDATE when "from-change" not supported by remote side.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
The argument 'cn' was ignored in v9 and earlier, now it can be used to identify a destination for the park.<br/><!--<br/>pbx_xml.cpp<br/>--><!---->
      
      
=== Phones: Touching should start dialing of selected directory entry ===
== Bug Fixes ==
 
 
   
=== SUBSCRIBE for MWI not correctly handled after change of IP address ===


{|
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|[http://mantis.innovaphone.com/view.php?id=96898 96898]
|}
|}
Touching should start dialing of selected directory entry<br/>instead of opening directory entry for editing.<br/>At least in Context of Wahlvorbereitung.<!---->
SUBSCRIBE for MWI not correctly handled after change of IP address<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== SNMP-Traps: agent-addr wasn't reflecting altered IP adress ===
=== phone: ip222,ip232: USB headset echo effects in call setup phase when a G722 call is started by a CTI application ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=85970 85970]
|[http://mantis.innovaphone.com/view.php?id=96974 96974]
|}
|}
SNMP-Traps: The SNMP trap's agent-addr member carried an outdated ip adress if the DHCP leased ip address changed.<br/><!--<br/>snmp.cpp<br/>snmp.h<br/>--><!---->
The caller hears the calling tones and it's own speech from the headset microphone but the callers speech is not transmitted to the called party.<br/><br/><!-- ac_phone3.cpp --> <!---->
      
      
=== SIP: Optimization when validating local media address ===
=== H.323: Offered packetization should be honored for SIP interoperability ===


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|[http://mantis.innovaphone.com/view.php?id=86029 86029]
|[http://mantis.innovaphone.com/view.php?id=96983 96983]
|}
|}
Optimization when validating local media address.<br/>Keep number of SOCKET_GET_LOCAL_ADDR low.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
Problem happend with calls from Samwin CBC<br/><!--<br/>h323ch.cpp<br/>h323ch.h<br/>--><!---->
      
      
=== DHCP: client IP connectivity lost when a renew/rebind request for the currently assigned address was refused by the server ===
=== RTP-DTMF: Must increase duration field when sending RTP-Event with END marker ===


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|[http://mantis.innovaphone.com/view.php?id=86030 86030]
|[http://mantis.innovaphone.com/view.php?id=97001 97001]
|}
|}
When a client renew/rebind request is refused by the server providing the current address the client starts a new discovery. But in case of success the new address was not set and the client could not be reached anymore. <!---->
Must increase duration field when sending RTP-Event with END marker to comply with RFC.<br/><br/><!--<br/>media.cpp<br/>--><!---->
      
      
=== ENUM: Port in SIP-URI was not honored ===
=== ASN.1 BER: Decoding of Sequence Member with indefinite length failed ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=86037 86037]
|[http://mantis.innovaphone.com/view.php?id=97072 97072]
|}
|}
Port in SIP-URI was not honored in regex of DNS result.<br/><br/><!--<br/>enumlib.cpp/h<br/>relay.cpp<br/>sip.cpp<br/>--><!---->
An Avaya supplementary service couldn't be decoded<br/><!--<br/>asn1_ber.cpp<br/>--><!---->
      
      
=== IP-DECT: Transferred/rerouted call display ===
=== H.323: Channel Close sometimes not sent on hold ===


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|[http://mantis.innovaphone.com/view.php?id=86041 86041]
|[http://mantis.innovaphone.com/view.php?id=97080 97080]
|}
|}
The remote party number of transferred and rerouted calls are not correctly shown in the handset's display. This fixes the display of CTI initiated calls.<br/><br/><!--<br/>Changed file:<br/>dectradio_call.cpp<br/>--><!---->
Esspecially on the second hold within a call the Channel Close was not sent to the party, which put the other on hold. This caused the channel not beeing turned off on this side (the other side receives music on hold in this case)<br/><!--<br/>h323ch.cpp<br/>channel.h<br/>channel.cpp<br/>--><!---->
      
      
=== IP-DECT: MAC-alias change of OEM device ===
=== IP4 did not work anymore when IP6 was disabled  via WEB interface ===


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|[http://mantis.innovaphone.com/view.php?id=86047 86047]
|[http://mantis.innovaphone.com/view.php?id=97111 97111]
|}
|}
The MAC-alias of an OEM device was changed and this results in conflicts within several DECT modules. Different product short names of the same device are correctly accepted now.<br/><br/><!--<br/>Changed files:<br/>dectmaster.h<br/>dectmaster.cpp<br/>cryptomaster.cpp<br/>dectmobmaster.h<br/>dectmobmaster.cpp<br/>cryptomaster_registration.h<br/>registration_context.h<br/>registration_context.cpp<br/>--><br/><!---->
When the "IP6/ETNx/IP6/Options/Disabled" was checked the IP4 operations did stop after a while.<br/><br/><!-- ip6_ethif.cpp --><!---->
      
      
=== ip1202: Send Inband DTMF did not work ===
=== phone: ip222,ip232: Plantronics Savi W440 dosn't report Talk-Key events in a call established at phone or by a CTI application ===


{|
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|[http://mantis.innovaphone.com/view.php?id=97115 97115]
|}
|}
the "DECT/Config/Master/Send Inband DTMF" checkmark had no effect. <!---->
When a call via this headset was initiated/accepted by the Redial-Key, the Headset(Mode:Control) function key or a CTI application, the call could not be disconnected by pressing the Talk-Key at the headset because the Headset did not report this action.<br/><br/><!-- keys_ip222.cpp --><br/><!---->
      
      
=== SIP: Support for multiple audio media descriptions ===
=== phone: DTMF digits following a comma in a number to be dialed were not handled correctly in some cases ===


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|[http://mantis.innovaphone.com/view.php?id=97150 97150]
|}
|}
One without encryption and one with encryption:<br/><br/>v=0<br/>o=OpenStage-Line_0 968610650 446118927 IN IP4 10.30.1.144<br/>s=SIP Call<br/>c=IN IP4 10.30.1.144<br/>t=0 0<br/>m=audio 5012 RTP/AVP 9 8 0 18 101<br/>a=rtpmap:9 G722/8000<br/>a=rtpmap:8 PCMA/8000<br/>a=rtpmap:0 PCMU/8000<br/>a=rtpmap:18 G729/8000<br/>a=rtpmap:101 telephone-event/8000<br/>a=silenceSupp:off - - - -<br/>a=fmtp:18 annexb=no<br/>a=fmtp:101 0-15<br/>a=sendrecv<br/>m=audio 5010 RTP/SAVP 9 8 0 18 101<br/>a=rtpmap:9 G722/8000<br/>a=rtpmap:8 PCMA/8000<br/>a=rtpmap:0 PCMU/8000<br/>a=rtpmap:18 G729/8000<br/>a=rtpmap:101 telephone-event/8000<br/>a=silenceSupp:off - - - -<br/>a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ujVU8G6kgknZnPflRwx8tadNskkp9glas/DFCbC3<br/>a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:+uoFBNVVhDY5OOGvMOdAlvpxvc98hX/VeProhlwH<br/>a=fmtp:18 annexb=no<br/>a=fmtp:101 0-15<br/>a=sendrecv<br/><br/><br/><!--<br/>sip.cpp/h<br/>sdpmsg.cpp/h<br/>--><!---->
- in the "Destination Number" configured under "Phone/Direct Dialing" in conjunction with a nonzero "Autodial Timeout": the DTMF digits were sent as dial digits<br/>- with a nonzereo "Enblock Dialing Timeout" configured under "Phone/User x/General/Options": sending of DTMF digits was delayed by the configured timeout\t<!---->
      
      
=== Change order of links on debug page ===
=== Gateway: FAX interface on IP800/IP305/IP302 ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=86121 86121]
|[http://mantis.innovaphone.com/view.php?id=97571 97571]
|}
|}
Make "Tracing" the first item so that no leak check is triggered when opening the page.<br/><!--<br/>debug.xml<br/>maintenance.xml<br/>--><!---->
Fix for the last fix #96660.<br/>The FAX interface on the IP800/IP305/IP302 can not connect to a remote device because of wrong protocol events. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>facsimile_def.h<br/>--><br/><!---->
      
      
=== Phones: Using config option "CGPN" as CGPN on outbound SIP calls ===
=== H.323: Fast Unregister/Register operations could lead to failed registrations, in case of fixed signaling ports ===


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|[http://mantis.innovaphone.com/view.php?id=86122 86122]
|[http://mantis.innovaphone.com/view.php?id=97637 97637]
|}
|}
Using config option "CGPN" as CGPN on outbound SIP calls.<br/><br/><!--<br/>phonesig.cpp<br/>--><!---->
A listening socket could still be in use. Only happened with IP-DECT and multi-master.<br/><!--<br/>h323sig.cpp<br/>h323.h<br/>--><!---->
      
      
=== phone: ip222, ip232: recover from USB port failure probably caused by electric sparks ===
=== PBX: Name-Id of busy destination was not forwarded to other PBX ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=86125 86125]
|[http://mantis.innovaphone.com/view.php?id=97646 97646]
|}
|}
An USB port failure is indicated when a Plantronics DA45 headset adapter is plugged and a certain kind of table lamp (halogen) is switched on or off.<br/>It happens independent of current state of the headset (idle or in call)<br/>but only with the abovementioned adapter.<br/>The exact reason is not known yet, may be it's an electric spark from the switch of the lamp or some pulse.<br/>The fix is to reset the port and to restart the plugin process, a possibly active call is terminated.<br/><br/><!-- keys_ip222.cpp ohci.cpp ohci.h ohci_defs.h --> <br/><!---->
Display was different when calling a busy phone on local PBX or on another PBX<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== phone: coder settings of a "Create Registration"  function key were not applied to the created registration ===
=== IP22 IP24 IP28 IP302 IP305: ASSERT on DSP queue overrun added ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=86164 86164]
|[http://mantis.innovaphone.com/view.php?id=97653 97653]
|}
|}
<!-- app_fkey.cpp --><!---->
<!-- ac494.cpp --><!---->
      
      
=== Disable leak check if debug flag is not set ===
=== phone: "Prepare Override" function key did not work since V9hotfix21, the overriding source address was ignored ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=86165 86165]
|[http://mantis.innovaphone.com/view.php?id=97665 97665]
|}
|}
The leak check is only allowed if the config flag CPU /debug is set.<br/><!--<br/>command.cpp<br/>leaks.xml<br/>--><!---->
<!-- phonesig.cpp --><!---->
      
      
=== AD Replication: Configuration Buffer Increased ===
=== SIP: Trap when terminating a call while re-negotiation is ongoing ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=86211 86211]
|[http://mantis.innovaphone.com/view.php?id=97675 97675]
|}
|}
Was too small for many maps<br/><!--<br/>ldaprep.cpp/.h<br/>--><!---->
Trap when terminating a call while re-INVITE is pending.<br/><br/><!--<br/>siptrans.cpp<br/>--><!---->
      
      
=== whistling tone in all ip28 a/b ports on incoming call ===
=== phone: ip230,ip240,ip241 :  unefined codes received from a DHSG Headset basestation were misinterpreted as Hookswitch indication ===


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|[http://mantis.innovaphone.com/view.php?id=97732 97732]
|}
|}
This phenomena occured after few day uptime after sending CLIP<!---->
sometimes DHSG Headset basestations send codes not defined for DHSG which are silently discarded now. <br/><br/><!-- keys_gen.cpp --><!---->
      
      
=== ip72 firmware did not boot anymore since V9hotfix11 ===
=== H.323: Potential Trap when reconfiguring an H.323 registration ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=86246 86246]
|[http://mantis.innovaphone.com/view.php?id=97820 97820]
|}
|}
<!-- ip220.cpp --><!---->
This is a collateral damage from<br/><br/>91815: IP-DECT: Static ports between Master and Radio<br/><!--<br/>h323sig.cpp<br/>--><!---->
      
      
=== SIP: Avoid re-configuration of DSP channel when processing re-INVITE ===
=== PBX Executive: Calls with calling id restriction and without calling id, were sent to executive, even if secretary available ===


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|[http://mantis.innovaphone.com/view.php?id=97897 97897]
|}
|}
Fix is required for interop with SIP devices sending re-INVITE for session-refresh,<br/>but incrementing version field in SDP body, altough there is no change in SDP.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
This happend for calls coming in from public ISDN with calling id presentation restriction thru a trunk object without number<br/><!--<br/>pbx_exec.cpp<br/>--><!---->
      
      
=== phone: headset function key mode 'control' could be configured via WEB interface only ===
=== SIP: Trap when using STUN ===


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|[http://mantis.innovaphone.com/view.php?id=97898 97898]
|}
|}
must be possible locally at the phone too<br/><br/><!-- phone_user_func.cpp phonetxt.h phonetxt-utf8.cpp phonetxt-latin1.cpp phonetxt-utf8.base --><!---->
Trap when using STUN.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== phone: Partner state provided via dialog info was not reset when the subscription call was released  because of network errors ===
=== SIP: Dialog-Info was encoded with wrong state attribute ===


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|[http://mantis.innovaphone.com/view.php?id=86330 86330]
|[http://mantis.innovaphone.com/view.php?id=97926 97926]
|}
|}
<!-- phonesig.cpp app_reg.cpp --><!---->
Dialog-Info was encoded with wrong state attribute "full".<br/>Must be "partial".<br/><br/><!--<br/>sip_dialog_info.cpp<br/>--><!---->
      
      
=== Gateway: Routing on blind transfer call starts from wrong IF ===
=== SIP: Logging was wrong ===


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|[http://mantis.innovaphone.com/view.php?id=98037 98037]
|}
|}
Transfered endpoint was used as source interface on routing.<br/>Better use transfering endpoint as source on routing of (blind) transfer call.<br/>Also transfer-to endpoint missed ctSetup.<br/>Also transfered endpoint missed ctComplete.<br/><br/><!--<br/>relay.cpp<br/>sip.cpp<br/>--><!---->
Logging was wrong.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
      
      
=== Linux: Start-up failures ===
=== PBX: Potential trap when receiving unknown presence activity ===


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|[http://mantis.innovaphone.com/view.php?id=98043 98043]
|}
|}
Linux start-up is improved. This fixes hanging Linux start-ups caused by compact-flash failures.<br/><br/><!--<br/>Changed files:<br/>ip6010.h<br/>ip6010.cpp<br/>--><!---->
In the respective version unknown activities are mapped to "busy"<br/><!--<br/>fty.cpp - rollback of this change<br/>h450.cpp<br/>--><!---->
   
= V9 Hotfix 24 =
Changes included in Version 9 hotfix24
[http://mantis.innovaphone.com/view.php?id=98531 Definition]
 
== New Features ==
 
 
      
      
=== IP-DECT: GUI Master Configuration ===
=== phone: ip222, ip232: support Jabra UC Voice 750 and Jabra Speak 510 ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=86442 86442]
|[http://mantis.innovaphone.com/view.php?id=98092 98092]
|}
|}
Disabling LDAP Directory Search fixed for the IP1202.<br/><br/><!--<br/>Changed file:<br/>_inn_asc_dectmaster.xsl<br/>--><!---->
<!-- keys_ip222.cpp --><!---->
      
      
=== Linux: GUI in viewer mode ===
=== IPVA: ETHx Transmit Queue Size Limited To 1MB ===


{|
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|[http://mantis.innovaphone.com/view.php?id=86455 86455]
|[http://mantis.innovaphone.com/view.php?id=98187 98187]
|}
|}
The Linux General page is fixed for the viewer only mode.<br/><br/><!--<br/>Changed files:<br/>config_options.cpp<br/>linux.xsl<br/>--><!---->
Was unlimited<br/><!--<br/>vlance_drv.cpp<br/>--><!---->
      
      
=== SIP: PRACK after CANCEL contains bad RAck header value ===
=== Linux: Shutdown warning message ===


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|[http://mantis.innovaphone.com/view.php?id=98438 98438]
|}
|}
CSeq or original INVITE transaction was damaged.<br/>But only if CANCEL has been sent right before PRACK.<br/>CANCEL is sent before PRACK only if SDP answer of provisional response is invalid.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
If Linux is running, a shutdown warning message is shown at the Linux General page now.<br/><br/><!--<br/>Changed file:<br/>linux.xsl<br/>--><!---->
      
      
=== PBX: Conference trap ===
=== SIP/TLS: Using domain name as fall back to proxy name when comparing to the certificat "subject" ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=86473 86473]
|[http://mantis.innovaphone.com/view.php?id=98643 98643]
|}
|}
Potential trap in the PBX BC conference object is fixed.<br/><br/><!--<br/>Changed files:<br/>pbx_bc_conf.h<br/>pbx_bc_conf.cpp<br/>--><!---->
Using domain name as fall back to proxy name when comparing to the certificat "subject".<br/><br/><!--<br/>siptrans.cpp<br/>--><!---->
      
      
=== SNMP: Obsoleted Enterprise-Specific Trap "innoIsdnFailure" ===
=== Voicemail: Allow SMTP Email Port Different Than tcp/25 ===


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|[http://mantis.innovaphone.com/view.php?id=98683 98683]
|}
|}
This SNMP trap is no longer necessary. Meanwhile it is covered more consistently by "innoDiagAlarm" and "innoDiagAlarmClear".<br/><!--<br/>gk.cpp/.h<br/>--><!---->
eg in email.xml: ..smtp.foo.bar:888..<br/><!--<br/>smpt_mta.cpp<br/>--><!---->
      
      
=== Linux: GUI removed from IP1060 ===
=== phone: directory search highlights first matching entry if the search expression contains any non numeric digit ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=86547 86547]
|[http://mantis.innovaphone.com/view.php?id=98806 98806]
|}
|}
The IP1060 does not support the Linux Application Platform. The GUI is removed now. Also a trap is fixed, if somebody tries to enable Linux.<br/><br/><!--<br/>Changed files:<br/>linux.xml<br/>ip6010/platform/config.h<br/>ip6010/platform/ip6010.cpp<br/>ip6010/platform/tabs.xml<br/>ip6000/platform/ip6000.cpp<br/>--><!---->
To prevent unintended dialing of a directory entry starting with numeric digits the search expression was checked if it consists of dialable digits (0-9*#,) only.<br/>In this case the first matching entry was not automatically highlighted (activated) so that the input (number) could be dialled by going off-hook. <br/>Now the check includes the numeric digits (0-9) only.   <!---->
      
      
=== Voicemail: &lt;pbx-disc&gt; failed sometimes ===
=== SIP: New interop tweak /accept_any_reg_interval ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=86569 86569]
|[http://mantis.innovaphone.com/view.php?id=98887 98887]
|}
|}
In case of unconnected calls that were subject of a prior <pbx-fwd><br/><!--<br/>pbx_vm.cpp<br/>--><!---->
For endpoints that do not read the expires value from 200/OK response to REGISTER.<br/>E.g. "User-Agent: TRBOnet.Enterprise"<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== phone: ip222, ip232: handset/handsfree speaker was not switched off when changing over to headset mode ===
=== phone: ip222,ip232: support Plantronics Blackwire C320, Blackwire C720, Voyager Legend UC, Calisto 620 ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=86706 86706]
|[http://mantis.innovaphone.com/view.php?id=99112 99112]
|}
|}
When the headset talk-key or the phone headset-control-key is pressed while the phone is in handset or handsfree mode the phone changes to headset mode, i.e. headset micro and speaker are activated. The handset or handsfree speaker should be switched off then.<br/><br/><!-- ac_codec3.cpp --><!---->
<!-- keys ip222.cpp --><!---->
   
== Bug Fixes ==
 
 
      
      
=== phone_orchid: pressing the speaker key while in headset mode did not switch over to handsfree mode ===
=== IPVA: Unused ETH1 Could Cause Out-Of-Memory Situation ===


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|[http://mantis.innovaphone.com/view.php?id=98179 98179]
|}
|}
When the speaker key is pressed while the phone is in headset mode the headset should be deactivated and the phone should enter handsfree mode.<br/><br/><!-- app_ctl.cpp --><!---->
Outgoing packets could queue up without ever getting purged.<br/><!--<br/>vlance_drv.cpp<br/>vlance_drv_s.xsl<br/>--><!---->
      
      
= V9 Hotfix 17 (9061152) =
=== SIP: Trap - not checking array boiundaries ===
Changes included in Version 9 hotfix17
[http://mantis.innovaphone.com/view.php?id=86708 Definition]
 
== New Features ==
 
 
   
=== Voicemail: &lt;pbx-query-obj type="filter"/&gt; couldn't read template distributed filter ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=86459 86459]
|[http://mantis.innovaphone.com/view.php?id=98219 98219]
|}
|}
Switched to a different internal api method<br/><!--<br/>shelved:<br/>pbx_vm.cpp<br/>--><!---->
Trap - not checking array boiundaries when processing User-Agent header.<br/><br/><!--<br/>sipmsg.cpp<br/>--><!---->
      
      
=== AD Replication: Refactoring Poll-Timer-based Replication ===
=== SIP: STUN not working ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=86476 86476]
|[http://mantis.innovaphone.com/view.php?id=98223 98223]
|}
|}
-Ridded usnChanged filter term<br/>-Added two status messages<br/><!--<br/>adrep.cpp<br/>--><!---->
STUN not working if STUN server IP address is configured.<br/>No problem if if STUN domain name is configured.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== PBX Mobility: Take precaution against hanging calls because of misbehaving mobile phone or client ===
=== phone: ip222,ip232: USB connection sometimes lost until reboot ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=86813 86813]
|[http://mantis.innovaphone.com/view.php?id=98290 98290]
|}
|}
Use a 20s timer to terminate any call to the mobility object if no internal call is initiated<br/><!--<br/>pbx_mobility.cpp<br/>pbx_mobility.h<br/>--><!---->
Observerved with with headsets from various manufacturers. Most probably caused by strong electrostatic discharges to the USB connection cable. In such case also unplugging/plugging of the headset was not noticed by the driver anymore.<br/><br/><!-- ohci.cpp ohci.h --><!---->
      
      
=== IPVA: Query Physical Link Status. ===
=== Logging: "Alarm and Event Forward Server" address could not be changed anymore once configured ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=86937 86937]
|[http://mantis.innovaphone.com/view.php?id=98457 98457]
|}
|}
-The physical link status wasn't evaluated so far. Now it is.<br/>-Added statistics page 'Interfaces/EthX/Statistics'<br/><!--<br/>vlance/vlance_drv.cpp<br/>vlance/vlance_drv_s.xls<br/>platform/interfaces.xml<br/>--><br/><!---->
<!-- logging.xsl --><!---->
      
      
=== IP TOS value is now traced with Wireshark ===
=== LDAP Replication: ASN.1 Decoding Failed For More Than 100 Attributes Per Object ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=87025 87025]
|[http://mantis.innovaphone.com/view.php?id=98468 98468]
|}
|}
This value is now correctly traced within IP4 UDP/TCP traces.<br/><br/><!--<br/>debug.cpp/h<br/>ip_user.cpp/h<br/>tcp.cpp/h<br/>udp.cpp/h<br/>ip4_user.h<br/>ipproc.cpp/h<br/>tcp4.cpp/h<br/>udp4.cpp/h<br/>ip6_icmpsock.cpp/h<br/>test_debug.cpp/h<br/>test_tcp.cpp/h<br/>test_udp.cpp/h<br/>--><!---->
Happend with 100 attributes in a PBX stored phone configuration.<br/><!--<br/>ldapsrv.cpp<br/>ldapdir.cpp<br/>--><!---->
      
      
=== PBX Trunk: New destination for rejected calls ===
=== Corrupt trap buffer when tracing UART messages ===


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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=87151 87151]
|[http://mantis.innovaphone.com/view.php?id=98533 98533]
|}
|}
Calls which are rejected (busy after alert), can be redirected to a configurable destination.<br/><!--<br/>pbx_trunk.cpp<br/>pbx_trunk.h<br/>pbx_edit_trunk.xsl<br/>--><!---->
<!-- uart_drv.cpp --><!---->
      
      
=== SIP: Offer media encryption as separate media description ===
=== IP232,IP222: Confirming blind transfer with redial key did not work ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=87152 87152]
|[http://mantis.innovaphone.com/view.php?id=98591 98591]
|}
|}
Interop issue.<br/>New config file option "config change SIP /separate-encryption".<br/><br/>Offer:<br/>\tv=0<br/>\to=- 10 1 IN IP4 172.16.16.156<br/>\ts=-<br/>\tc=IN IP4 172.16.16.156<br/>\tt=0 0<br/>\tm=audio 16390 RTP/AVP 9 8 0 18 101 13<br/>\tc=IN IP4 172.16.16.156<br/>\ta=rtpmap:101 telephone-event/8000<br/>\ta=fmtp:18 annexa=yes<br/>\ta=fmtp:18 annexb=no<br/>\ta=fmtp:101 0-15<br/>\ta=ptime:30<br/>\ta=silenceSupp:off - - - -<br/>\ta=sendrecv<br/>\tm=audio 16390 RTP/SAVP 9 8 0 18 101 13<br/>\tc=IN IP4 172.16.16.156<br/>\ta=rtpmap:101 telephone-event/8000<br/>\ta=fmtp:18 annexa=yes<br/>\ta=fmtp:18 annexb=no<br/>\ta=fmtp:101 0-15<br/>\ta=ptime:30<br/>\ta=silenceSupp:off - - - -<br/>\ta=sendrecv<br/>\ta=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:VoGZgKwImVwTkJg4jBeYFBafl/CyJpfMX66WqDMZ<br/><br/>Instead of:<br/>\tv=0<br/>\to=- 9 1 IN IP4 172.16.16.156<br/>\ts=-<br/>\tc=IN IP4 172.16.16.156<br/>\tt=0 0<br/>\tm=audio 16386 RTP/SAVP 9 8 0 18 101 13<br/>\tc=IN IP4 172.16.16.156<br/>\ta=rtpmap:101 telephone-event/8000<br/>\ta=fmtp:18 annexa=yes<br/>\ta=fmtp:18 annexb=no<br/>\ta=fmtp:101 0-15<br/>\ta=ptime:30<br/>\ta=silenceSupp:off - - - -<br/>\ta=sendrecv<br/>\ta=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:mDPFKhNdhm6fhkeyPbAI9uzZyTmtA4t9fy1MwXiD<br/><br/><br/><!--<br/>sip.ccp/h<br/>--><!---->
Short user guide:<br/><br/> Press 'redial' before accepting the call, enter telephone<br/> number for the call diversion and confirm by pressing 'redial' again.<br/><br/><!--<br/>app_ctl.cpp<br/>--><!---->
      
      
=== IP-DECT: DECT security (4) ===
=== phone: ip222,ip232: USB headset media connection lost after a release received from a remote conference peer ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=87157 87157]
|[http://mantis.innovaphone.com/view.php?id=98600 98600]
|}
|}
Some changes in preparation for the new feature "DECT security". This feature will be available later (IP1202).<br/><br/><!--<br/>Changed file:<br/>dectusers.cpp<br/>--><!---->
Happened only on a release of the call which was the active call when the conference was established. The remaining VOIP connection was OK but the media stream was not passed from/to headset anymore.<br/><br/><!-- ac_phone3.cpp ac_codec3.cpp --> <!---->
      
      
=== SIP: Do registration refresh more ahead of expiration ===
=== IP241: DHSG Headset messages are sometimes wrong ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=87381 87381]
|[http://mantis.innovaphone.com/view.php?id=98642 98642]
|}
|}
Instead of refreshing 2 secs before registration expiration,<br/>do the refresh after 98,33% of the registration ttl.<br/><br/>TTL=120secs -> Refresh 2secs before expiration<br/>TTL=3600secs -> Refresh 60secs before expiration<br/><br/><!--<br/>sip.cpp<br/>--><!---->
<!-- orchid_lcd.cpp orchid_lcd.h config.h --><br/><br/>Beim Neuladen des Sequencers bei Änderung der LCD-Helligkeit kommt der UART-Takt gelegentlich zu schnell. Besser noch wäre 2 sequencerprogramme zu definiere, aber wie das geht ist nicht im Orchid/Titan Usermanual nicht offensichtlich --><!---->
      
      
=== phone: Modification of function key on phone can be disabled for any single key ===
=== SIP: SDP version not increased when answering an offer where only media-mode has changed ===


{|
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|-
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=87405 87405]
|[http://mantis.innovaphone.com/view.php?id=98739 98739]
|}
|}
A "Disable Modification on Phone" checkmark will be provided in the edit menu for  each key. If checked the key cannot be edited on the phone anymore.<br/>This mechanism works in addition to the phone local key type mask set via<br/>   "Phone/Protect/Function keys not modifiable on the phone"<br/>A key of a type NOT marked as ``not editable´´ in this mask can be made ``not editable´´ by setting the above mentioned checkmark<br/>A key of a type marked as ``not editable´´ remmains not editable, independent of the checkmark setting.<br/><br/><!-- phone_config.cpp phone_config.h phone_edit.cpp phone_user_func.cpp fkey_edit_*.xsl --><br/><br/><!---->
If remote side changes from 'sendrecv' to 'inactive'<br/>the SDP answer follows this change of media-mode,<br/>but SDP version was not increased.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== phone: "Spare" function key to reserve key positions for administrative purposes ===
=== Gatway: Configuration of Blockdial Timeout at Routes did not work ===


{|
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|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=87406 87406]
|[http://mantis.innovaphone.com/view.php?id=98748 98748]
|}
|}
This new key permits to reserve key positions for later definition by administrator.<br/><br/><!-- phone.mak phone_config.h phone_config.cpp phone_edit.cpp fkey_edit_spare.xsl app_disp.cpp app_fkey.cpp phone_user_func.cpp phone_strings_utf8.h phone_strings_latin1.h phone_text_tables.h --><!---->
Wrong value was calculated, if multiple maps were used in a single route blockdial timeout configuration of a map was lost, when another map was configured.<br/><!--<br/>gk.cpp<br/>--><!---->
      
      
=== IP-DECT: Remote hold in conference mode ===
=== enabled state of an external directory configured via a PBX config template was lost in some cases, ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=87487 87487]
|[http://mantis.innovaphone.com/view.php?id=98816 98816]
|}
|}
If a hold notify message is received from a remote party and the conference mode is active, now the message is forwarded to the conference unit. This prevents the music on hold in conference calls. The state is also shown in the radio call list.<br/><br/><!--<br/>Changed files:<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp<br/>dectradio_calls.xsl<br/>--><br/><!---->
This did happen for example when<br/>- a second phone was registered to the same PBX user (twin phone)<br/>- a "Phone/Reset/Reset User Specific Configuration" was done via the phones WEB GUI (but not when this was done via the PBX GUI)<br/><br/><!-- phone_edit.cpp phone_user.cpp phone_user_ui.cpp --><!---->
      
      
=== Webdav: Write information into trace if DELETE fails because file is in open state ===
=== SIP: Trap when sending &lt;dialog-info&gt; ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=87488 87488]
|[http://mantis.innovaphone.com/view.php?id=98902 98902]
|}
|}
Write information about Webdav session holding open file handle.<br/><br/><!--<br/>http.h<br/>servlet.h<br/>servlet_webdav.cpp/h<br/>--><!---->
Trap when interworking group-indications into dialog-info.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== H.323: Better error handling in case of incompatible SRTP parameters ===
=== Memory Leak when deleting voicemails ===


{|
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|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=87624 87624]
|[http://mantis.innovaphone.com/view.php?id=98929 98929]
|}
|}
Was needed for DECT OEM.<br/><!--<br/>h323ch.cpp<br/>--><!---->
A list wasn't cleaned<br/><!--<br/>pbx_vm.cpp<br/>vm_script_if.cpp<br/>test\\9.00\\pbx\\vm\\vm-box-to-box\\vm\\cf-files\\vm-listfiles.xml<br/>--><!---->
      
      
=== PBX: No CDR was generated for enblock call, which was rejected because of filter config ===
=== IP222 IP232: Noise in the microphone at 6400Hz ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=87869 87869]
|[http://mantis.innovaphone.com/view.php?id=98941 98941]
|}
|}
A CDR with cause 52 (outgoing call barred) is generated<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_api.h<br/>--><!---->
.<br/><!-- ac_codec.cpp platform/filter/ip222_mic_*.* --><!---->
      
      
=== Voicemail: Add &lt;pbx-getcallinfo out-leg2-orig="..."&gt; ===
=== IP222 IP232 IP241: Codec register debugs added ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="red">Assigned</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=87880 87880]
|[http://mantis.innovaphone.com/view.php?id=98972 98972]
|}
|}
Allows to query the divertingLegInformation2.orignalCalled number<!---->
Codec register debugs are enabled with dsp  trace.<br/>Codec register are dumped after changes to analyse Manits 97903 <br/><!-- ac_codec3.cpp --><!---->
      
      
=== IP-DECT: Fault logging for Master module ===
=== PBX: No config updates were sent to the phones, when selection of templates was changed ===


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|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=88005 88005]
|[http://mantis.innovaphone.com/view.php?id=98983 98983]
|}
|}
Call and channel fault logging for the IP-DECT Master module are added.<br/><br/><!--<br/>Changed files:<br/>signal.h<br/>signal.cpp<br/>log_if.h<br/>log_if.cpp<br/>dectmaster.h<br/>dectmaster.cpp<br/>fault_dect.xsl<br/>--><!---->
When the config of the template itself was changed an update was sent, but if it was changed which templates were used on a user object, no update was sent.<br/><!--<br/>pbx.cpp<br/>--><!---->
      
      
=== Voicemail: &lt;pbx-prepcallinfo leg2=".." leg2-name=".."/&gt; ===
=== Phone: Could not configure fkey labels containing single quotation mark ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=88351 88351]
|[http://mantis.innovaphone.com/view.php?id=98986 98986]
|}
|}
New attributes allow to prepare <pbx-fwd> in such, that the supplementary service divertingLegInformation2 is going to be sent.<!---->
Could not configure fkey labels containing single quotation mark.<br/><br/><!--<br/>phone_edit.cpp<br/>--><!---->
      
      
=== PBX: Allow configuration of default presence/dialog-info visibility for group members ===
=== myPBX: Missed calls for mobility calls accepted somewhere else ===


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|Id
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|[http://mantis.innovaphone.com/view.php?id=98995 98995]
|}
|}
Active group members got full presence/dialog-info because this matched the visibility be group-indications. However this is not desired always, so it can now be configured to restrict this.<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_general.xsl<br/>config_options.cpp<br/>config_options.h<br/>--><!---->
A cause code was missing in CDRs generated by mobility<br/><!--<br/>pbx_mobility.cpp<br/>--><!---->
      
      
=== IP-DECT: Unused OEM modules removed ===
=== SIP: Switch-over to t38 did not work in one configuration scenario ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=88413 88413]
|[http://mantis.innovaphone.com/view.php?id=99130 99130]
|}
|}
Unused OEM modules has been removed now.<br/><br/><!--<br/>Changed files:<br/>config.h<br/>dect.cpp<br/>--><!---->
Interworking of SIP and H.323.<br/>Switch-over to t38 did not work if both Gateway interfaces were configured to media-relay with exclusive audio codec.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
      
      
=== Gateway: Make interop flag "Ack incoming call" configurable on UI ===
=== phone: ip222,ip232: ignore  HID function of USB headset charging cables ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="orange">To-decide</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=88499 88499]
|[http://mantis.innovaphone.com/view.php?id=99133 99133]
|}
|}
This was a hidden interop flag, but was now needed multiple times, so it is more efficient to have it configurable on the UI<br/><!--<br/>gk.cpp<br/>gk.h<br/>relay_edit_phys.xsl<br/>--><!---->
Some USB headset charging cables present a HID function as long as the headset is connected to the cable (probably used for headset firmware updates).<br/>Dependent on the enumeration sequence the HID function of the cable could hide the HID function of the headset when the headset base station (or the bluetooth or DECT dongle) is plugged in paralll to the phone.<br/><br/><!-- usb_host_if.h ohci.h ohci.cpp usb_hid.cpp usb_audio.cpp --><!---->
      
      
=== IP-DECT: Logging for handover calls ===
=== IP22 IP24 IP28 IP302 IP305: ASSERT on DSP queue overrun added (2) ===


{|
{|
|Status
|Status
|<font><font color="green">Closed</font></font>
|<font><font color="red">Assigned</font></font>
|-
|-
|Id
|Id
|[http://mantis.innovaphone.com/view.php?id=88536 88536]
|[http://mantis.innovaphone.com/view.php?id=99155 99155]
|}
|}
Logging events for IP-DECT handover calls added.<br/><br/><!--<br/>Changed files:<br/>dectradio_call.h<br/>dectradio_call.cpp<br/>dectradio_mastercall.h<br/>dectradio_dectcall.h<br/>dectradio_dectcall.cpp<br/>dectradio_handover_out.h<br/>dectradio_handover_out.cpp<br/>dectradio_handover_in.cpp<br/>--><!---->
<!--  ac494.cpp  --><!---->
   
== Bug Fixes ==
 
 
   
=== make update script parsing more tolerant to suspicious line ends ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=84349 84349]
|}
Some WEBDAV tools garble line end when a text file is stored after editing. Last seen \\r\\r<br/> instead of \\r<br/>. Any sequence consisting only of \\r <br/> chars should be read as one line end because empty lines have no meaning in an update script.<br/><br/><!-- update.cpp update.h -->  <!---->
   
=== SIP: Do not interwork holdNotific and retrieveNotific while on hold ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86736 86736]
|}
Interworking problem with VOXTRON/XCAPI<br/>Do not interwork holdNotific and retrieveNotific after remoteHold.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== Disable LDAP Server When Erasing Flash Directory ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86763 86763]
|}
During an upload of a complete configuration the command "mod cmd FLASHDIR0 erase-all" will erase all flash directory content. Replication clients are going to receive nil-responses making them assume a certain entry does no longer exist.<br/><br/>Replication clients are now barred from accessing the LDAP server as long as the box didn't process the post-upload reset.<br/><!--<br/>interface/ldapapi.h<br/>interface/ldapapi.cpp<br/>service/ldap/ldapsrv.h<br/>service/ldap/ldapsrv.cpp<br/>service/ldap/flashdir.h<br/>service/ldap/flashdir.cpp<br/>--><br/><!---->
   
=== IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: fax bad signal quality events are sent on good fax connections ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86788 86788]
|}
Sometimes during fax transfer a bad signal quality (e.g.50) is reported, even if the connection is good. This happens during the TCF phase, in the image phase the signal quality is fine ( e.g. 3) <br/><br/><!-- ac_fax3.cpp --><br/><!---->
   
=== Mis-typing on Maintenance/Diagnostics/Counters ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86812 86812]
|}
Show "kbit/s" instead of "kBbit/s".<br/><br/><!--<br/>log_counter.xsl<br/>--><!---->
   
=== Gateway: #11 could not be dialed on analog interfaces with feature codes enabled ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86819 86819]
|}
This is a featiure code used on DECT systems and it was not disabled on analog interfaces<br/><!--<br/>relayfty.cpp<br/>dtmffty.cpp<br/>--><!---->
   
=== Gateway: Potential trap when recording is configured and a transfer happens ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86837 86837]
|}
The problem is with a transfer, which is executed in the Gateway, not the PBX<br/><!--<br/>relay.cpp<br/>--><!---->
   
=== PBX: XML Export/Import did not work, if DTMF Feature objects present ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86870 86870]
|}
The automatically generated user objects caused a problem. This could result in a config that caused the PBX to restart in a loop. The export/import was fixed and the PBX does not restart because of the corrupt config any more.<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_api.h<br/>pbx_admin.cpp<br/>--><!---->
   
=== PBX: Trap if a Hold was attempted for a call without media ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86874 86874]
|}
Could be caused by a misbehaving application or voip device<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== PBX: Name beginning with '*' caused problems with SOAP applications ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86882 86882]
|}
A search for such a name was treated as wildcard search for all users. The name '*' is now a reserved name, which cannot be used for an object.<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== Linux: IP0010 available again ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86903 86903]
|}
With V9 hotfix 16 Linux can not be started on the IP0010. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>linux.cpp<br/>--><br/><!---->
   
=== phone: call diversion override via indirect dialing could not be disabled ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86944 86944]
|}
In the indirect dialing screen the right arrow key opens a menu with different options how to place the call.<br/>"Dial - No Diversion" ssets up a call which will ignore the diversions active on the target phone. This menu item can be supressed via the "Fine grained function locking" bit PHONE_LOCK_DIVERSION_OVERRIDE 0x04000000<br/><br/><!-- phone_dir_ui.cpp phone_config.h --><!---->
   
=== PBX Waiting: CFNR with number filter did not work ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86959 86959]
|}
The CFNR was not executed<!---->
   
=== PBX Conference: Calling Party was missing in the CDRs for calls to conference members ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86977 86977]
|}
Only the conference object itself was present as forwarding party<br/><!--<br/>pbx_bc_conf.cpp<br/>--><!---->
   
=== SIP: Rare trap when cancelling call ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86994 86994]
|}
When call abort interferes with re-routing in gateway application.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== PBX: Replication from a dyn PBX was not possible ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87065 87065]
|}
Configuration of dyn PBX id now possible for replication<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_general.xml<br/>--><!---->
   
=== TLS: Ignore incoming HelloRequest messages ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87091 87091]
|}
When receiving a HelloRequest the TLS client dopped the connection and sent an "Unexpected message" alert.<br/><br/>Now the HelloRequest is ignored and an "No renegotiation" warning is sent.<br/><br/><!--<br/>tls.cpp<br/>unit-tests/tls/basic/520-data-hellorequest.xml<br/>--><!---->
   
=== PBX: Call, which was rejected with busy, because of CF loop, could not be forwarded by Trunk Busy destination ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87150 87150]
|}
Such a call was always disconnected<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: DSP Update to Version 680.05 ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87179 87179]
|}
The jitter buffer performance has been improved.<br/>The fax/modem bypass performance has been improved.<br/>IP28 firmware size was reduced ( unused code is not linked ).<br/>IP28 G279 didnt work with more than 4 channels.<br/><br/><!-- <br/>DSP code details at \\\\innovaphone.sifi\\dfs\\Dokumente\\Hardware\\G7xx\\Audiocodes\\firmware\\firmware_ac49x\\680.05<br/><br/>ac49x_680/*  ac_494e.cpp ip24.mak ip6010.mak --><!---->
   
=== IP222 IP232 IP241: Disable PC port didnt work ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87250 87250]
|}
<!-- orchid_drv.cpp --><!---->
   
=== IP-DECT: Login feature with disabled local R-key handling ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87269 87269]
|}
The login feature is fixed now if the local R-key handling is disabled.<br/><br/><!--<br/>Changed file:<br/>dectradio_call.cpp<br/>--><!---->
   
=== Webdav: DELETE may fail if file has been opened for reading before ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87288 87288]
|}
If reading (GET) stops but HTTP session remains open, the file remains in state 'open'<br/>and subsequent DELETE request fails with "500 Internal Server Error".<br/>Close file and re-try to delete.<br/><br/><!--<br/>servlet_webdav.cpp/h<br/>--><!---->
   
=== HTTP: Chunked transfer fails if the chunk header is not in a single packet ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87292 87292]
|}
<!-- http.cpp --><!---->
   
=== PBX: Conference and hanging VM script ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87312 87312]
|}
Wrong media initializing in the BC Conference object causes a hanging VM script if a file should be played. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>pbx_bc_conf.cpp<br/>--><!---->
   
=== IP-DECT: No voice with early handover ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87504 87504]
|}
This fixes outgoing calls without voice if early handover is done before the media channel is initialized.<br/><br/><!--<br/>Changed files:<br/>dectmaster.cpp<br/>dectradio_call.cpp<br/>dectradio_dectcall.cpp<br/>--><br/><!---->
   
=== phone: ip222, ip232: multiple reports indicating offhook were misinterpreted and could drop a call ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87506 87506]
|}
Wireless USB headsets may send more than one report indicating headset offhook state in conjunction with different wireless link states. If the interval between the first and the second indication was very short (8 ms) the second indication was misinterpreted and the just setup call was dropped (observed with a Jabra PRO 930 after plugin). <!---->
   
=== SIP: Fix for Contact-URI in 200/OK for SUBSCRIBE ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87507 87507]
|}
Fix for Contact-URI in 200/OK for SUBSCRIBE and NOTIFY requests.<br/>May contain double port attribue:<br/><br/>SUBSCRIBE sip:8011@172.20.11.53:2053 SIP/2.0<br/><br/>Contact: <sip:8011@172.20.11.53:2053:2053;maddr=172.20.11.53;transport=UDP><br/><br/><!--<br/>sip.cpp<br/>--><!---->
''Status:''
Fixed in 10.00 and 9.00
   
=== IP-DECT: Disturbances GUI info change ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87522 87522]
|}
The text of the GUI page disturbances is changed.<br/><br/><!--<br/>New file:<br/>_inn_disturbances.xsl<br/>Changed file:<br/>httpfiles.mak<br/>--><!---->
   
=== IP-DECT: Web UI  info page, version and release state ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87538 87538]
|}
Now the version and the release state are shown on the web UI info page of the IP1202.<br/><br/><!--<br/>Changed file:<br/>dect_comcerto.cpp<br/>--><!---->
   
=== PBX: Blind transfer to WQ by IP-DECT caused hanging calls ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87542 87542]
|}
The call-leg to the transfering phone was not cleared by the PBX, so if the phone did not clear this call, it was hanging for ever. Other phones clear such a call after a timeout, but this is only a workaround, the call must be cleared by the PBX<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== H.323: Problem sending real big signaling messages ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87543 87543]
|}
Happened with a configuration with more the 60 Join Group function keys on the phone. The message was not sent and the operation failed.<br/><!--<br/>h323sig.cpp<br/>--><!---->
   
=== PBX-SOAP: UserCall on WQ did not take "Send Number" into account and call was not marked as internal ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87559 87559]
|}
Calls should be send with "Send Number" as source if configured<br/><!--<br/>pbx_xml.cpp<br/>--><!---->
   
=== PBX: CFB configured at PBX object did not patch number correctly ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87581 87581]
|}
In case the called endpoint was in different node then PBX object, the number did not contain all the needed prefixes<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== SIP: Trap when logging out dect user ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87606 87606]
|}
Trap when logging out dect user.<br/><br/>0:0287:977:1 - MASTER_EP-SIG.0 default(948f3e18): serial_event(100) src=DECTMASTER-CALL.0 mod=SIP<br/>0:0287:977:1 - Assertion failed line 790 in common/os/os.cpp, object deleted<br/>0:0287:977:2 - assert-ep 94004474  called from 94055498<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
   
=== PBX: Called number was missing in CDRs for calls to busy endpoint ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87620 87620]
|}
In case there was no other event then rel-to/from and the call was dialed with overlap sending<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== SIP: Handset type information also in Subscribe User-Agent ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87621 87621]
|}
Handset type information also in Subscribe User-Agent as sent in REGISTER request.<br/><br/><!--<br/>siptrans.cpp/h<br/>sip.cpp/h<br/>--><br/><!---->
   
=== H.323: Media Negotiation did not work for Hold/Retrieve after a DECT conference ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87635 87635]
|}
No media after retrieve<br/><!--<br/>h323ch.cpp<br/>--><!---->
   
=== PBX: Conference trap ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87662 87662]
|}
A trap in the BC conference PBX object is fixed.<br/><br/><!--<br/>Changed file:<br/>pbx_bc_conf.cpp<br/>--><!---->
   
=== PBX Waiting: Missing ringback on call forward after announcement ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87674 87674]
|}
This was a collateral damage of<br/><br/>fix: #81370: PBX Waiting: Call state shows "Disconnecting" after switch from announcement 1 to announcement 2<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
   
=== Gateway: Routing on blind transfer call starts from wrong IF (undo of  #86386) ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87678 87678]
|}
Rollback of #86386: Gateway: Routing on blind transfer call starts from wrong IF<br/><br/>There are other installations where the old behaviour is the expected behaviour.<br/><br/><!--<br/>relay.cpp<br/>--><!---->
   
=== PBX Waiting: DTMF overlap dialing or blind transfer to same Waiting object was rejected with busy ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87681 87681]
|}
Even if this was caused by a CFB or CFU on the dialed destination<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
   
=== IP-DECT: No idle display update with own message ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87715 87715]
|}
A message with the same sender and receiver name overwrites the handset's idle display. This feature is removed now.<br/><br/><!--<br/>Changed file:<br/>dect_comcerto/platform/config.h<br/>--><!---->
   
=== Voicemail: &lt;pbx-fwd&gt;, Forward Timer Didn't Fire For Connected Calls ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87717 87717]
|}
Voicemail: <pbx-fwd>, Forward Timer Didn't Fire For Connected Calls<br/><!-<br/>pbx_vm.cpp<br/>--><!---->
   
=== IP-DECT: Trap with rejected handovers ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87730 87730]
|}
A trap occurs in the rare situation if the handover is rejected by the radio.<br/><br/><!--<br/>Changed file:<br/>dectradio.cpp<br/>--><!---->
   
=== H.323: Q.931 User Info could not be sent in incoming call proceeding state ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87740 87740]
|}
This is used by the Fax interface<br/><!--<br/>h323_tbl.tbl<br/>--><!---->
   
=== SIP: Parsing of multiple Allow lines was not implemented ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87753 87753]
|}
Now all Allow lines are parsed (not only first one).<br/><br/><!--<br/>sipmsg.cpp/h<br/>--><!---->
   
=== syslog packets were sent with the initially assigned source address even after a new address had been assigned ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87754 87754]
|}
When a device runs as DHCP client the IP-adress assigned to an interface may change either because the DHCP-Server rejects a renew request and provides a new lease or because a WLAN device enters another network.<br/>An IP-address change may also happen when the DHCP mode of a device is changed from 'disabled' to 'client' without reboot.<br/>The source address of syslog packets does reflect such changes now. <br/><br/><!-- logging.cpp logging.h -->  <br/><!---->
   
=== Linux: Trap of IP810 if Linux is started ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87763 87763]
|}
The IP810 with the V9 hotfix 16 firmware traps if Linux is started. Please do not use this firmware. It is fixed again now.<br/><br/><!--<br/>Changed file:<br/>config.h<br/>--><!---->
   
=== PBX: CFNR to same node but different PBX failed on Slave ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87772 87772]
|}
The PBX routing did not work correctly in this case<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== PBX administration: Viewer could change phone configuration of users ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87774 87774]
|}
<!-- pbx_phone.cpp pbx_phone.h --><!---->
   
=== SIP: Don't try to dns-resolve IP adresses ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87781 87781]
|}
Don't try to dns-resolve IP adresses.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== phone: Function key options which need to be enabled in Phone/Userx/Preferences could be edited on phone although disabled ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87792 87792]
|}
Partner-Intrude: hide when "Phone/Userx/Preferences/Enable Call Intrusion" is not checked  <br/>Dial-Announce: hide when "Phone/Userx/Preferences/Announcement Calls/Outgoing/Allow" is not checked<br/><br/><!-- phone_user_func.cpp phone_user_config.h --><br/><!---->
   
=== SIP: Error handling huge INVITE requests ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87811 87811]
|}
Failed to compose provisional and final response.<br/>Lots of error messages in trace and truncated responses.<br/><br/>  ERROR: SIP message buffer (2049) exceeded! (40,35,941b0c90)<br/><br/><!--<br/>siptrans.cpp<br/>--><!---->
   
=== Gateway: MOH from call on hold in CONF interface calls ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87823 87823]
|}
If a new call joins the conference or a call on hold retrieves the conference, and there are calls on hold in the conference, the music on hold can be heard in the new or retrieving calls. Now this is fixed.<br/>This affects all devices with a CONF interface, but not the IP800 and the IP305.<br/><br/><!--<br/>Changed files:<br/>adsp2191_drv.h<br/>adsp2191_drv.cpp<br/>--><!---->
   
=== IP241,IP222,IP232: Presence note may is not enough truncated on 'presence' fkey ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87825 87825]
|}
Presence note may is not enough truncated on 'presence' fkey.<br/>Esp. when there's also an icon to be displayed.<br/><br/><!--<br/>forms_color.cpp<br/>--><!---->
   
=== PBX Trunk: If a call was forked to a trunk with no-name option, name was removed from the original call as well ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87843 87843]
|}
A global flag was set on the incoming call, which was wrong.<br/><!--<br/>pbx_trunk.cpp<br/>--><!---->
   
=== IP230 IP240: Handset receiver volume increased ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87878 87878]
|}
receiver volume adapted to changed equalizer settings from fix #80403<br/><br/><!-- inca_afe.cpp --> <br/> <!---->
   
=== phone: ip222, ip232: better handling of spurios headset disconnects ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87908 87908]
|}
Sometimes USB headsets get disconnected from USB port because of certain electric pulses. To overcome this problem the headset port is reset and the media stream routed to the handset. If the headset comes up (logical plugged) again in a reasonable time the media stream is routed to the headset again. Otherwise the media stream remains on the handset and the call can be continued by taking off the handset.<br/><br/><!-- keys_ip222.cpp ohci.cpp --><br/><!---->
   
=== PBX Trunk: Diverting as Calling Feature should replace the name as well ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87913 87913]
|}
With this feature the Trunk object uses a Diverting Number as calling party number. But not only the number, but also the Name and Name Id should be replaced<br/><!--<br/>pbx_trunk.cpp<br/>pbx.cpp<br/>--><!---->
   
=== Phones: Allow lcd_dump.bmp to be retrieved with viewer credentials ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87945 87945]
|}
Allow lcd_dump.bmp to be retrieved with viewer credentials.<br/><br/><!--<br/>lcd_dump.bmp<br/>--><br/><!---->
   
=== IP-DECT: Wrong radio list after MAC-alias change ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87950 87950]
|}
The radio list can be wrong after the MAC-alias change. This fixes a bug of the feature "MAC-alias change of OEM device" (#86047). This is only relevant for OEM devices.<br/><br/><!--<br/>Changed file:<br/>dectmaster.cpp<br/>--><!---->
   
=== SIP: Keep Contact-URI when registering via TCP ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=87969 87969]
|}
Keep Contact-URI when registering via TCP, even if rport is present in 200/OK for REGISTER<br/><br/><!--<br/>sip.cpp<br/>--><br/><!---->
   
=== PBX-SOAP: UserCall with 'cn' as destination to a user in different node did not work ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88008 88008]
|}
The call is done using the number, but the number needs to be adjusted according to the nodes<br/><!--<br/>pbx_xml.cpp<br/>--><!---->
   
=== OS: Quota mechanism did not work for CPU time consumed by timer handling ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88034 88034]
|}
This could result in a MAX_BUSY_TICK restart if many timer expired at the same time<br/><!--<br/>os.cpp<br/>command.cpp (for test only)<br/>command.h (for test only)<br/>--><!---->
   
=== HTTP: Chunked transfer sporadic fails with webdav ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88078 88078]
|}
<!-- http.cpp --><!---->
   
=== IP6000 IP2000: Webinterface for ETH1 Link setting didnt work ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88137 88137]
|}
..<br/><!-- interfaces.xml --><!---->
   
=== SIP: Decoding problem on application/simple-message-summary ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88188 88188]
|}
application/simple-message-summary may get decoded wrongly.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== Voicemail: Possibility To Switch Off An Internal Automatism ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88246 88246]
|}
An internal automatism could lead to VM-calls getting disconnected after 15s.<br/>The new URI variable "$_divconn=false" turns off auto-connection for diverted/transferred calls:<br/>http://wiki.innovaphone.com/index.php?title=Howto:Configure_the_innovaphone_Voicemail#URL_Query_String_Variables<br/><!--<br/>pbx_vm.cpp/h<br/>--><!---->
   
=== SIP: DNS priority value not honored always ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88248 88248]
|}
DNS priority value not honored, if SRV query returns names without address in additional records.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== SIP: Do not take "9564+4631559300" as E.164 number ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88254 88254]
|}
Do not take "9564+4631559300" as E.164 number.<br/><br/><!--<br/>sip.cpp<br/>siputil.cpp/h<br/>--><!---->
   
=== PBX Boolean: Unpredictable behaviour if more the 16 times are entered ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88263 88263]
|}
There was a limit of 16 times, which was not checked everywhere. The limit is now 32 and checked, so that no more of 32 times can be entered.<br/><!--<br/>pbx_bool.cpp<br/>pbx_bool.h<br/>--><!---->
   
=== IP222 IP232 IP241: Restart on Jitter buffer overrun ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88312 88312]
|}
..<br/><!-- ac494e.cpp --><!---->
   
=== SIP: Fix for auto-answer handling ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88334 88334]
|}
Fix for auto-answer handling.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== IP-DECT: SIP/Intop - rejected call transfer by target ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88356 88356]
|}
If the call transfer target rejects the call in ringing state, no fall-back to the initiator call is done and it is not released. This is fixed now. It is only important for a third party PBX.<br/><br/><!--<br/>Changed files:<br/>dectradio_call.cpp<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp<br/>--><!---->
   
=== H.323: Problem with Media Re-Negotiation on a DECT handover call ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88379 88379]
|}
The DECT handover call works a little special concerning media renegotiation in a way that local preferences are never honored (the real media negotiation takes place between the original radio and the remote endpoint, the handover radio is just told the result. This special mode did not work correctly<br/><!--<br/>h323ch.cpp<br/>dectradio_ho_out.cpp<br/>dectradio_ho_out.h<br/>--><!---->
   
=== PBX Waiting: User Information Message from announcement interface accidentally forwarded to caller ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88426 88426]
|}
The announcement interface uses User Information signaling messages to send status information for example at the end of the announcement. This was forwarded to the caller by accident. Usually this does not do any harm, but on some ISDN networks it could result in clearing of the call because of unexpected message.<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
   
=== H.323: Potential Max-Busy-Ticks restart ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88441 88441]
|}
The H.323 state machine could enter a endless loop<br/><!--<br/>h323sig.cpp<br/>--><!---->
   
=== IP-DECT: No media after conference and toggled to held call ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88442 88442]
|}
If the DECT user leaves the conference mode and toggles to an held call, there is no voice. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>decradio_mastercall.h<br/>decradio_mastercall.cpp<br/>--><!---->
   
=== Admin IP: static routes configured at an ETH interface disappear when NAT or VLAN config of this interface is updated ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88451 88451]
|}
Routes configured under "IP4/ETHn/IP/Static IP Routes" were cleared when the "IP4/ETHn/NAT" page or the "ETHn/VLAN" page was left by pressing "OK".<br/><br/><!-- ip_ethif_nat.xsl ip_ethif_vlan.xsl -->  <!---->
   
=== IP232,IP222,IP241: Reduce memory requirements of display rendering ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88485 88485]
|}
Reduce memory requirements of display rendering.<br/><br/><!--<br/>lcd_map.cpp/h<br/>forms_color.cpp/h<br/>read_png.cpp<br/>--><!---->
   
=== PBX: Call Completion monitoring call should be terminated only after the CC ringout call is alerting ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88489 88489]
|}
This is needed for QSIG interoperability<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>--><!---->
   
=== Viewer was able to download config with standard password ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88503 88503]
|}
This was a security hole<br/><!--<br/>cfg-standard.txt<br/>--><!---->
   
=== Gateway: Hanging calls in state "Clearing" ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88527 88527]
|}
Hanging calls in state "Clearing".<br/>Caused by failed call replacement.<br/><br/><!--<br/>relay.cpp<br/>--><!---->
   
=== IP-DECT: Potential buffer overrun ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88535 88535]
|}
Buffer overrun check added in dectusers module for command 'show'.<br/><br/><!--<br/>Changed file:<br/>dectusers.cpp<br/>--><!---->
   
=== SIP: Missing UPDATE on call pickup ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88550 88550]
|}
The caller does not received UPDATE containing the connected party information.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== SIP: Handle one way of DTMF only ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88572 88572]
|}
Handle DTMF received via signaling message (INFO)<br/>or DTMF received via RTP (tlephone-event).<br/>Not both at the same call.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
   
=== PBX Conference: No Media in case media offer only received with alert/connect from called member ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88590 88590]
|}
This could happen if a broadcast object was called by a conference<br/><!--<br/>pbx_bc_conf.cpp<br/>--><!---->
   
=== IP222 IP232 IP241: Adjust equalizer above 6.3Khz for lower noise level ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88658 88658]
|}
to reduce noise with G722 codec<br/><br/><!-- ac_codec3.cpp platform/filter --><br/><!---->
   
=== SIP: Memory leak ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88688 88688]
|}
Memory leak on postponed RETRIEVE_NOTIFY.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== SIP: Invalid SDP answer if SDP offer contains RED ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88700 88700]
|}
Invalid SDP answer if SDP offer contains RED.<br/>Only on local media.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
= V9 Hotfix 18 (9061158) =
Changes included in Version 9 hotfix18
[http://mantis.innovaphone.com/view.php?id=88994 Definition]
 
== New Features ==
 
 
   
=== IPVA: Enhancement for innovaphone testbed: Indicate whether Upload to DRAM took place ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88223 88223]
|}
Required for innovaphones' automated testbed<br/><!--<br/>ipva.cpp<br/>--><!---->
   
=== Announcement Calls with DTMF feature code object ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88643 88643]
|}
A new DTMF feature code allows to make announcement calls to a dialed number.<br/><br/><!--<br/>pbx_dtmf.cpp<br/>pbx_dtmf.h<br/>pbx_edit_dtmf-ctrl.xsl<br/>--><!---->
   
=== Alarm and Event forwarding to a SYSLOG server ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88659 88659]
|}
If "Services/Logging/Alarm and Event Forward Server/Type" is set to SYSLOG the xml-formatted alarm and event info is sent to the Server(s) specified under "Services/Logging/Alarm and Event Forward Server/Address".<br/><br/><!-- logging.cpp logging.h logging.xsl fault_handler.cpp fault_handler.h --><br/><!---->
   
== Bug Fixes ==
 
 
   
=== TLS: Do not ignore early SOCKET_RECV ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88668 88668]
|}
Allow SOCKET_RECV from application between SOCKET_CONNECT and SOCKET_CONNECT_COMPLETE. This event flow is used by the HTTP client.<br/><!--<br/>tls.cpp<br/>unit-tests/tls/basic/102-handshake-client-normal-early-recv<br/>--><br/><!---->
   
=== Kerberos: Admin UI trap when having too many Kerberos hosts ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88698 88698]
|}
The problem occured if many Kerberos hosts (~1000) were registered on the server. In this case the box trapped due to an XML encoding problem when opening the page General/Kerberos or PBX/Config/Security.<br/><!--<br/>kerberos_db.cpp<br/>kerberos_ldap.cpp<br/>--><!---->
   
=== SIP: Provide display name in 200/OK for SUBSCRIBE ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88764 88764]
|}
Add display name to To header in 200/OK for SUBSCRIBE.<br/><br/><!--<br/>sip.cpp/h<br/>sipmsg.cpp/h<br/>siptrans.h<br/>--><!---->
   
=== IP22 IP24 IP28 IP305: Sometimes the DSP stops after sending CLIP ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89130 89130]
|}
..<br/><!-- ac_494e.cpp ac494004ce3_680.00.pf.01.h ac498004ce1_680.00.pf.01.h --><!---->
   
=== PBX: Configuration UI broken for checkmarks on PBX/Config/General and PBX/Config/myPBX ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89140 89140]
|}
The checkmarks on these two pages where all cleared when the other page was edited.<br/>Collateral damage of<br/><br/>fix: #88352: PBX: Allow configuration of default presence/dialog-info visibility for group members<br/><!--<br/>config_options.cpp<br/>pbx_general.xsl<br/>--><!---->
   
=== phone: ip222,ip232: the destination of an acticve diversion was not saved when the diversion was deactivated at  phone ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89177 89177]
|}
The destination (name or number) disappered when the diversion was deactivated. It was also not visible via the WEB interface.<br/><br/><!-- app_gui.cpp --><!---->
   
=== Gateway: Missed FAX pages received with the FAX interface ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89587 89587]
|}
If the FAX interface is used to receive a FAX document with ECM mode and the transmitting terminal appends additional EOLs, the page counter is wrong and document pages are not written. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>t30.cpp<br/>facsimile_g3.h<br/>facsimile_g3.cpp<br/>facsimile_ECM.h<br/>facsimile_ECM.cpp<br/>--><!---->
   
= V9 Hotfix 19 (9061180) =
Changes included in Version 9 hotfix19
[http://mantis.innovaphone.com/view.php?id=89318 Definition]
 
== New Features ==
 
 
   
=== PBX-SOAP: Present normalized number of peer also ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88521 88521]
|}
The SOAP API presents the adjusted number of the peer (called/calling), which is the shortest possible number which can be dialed to call this. It is the same number as displayed on the phone. Sometimes an application needs to know the normalized number of the peer, which is the number in the context of the root node. This number is sent as additional number with the identifier "norm"<br/><!--<br/>pbx_xml.cpp<br/>pbx.cpp<br/>--><!---->
   
=== IP232,IP222,IP241: Config option to adjust LCD brightness in idle state ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89261 89261]
|}
-> Main Menu -> Phone Setup -> LCD light (idle state)<br/>Can be tuned down to zero.<br/><br/><!--<br/>9.00\\phone_orchid\\box\\forms\\lcd_lib\\forms_color.cpp<br/>9.00\\phone_orchid\\box\\forms\\lcd_lib\\forms_color.h<br/>9.00\\phone_orchid\\box\\forms\\lcd_lib\\forms_color_mod.cpp<br/>9.00\\phone_orchid\\common\\interface\\forms.cpp<br/>9.00\\phone_orchid\\common\\interface\\forms.h<br/>9.00\\phone_orchid\\phone\\app\\app_form.cpp<br/>9.00\\phone_orchid\\phone\\app\\app_gui.cpp<br/>9.00\\phone_orchid\\phone\\app\\app_gui.h<br/>9.00\\phone_orchid\\phone\\phone_user_if.h<br/>9.00\\phone_orchid\\phone\\phonetxt.h<br/>9.00\\phone_orchid\\phone\\txt\\phonetxt-latin1.cpp<br/>9.00\\phone_orchid\\phone\\txt\\phonetxt-utf8.base<br/>9.00\\phone_orchid\\phone\\txt\\phonetxt-utf8.cpp<br/>9.00\\phone_orchid\\phone\\user\\phone_user.cpp<br/>9.00\\phone_orchid\\phone\\user\\phone_user.h<br/>9.00\\phone_orchid\\platform\\config.h<br/>--><!---->
   
=== H.323: Automatically connect signaling TCP if NAT router is detected ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89497 89497]
|}
When regestering an endpoint from a private network to a PBX within the public network, the signaling TCP connection must be established and maintained by the endpoint. Otherwise calls to the endpoint are not possible.<br/><!--<br/>h323sig.cpp<br/>h323ras.cpp<br/>h323.h<br/>h323asn1.h<br/>--><!---->
   
=== PRI-QSIG: Interop config for channel numbering ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89578 89578]
|}
The QSIG standard defines to use Channel numbers (1-30) instead of timeslot (1-15, 16-31) as it is defined for EDSS1. There are many 'old' QSIG implementations around, which do it wrong. The QSIG-ECMA1 protocol setting is used for these 'old' implementations and the QSIG-ECMA2 setting for standard conform inplementations.<br/><br/>With the QSIG-ECMA1 also 'old' facility coding is used. There is also the combination of standard facility coding and timeslots for channels around so an independent mechanism to configure the channel numbering is needed.<br/><!--<br/>q931.cpp<br/>q931.h<br/>isdn_interop.xsl<br/>--><!---->
=== Other new Features ===
 
{|
|-
|valign=top nowrap=true|[http://mantis.innovaphone.com/view.php?id=89367 89367]
|valign=top nowrap=true|jfr
|case independence for the characters of the Basic Russian Alphabet added
|}
 
   
== Bug Fixes ==
 
 
   
=== SIP/TLS: Rejecting server certificate ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88444 88444]
|}
Validating server certificate against configured domain name.<br/>Must be validated against configured proxy domsina name.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
   
=== SIP: Wrong branch value in Via header in ACK request ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89317 89317]
|}
Branch value in Via header in ACK request must be new after 200 response.<br/>Branch value in Via header in ACK request must be same after non-200 response.<br/><br/><!--<br/>siptrans.cpp/h<br/>--><!---->
   
=== H.323: RTP-DTMF did not work on exclusive coder/media relay configurations ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89328 89328]
|}
Problem for DTMF on SIP trunks<br/><!--<br/>h323ch.cpp<br/>--><!---->
   
=== AD Replication: A Buffer for Processing The Paged Result Cookie Was Too Small ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89385 89385]
|}
1KB wasn't enough, now 2KB.<br/><!--<br/>interface/ldapapi.cpp/.h<br/>--><!---->
   
=== LDAP Expert: "Next"-Browsing through DB failed ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89391 89391]
|}
URI encoding error<br/><!--<br/>fdirui_right.xsl<br/>--><!---->
   
=== phone: ip222, ip232,ip241:  the notification tone indicating a new message when a call is active was sent to remote ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89392 89392]
|}
instead to notify the receiver of the message the remot party did hear the tone<br/><br/><!-- phonesig.cpp app_call.cpp --><!---->
   
=== SIP: "Supported: timer" missing in UPDATE message ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89429 89429]
|}
"Supported: timer" missing in UPDATE message.<br/><br/><!--<br/>sip.cpp/h<br/>siptrans.cpp/h<br/>--><!---->
   
=== phone: ip222,ip232: humming noise in USB headset speaker in outbound call setup phase, disappears once connected ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89432 89432]
|}
sometimes a humming noise was heard in the USB headset speaker in the setup phase of an outbound call. it disappeared as soon as the call was connected.  <br/><!-- ac_phone3.cpp --><!---->
   
=== phone:  an intrusion call set up via Partner function key could not be cleared at the intruding phone via TAPI ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89443 89443]
|}
The TAPI interpreted the recording state as a conference<br/><br/><!-- app_call.cpp --><!---->
   
=== log message forwarding to another innovaphone device did not work since V9hotfix15 (on the receiving device) ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89492 89492]
|}
this problem is located on the reciving device, it does not depend on the the version of the forwarding device.<br/><br/><!-- logging.cpp logging.h --><!---->
   
=== IP22 IP24 IP28 IP305: DSP debug code added ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89493 89493]
|}
..<br/><!-- ac_494.cpp --><!---->
   
=== PBX Mobility: Conference ID not set for outgoing calls initiated by myPBX ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89502 89502]
|}
This could cause several problems:<br/>- When the call was sent to a local user with multiple registrations, the call to each registration had a different conferenceID, so myPBX could not match these calls to actually being only a single call, so multiple calls were dissplayed<br/>- The CDRs created for this call could not be matched<!---->
   
=== SIP: Don't tell application that registration is down when handling redirect response ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89582 89582]
|}
Don't tell application that registration is down when handling redirect response for REGISTER.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== SIP: No T.38  parameter when indicating capabilitity only ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89617 89617]
|}
No T.38  parameter when indicating capabilitity only.<br/><br/>Offer<br/>\tv=0<br/>\to=- 2 1 IN IP4 172.16.16.124<br/>\ts=-<br/>\tt=0 0<br/>\tm=audio 16386 RTP/SAVP 8 101 13<br/>\tc=IN IP4 172.16.16.124<br/>\ta=rtpmap:101 telephone-event/8000<br/>\ta=fmtp:101 0-15<br/>\ta=ptime:20<br/>\ta=silenceSupp:off - - - -<br/>\ta=sendrecv<br/>\ta=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:WJl714I+mpSr47ld1YjYMf8t9xQo0xYHUng1CnDi<br/>\tm=image 0 udptl t38<br/>\tc=IN IP4 172.16.16.124<br/><br/>Instead of<br/>\tv=0<br/>\to=- 2 1 IN IP4 172.16.16.124<br/>\ts=-<br/>\tt=0 0<br/>\tm=audio 16394 RTP/SAVP 8 101 13<br/>\tc=IN IP4 172.16.16.124<br/>\ta=rtpmap:101 telephone-event/8000<br/>\ta=fmtp:101 0-15<br/>\ta=ptime:20<br/>\ta=silenceSupp:off - - - -<br/>\ta=sendrecv<br/>\ta=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:sPngj1zoO9wI1pc1tMTymWCzTgwMoamPuJAFIhga<br/>\tm=image 0 udptl t38<br/>\tc=IN IP4 172.16.16.124<br/>\ta=T38FaxVersion:0<br/>\ta=T38MaxBitRate:14400<br/>\ta=T38FaxFillBitRemoval:0<br/>\ta=T38FaxTranscodingMMR:0<br/>\ta=T38FaxTranscodingJBIG:0<br/>\ta=T38FaxRateManagement:transferredTCF<br/>\ta=T38FaxUdpEC:t38UDPRedundancy<br/><br/><br/><!--<br/>sdpmsg.cpp<br/>--><!---->
   
=== OS: Potential trap when starting a timer, which is already deleted ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89628 89628]
|}
Could result in a strange trap with PBX transfer/recall<!---->
   
=== phone:  while recording was running for a call this call ould not be cleared via SOAP/TAPI ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89629 89629]
|}
The TAPI assumed the intruding phone to be in a conference<br/><br/><!-- app_call.cpp --><!---->
   
=== PBX Admin UI: Reconfiguring user visibility (Access) did not work correctly ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89636 89636]
|}
If a setting for a user/group was deleted, it could happen that some of the settings (Group, Online, Presence, Dialog, Ids) where copied to the next entry.<br/><!--<br/>pbx_admin.cpp<br/>--><!---->
   
=== Voicemail: Trap ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89648 89648]
|}
If Trace-Checkmark activated, a diverted call carrying an mwi interrogate was processed<br/><!--<br/>pbx_vm.cpp<br/>--><!---->
   
=== Possible Trap While Getting Post-Mortem Log ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89652 89652]
|}
Possible Trap While Getting Post-Mortem Log<br/><!--<br/>httpclient.cpp<br/>signal.cpp<br/>--><!---->
   
=== Media: Redirecting SRTP streams for NAT clients only after successful SRTP authentication ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89661 89661]
|}
Media endpoints support NAT.<br/>If receiving RTP/SRTP from an address other than negotiated one<br/>media endpoints redirecting their media stream towards source of incoming media stream.<br/>In case of SRTP, this NAT workaround is only executed if incoming media stream has passed authentication.<br/>For securitiy reasons.<br/><br/><!--<br/>media.cpp<br/>--><!---->
   
=== AD Replication: Merge v10 code into v9 code ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89671 89671]
|}
Some recent refactoring works weren't in sync. A state machine didn't reach "Completed".<br/><!--<br/>service/ldap/<br/>--><!---->
   
=== phone: if a call ringing while the handset was offhook was accepted via SOAP/TAPI the call was not cleared when going onhook ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89680 89680]
|}
This could happen when the handset was lifted and kept lifted after the disconnect key was prressed. an inbound call arriving in this state could be accepted via SOAP/TAPI and was connected to the handset but the call could not be cleared by going onhook. Only the disconnect key did clear the call.  <br/>Now the call is cleared as expected when going onhook.    <!---->
   
=== phone_orchid: pressing speaker key in handset/headset mode switches to handsfree mode, pressing again returns to previous mode ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89730 89730]
|}
handset/headset plus speaker is not supported on phone_orchid, the previous solution where the connection was dropped when the speaker key was pressed again  (see #84297) was perceived as irritating.<br/><br/><!-- ac_codec3.cpp ac_codec3.h phonesig.cpp app_ctl.cpp --><br/><br/><!---->
   
=== IP22 IP24 IP28 IP305: Sometimes the DSP stops after sending CLIP (2) ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89760 89760]
|}
..<br/><!-- ac_494e.cpp ac494004ce3_680.00.pf.02.h  --><!---->
   
=== SIP: Bug in handling of INVITE with Replaces ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89777 89777]
|}
Bug in handling of INVITE with Replaces.<br/>Results in hanging call.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== IP-DECT: Trap with call transfer ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89786 89786]
|}
A trap occurs if a call transfer is received in the IP-DECT radio module. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dectradio.cpp<br/>--><!---->
   
=== Fix for MIPS counter ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89804 89804]
|}
MIPS counter was incorrect on IP1201 and IP4001<br/><br/><!--<br/>ip1201.cpp<br/>ip4001.cpp<br/>--><!---->
   
=== SIP: Media negotiation fails on calls into Waiting Queue ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89838 89838]
|}
Media negotiation fails on calls into Waiting Queue if<br/>caller put call on hold before WQ agent accepts the call.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== 'power-off loop' relay switching function failure ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89942 89942]
|}
Especially POE-switches with higher supply voltages than 48V lead to a decreased timespan of powering the build-in relays of a ip6010/ip810 gateway. The detection of a power-fail condition is therefore derived from the POE ICs which react earlier and thus increases powering time of the relays.<!---->
   
=== myPBX: Default group visibility was not displayed correctly ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89954 89954]
|}
The default group visibility can be configured from v9hotfix17. Regardless of that configuration myPBX showed full visibility in the visibilty settings.<br/><!--<br/>pbx_api.h<br/>pbx.h<br/>pbx_client.cpp<br/>--><!---->
   
=== IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: DSP Update to Version 680.07 ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89956 89956]
|}
Fixed modem bypass with slow modems<br/><br/><!-- <br/>DSP code details at <br/>\\\\innovaphone.sifi\\dfs\\Dokumente\\Hardware\\G7xx\\Audiocodes\\firmware\\firmware_ac49x\\680.07<br/><br/>ac49x_680/ac49*.h  ac_494e.cpp ac_491.cpp ac_dsp3.cpp<br/><br/>Also AC494 DSP clock changed from 100 to 125MHz and report interval changed to 5seconds. <br/>--><!---->
   
= V9 Hotfix 20 (9061198) =
Changes included in Version 9 hotfix20
[http://mantis.innovaphone.com/view.php?id=90101 Definition]
 
== New Features ==
 
 
   
=== Linux: Memory allocation changed for IP810 ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=86420 86420]
|}
The memory allocation for the IP810 is changed to 128MB/384MB for innovaphone/Linux.<br/><br/>Important:<br/>Linux [[Support:Linux_Application_Platform_Hotfix_11_and_before_won't_run_on_Firmware_V9_hotfix_20_and_later_on_IP810|must be used with the kernel version 3.4.10 or later]]. This kernel is included in the Linux Application Platform V9.00 hotfix12 and later. The kernel is automatically updated with the Linux Application Platform V9.00 hotfix12.<br/><br/><!--<br/>Changed files:<br/>ip6010/platform/config.h<br/>ip6010/platform/startup.S<br/>ip6010/platform/ip6010.cpp<br/>--><!---->
   
=== phone: new "Do Not Disturb" action "ring once" ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89960 89960]
|}
If "Phone/User-x/Preferences/Do Not Disturb/Action: ring once" is selected a new inbound call is indicated with a short tone only.<br/>Both the the tone and the duration of the tone can be configured under ""Phone/User-x/Preferences/Ring Tones/Do Not Disturb".<br/>If not configured the default ring tone is played for one and a half second. <br/><br/><!-- phone_config.cpp phone_edit.cpp reg_edit_general.xsl phone_config.h phone-strings-latin1.h phone-strings-utf8.h phone-text-tables.h app_ctl.h app_call.cpp app_gui.cpp phonetxt.h phonetxt-utf8.cpp phonetxt-latin1.cpp phonetxt-utf8.base<br/> --><!---->
   
=== PBX: Support of long user-user-informations by SOAP ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90029 90029]
|}
Support of long user-user-informations (UUI) for SOAP sessions added. A long UUI is split into multiple short UUIs supported by Q.931.<br/>It is required by the FAX interface.<br/><br/><!--<br/>Changed files:<br/>pbx_xml.h<br/>pbx_xml.cpp<br/>--><!---->
   
=== Relay: Support of long user-user-informations by FAX ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90030 90030]
|}
Support of long user-user-informations (UUI) for the FAX interface added. A long UUI is split into multiple short UUIs supported by Q.931.<br/><br/><!--<br/>Changed files:<br/>webfax.h<br/>webfax.cpp<br/>--><!---->
   
=== Gateway: Support of a header line for FAX documents ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90374 90374]
|}
Support of a header line for FAX documents is added in the FAX interface.<br/><br/><!--<br/>Changed files:<br/>webfax.h<br/>webfax.cpp<br/>t30.h<br/>t30.cpp<br/>t4.h<br/>t4.cpp<br/>facsimile_g3.h<br/>facsimile_g3.cpp<br/>--><br/><!---->
   
=== SIP: Support for MESSAGE inside voice call ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90408 90408]
|}
Support for text messages inside voice call.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== PBX-SOAP: Support for 'rc' and 'srce164' on UserCall on Waiting Queue ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90537 90537]
|}
If a Waiting Queue is used for outgoing calls, these features can be usefull for some applications<br/><!--<br/>pbx_api.h<br/>pbx_xml.cpp<br/>pbx_wait.cpp<br/>pbx_wait.h<br/>--><!---->
   
=== PBX: New configuration option 'Hide connected Number' at object ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90693 90693]
|}
In some cases it is desireable not to reveal the final destination of a call to a caller. For example a call center agent should not be called directly by the customer.<br/><!--<br/>pbx.cpp<br/>pbx_api.h<br/>pbx_admin.cpp<br/>pbx_edit_object.xsl<br/>pbx_wait.cpp<br/>--><!---->
   
=== IP-DECT: Configuration option 'Registration with system password' ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91460 91460]
|}
The configuration option 'Registration with system password' is added. If ticked, all users are registered with the system password. This is useful, if the PBX users are only allowed to register with the PBX password.<br/><br/><!--<br/>Changed files:<br/>dectmaster.h<br/>dectmaster.cpp<br/>dectmaster.xsl<br/>_inn_asc_dectmaster.xsl<br/>--><!---->
   
=== Voicemail: URL-En-/Decoding ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91609 91609]
|}
new statement allows to URL-encode or URL-decode a string<br/><lib-enc string=".." string_out="$var" type="url"/><br/><lib-dec string=".." string_out="$var" type="url"/><br/><!--<br/>vm_script_if.cpp<br/>--><br/><!---->
   
== Bug Fixes ==
 
 
   
=== SIP: Auto answer with SDP in ACK ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89539 89539]
|}
Auto answer results into no-audio when INVITE comes w/o SDP offer.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== SIP: Fix for overlap dialing with KPML ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89581 89581]
|}
Dialing digits entered before KPML subscription is established<br/>need to be queued until KPML subscription is established.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
   
=== SIP: Re-try INVITE after 407 even if no password configured ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90024 90024]
|}
Calculate Digest with zero-length password and re-try INVITE.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== IP232,IP222,IP241: Truncate directory entry information ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90027 90027]
|}
Truncate directory entry information to keep from overlapping with number type indication.<br/><br/><!--<br/>forms_color.cpp<br/>--><!---->
   
=== IP-DECT: Don't show DTMF in radio call list ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90034 90034]
|}
User dialled digits during calls which are sent as DTMF should not be shown in the radio call list. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>dect_comcerto/platform/config.h<br/>ip1201/platform/config.h<br/>ip4001/platform/config.h<br/>dectradio.h<br/>dectradio.cpp<br/>dectradio_mastercall.cpp<br/>--><!---->
   
=== H.323: Allow media offers with 0.0.0.0 as address ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90053 90053]
|}
Needed for SIP interoperability. Some third party SIP PBXs use addresses of 0.0.0.0 to indicate that they don't receive media. This may happens if an endpoint is put on hold. We did not forward such an offer and thus no Music on Hold was heard.<br/><!--<br/>h323ch.cpp<br/>--><!---->
   
=== PBX: Busy On ... Calls at PBX objects did not take into acccount that a call may be routed back to Slave ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90054 90054]
|}
The busy on ... calls on PBX objects can be used to limit bandwidth usage between a master and a slave to a certain number of calls. Some calls are sent from a slave to the master and back to the slave if the routing decision cannot be done on the slave alone. This happens if escapes are used which overlap other obects (e.g. the local trunk). It is a common configuration the the E.164 routing scheme.<br/>With this fix, these calls are not counted for this purpose.<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_api.h<br/>pbx_wait.cpp<br/>pbx_wait.h<br/>pbx_trunk.h<br/>pbx_gw.h<br/>pbx_exec.h<br/>pbx_bc_conf.cpp<br/>pbx_bc_conf.h<br/>--><!---->
   
=== Voicemail: Send silence RTP during recording ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90095 90095]
|}
Send silence RTP during recording for some SIP carriers that do not send RTP without receiving RTP.<br/><br/><!--<br/>webmedia.cpp<br/>--><!---->
   
=== IP2000: Prevent blinking error LED  IP2000 ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90102 90102]
|}
The firmware tried to load  the conference DSP, which is not available on the IP2000<br/><br/><!-- config.h --><br/><!---->
   
=== SIP: Memory leak when receiving NOTIFY(message-summary) ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90106 90106]
|}
Memory leak when receiving NOTIFY(message-summary)<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== H.323: Support for registration from a private network thru NAT ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90306 90306]
|}
In case an endpoint registers to a PBX from within a private network thru a NAT router, the signaling TCP connection must be maintained in order to be able to receive calls. When the registration is up a dummy call is sent to the PBX to establish the signaling TCP. This TCP connection is maintained after the dummy call is cleared. If this TCP connection is lost (e.g. NAT Router reset), the Registration is cleared and restarted, so that after the re-registration another dummy call is sent.<br/><br/>This is a fix for the previous fix<br/><br/>fix: #89497: H.323: Automatically connect signaling TCP if NAT router is detected<br/><br/>which did not work well<br/><!--<br/>h323sig.cpp<br/>h323.h<br/>voip.h<br/>gk.cpp<br/>relay.cpp<br/>relay_api.h<br/>relay.h<br/>--><!---->
   
=== SIP: SUBSCRIBE using old IP address in Contact field ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90320 90320]
|}
If the IP address is changed at DHCP renew (or network change) the endpoint will immediately do a re-register to update the SIP Proxy with the new IP address.<br/>All SIP messages but SUBSCRIBE uses the new IP address in the Contact field.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== Voicemail: Memory Load High With Repeated Calls To &lt;store-getnext&gt; ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90362 90362]
|}
The use-case does not happen within the standard voicemail, but within e.g. innovaphone conferencing.<br/><!--<br/>vm_script_if.cpp<br/>--><!---->
   
=== PBX: CC Requests were sent with wrong number if a SendNumber was configured at the user ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90432 90432]
|}
Usually this is no problem only when interworking with some QSIG PBX's this causes the call-completion to fail.<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== phone_orchid: dialtone missing when recording is  active and the active call is held to open a consultation call ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90433 90433]
|}
the consultation call could be established but there was no dialtone after pressing the R-key and no ringback tone after the number had been entered.<br/><br/><!-- ac_phone3.cpp ac_phone3.h codec.cpp codec.h phonesig_if.h phonesig.cpp phonesig.h app_call.cpp --><!---->
   
=== Possible trap when doing a leak check ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90451 90451]
|}
When many leaks exist or leak check is done when much tracing is turned on. The leak check itself could cause a watchdog trap, because the collecting of the leaks is done on highest priority so not even the timer interrupt could trigger the watchdog.<br/><!--<br/>os.cpp<br/>--><!---->
   
=== IP-DECT: Busy state on maximum call count ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90461 90461]
|}
The base station does not go to the busy state if the maximum call count is reached and the last call is an incoming call. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>dectradio.cpp<br/>dectradio_dectcall.cpp<br/>--><!---->
   
=== PBX: Support for Opticaller Data Callthru did not work ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90480 90480]
|}
Support for Opticaller data callthru was added, but did not work<br/><!--<br/>pbx_admin.cpp<br/>--><!---->
   
=== 802.1x: Version 3 Frames Weren't Processed ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90485 90485]
|}
Now they're read initially. Handshakes remains to be in version 1.<br/><!--<br/>8021x.cpp<br/>--><!---->
   
=== PBX: Forking a call to a Trunk with "Outgoing call restricted", causes the original call to be restricted as well ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90487 90487]
|}
The "Outgoing call restricted" flag on the trunk object to which the call was forked caused the call as a whole to be marked as Calling Line Presentation Restricted.<br/><!--<br/>pbx_trunk.cpp<br/>--><!---->
   
=== SIP: Locally configured DNS entries were not used if no DNS server configured ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90508 90508]
|}
If no DNS server was configured, but DNS names are to be resolved,<br/>local DNS entries can be added (Services/DNS/Hosts).<br/>SIP stack fails with SRV query and does not try A query which would deliver IP address.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== IP-DECT: RTP stream ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90539 90539]
|}
If a remote hold event is received, no RTP data should be sent by the IP-DECT device.<br/>A CTI initiated call is established with a call transfer and a "No Media data received" error event can occur.<br/>This is fixed now.<br/><br/><!--<br/>Changed files:<br/>h323ch.cpp<br/>dectradio_call.cpp<br/>dectradio_mastercall.h<br/>dectradio_mastercall.cpp<br/>--><!---->
   
=== SIP: Trap on IP-DECT ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90569 90569]
|}
Trap in GK-CHANNEL when Dectmaster application sends DTMF before call if created.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== SIP: Disabled IP-DECT interface tries to register ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90597 90597]
|}
Disabled IP-DECT interface tries to register.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== H.323: Timer to monitor response to setup too short for some traffic cases ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90696 90696]
|}
This created unnecessary event in IP-DECT systems when calling powered off or out of range handsets<br/><!--<br/>h323sig.cpp<br/>--><!---->
   
=== PBX Waiting: Potential Trap if editing while a call is initiated with SOAP ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90766 90766]
|}
The Waiting object can be used as outgoing dialing object with SOAP. If this is done and the configuration is changed while an outgoing call was pending, a trap could happen<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
   
=== H.323: Potential Trap in special case which could only happen in version 10 ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90768 90768]
|}
This fixed is merged to version 9 only of consistency reasons<br/><!--<br/>h323_ras.cpp<br/>--><!---->
   
=== SIP: Respect changes in PAI/PPI header when receiving UPDATE with SDP offer ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90778 90778]
|}
PAI/PPI was processed when receiving UPDATE without SDP offer.<br/>PAI/PPI was ignored when receiving UPDATE with SDP offer.<br/>Now PAI/PPI is processed when receiving UPDATE with SDP offer.<br/><br/><!--<br/>sip.cpp<br/>--><br/><!---->
   
=== SIP: Possible buffer overrun ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90780 90780]
|}
Fix for possible buffer overrun.<br/><br/><!--<br/>siputil.cpp/h<br/>--><!---->
   
=== SIP: Wrong error log "Timeout during media negotiation for call" ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90821 90821]
|}
Error log "Timeout during media negotiation for call" may occur after re-negotioation.<br/>Re-negotioation occurs during hold/retrieve/transfer.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== PBX Trunk: Name to Number Feature did not work with calls to extern ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90858 90858]
|}
If an endpoint cannot be found by name, the call should be forwarded to extern<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_api.h<br/>pbx_trunk.cpp<br/>pbx_trunk.h<br/>--><!---->
   
=== PBX CSV Import: Corrupted objects at buffer boundaries ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90942 90942]
|}
The upload is processed in chunks of 2K. At boundaries of these chunks data could be corrupted. This was fixed and the chunk size increased to 10K<br/><!--<br/>pbx_admin.cpp<br/>pbx_admin.h<br/>--><!---->
   
=== Phone: Trap when selecting registration for a directory entry ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90999 90999]
|}
Trap when selecting registration for a directory entry,<br/>but only if the registration has either no name or no number.<br/><br/><!--<br/>phone_dir_ui.cpp<br/>--><!---->
   
=== H.323: No event should be generated in State 11 and 25 ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91020 91020]
|}
State 25 is incoming overlap sending. This means a call was received with incomplete dialing information and the caller failed to dial more digits within the timeout of 2min. This is no indication of any malfunction but only a usage problem, so no event should be generated.<br/><br/>State 11 is disconnecting with inband announcement. A timeout happens if a user listens to the announcement for more then 30s. This could be normal.<br/><!--<br/>h323sig.cpp<br/>--><!---->
   
=== SIP: Trap when cancelling call ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91023 91023]
|}
Double delete of a call entity.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
   
=== SIP: Heavy TLS retry load when server certificate was rejected ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91033 91033]
|}
Collateral damage from earlier fixed for DNS refreshing.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== H.323: Incoming faststart call was sometimes not accepted as faststart ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91136 91136]
|}
Only happens if non EFC is used on the incoming call, so this only happens in interop cases with other H.323 equipment<br/><!--<br/>h323ch.cpp<br/>--><!---->
   
=== SNMP Get-Next Requests Carrying an Octet-String Value Caused Memory Leak ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91215 91215]
|}
<!--<br/>mib.cpp<br/>--><!---->
   
=== SIP: Media negotiation for video fails if called through waiting queue or multi reg ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91235 91235]
|}
Media negotiation for video fails if called through waiting queue or multi reg.<br/>In this case the PBX has to handle offer/offer-collision.<br/>In this case the PBX must select audio and video codec.<br/>In this case the PBX must send SDP answers to both endpoints.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== SIP: Secondary target (hostname) is not resolved ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91287 91287]
|}
Usually a response to a SRV query delivers additional records containing the ip address of any target (hostname).<br/>Some DNS servers do not.<br/>Additional A querys are required.<br/>An A query was issued for the primnary target (most preferred hostname).<br/>No A query was issued for the secondary target (less preferred hostname).<br/>Fixed now.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
   
=== IP-DECT: Trap in Radio module (IP1202) ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91315 91315]
|}
A trap in the IP-DECT Radio module occurs if the Mobility Master is used and a duplicate IPEI command is sent to the Master. The Master handles it with a location cancel and an endpoint delete command sent to the radio. If the two commands arrives with no delay, the Radio module traps. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>dectradio.cpp<br/>dectlocalusers.cpp<br/>--><!---->
   
=== Ldap Replication from NDS ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91347 91347]
|}
Skip isDeleted attribute with content others than 'true'. Occurred when replicating from an NDS running in AD compatibility mode.<br/><!--<br/>ldaprep.cpp<br/>--><!---->
   
=== SIP: Wrong call was disconnected after successful transfer ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91349 91349]
|}
Wrong call was disconnected after successful transfer.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== SIP: SDP answer for T.38 switch-over must contain multiple media descriptions ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91377 91377]
|}
... if SDP offer contains multiple media descriptions.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== PBX: Not possible to login as user with non-full admin rights if 'Password protect all Pages' ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91414 91414]
|}
Some pages needed for the UI, which are normally not password protected, could not accessed with the reduced rights<br/><!--<br/>http.cpp<br/>--><!---->
   
=== SIP: Follow offers ptime proposal ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91421 91421]
|}
Better follow offers ptime proposal.<br/>Otherwise SAMwin operator does not stop sending re-INVITE.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== IP0010 IP1060 IP3010 IP6010 IP22 IP24 IP28 IP302 IP305: switch from modem bypass to voice did not work ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91429 91429]
|}
In some cases announcments with music trigger modembypass, in this case at least the fallback to voice should work.<br/><br/><!-- ac_dsp3.cpp --><!---->
   
=== H.323: Allow update of Registration password ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91440 91440]
|}
needed in case a password is changed, which is replicated to a DECT system<br/><!--<br/>h323sig.cpp<br/>h323.h<br/>voip.cpp<br/>voip.j<br/>--><!---->
   
=== H.323: PROGRESS in connected state was treated as 'unexpected' ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91483 91483]
|}
This happened with H.323 connections without registration when disconnecting a call with inband information (e.g. a call to an ISDN interface). Unnecessary events were generated.<br/><!--<br/>h323sig.cpp<br/>h323sig.h<br/>h323_tbl.tbl<br/>--><!---->
   
= V9 Hotfix 21 (9061222) =
Changes included in Version 9 hotfix21
[http://mantis.innovaphone.com/view.php?id=91721 Definition]
 
== New Features ==
 
 
   
=== PBX: Allow Name (instead of Long Name) to identify user for mobility data call thru ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91660 91660]
|}
This should simplify Opticaller configuration<br/><!--<br/>pbx_admin.cpp<br/>--><!---->
   
=== IP-DECT: Static ports between Master and Radio ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91815 91815]
|}
Now the VOIP connections between the Master and the Radio use static ports instead of dynamic ones. This is useful if only a few ports should be opened through a firewall. For calls from the Radio to the Master the ports 1716 and 1717 (TLS) are used. For the default Master connection for calls from the Master to the Radio the ports 1718 and 1719 (TLS) are used. For dynamic Radio-Master connections the ports from 1722 are used. Every connection needs two ports.<br/><br/><!--<br/>Changed files:<br/>dectmaster.cpp<br/>dectradio.h<br/>dectradio.cpp<br/>dectradio_reg.h<br/>dectradio_reg.cpp<br/>h323sig.cpp<br/>--><br/><!---->
   
=== IP-DECT: Cipher key index request for security test devices ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92223 92223]
|}
Cipher key index request procedure is changed to pass the test with security test devices. The cipher key index is used for DECT "Early Encryption"(EE).<br/><br/><!--<br/>Changed files:<br/>dectusers_if.h<br/>dectmaster.cpp<br/>--><!---->
   
=== Voicemail: &lt;pbx-getcallinfo out-calling-name="..."/&gt; ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92286 92286]
|}
Pass H.450 callingName into the script<br/><!--<br/>pbx/vm/*<br/>--><!---->
   
=== PHONE_SIG_MODE_KEEP_NUMBER_TYPE can be set at phonesig startup to keep type of number in all q931 numbers ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92385 92385]
|}
<!-- phonesig_if.h phonesig.cpp --><!---->
   
=== PBX Trunk: Flag to block presence/dialog-info subscriptions ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92824 92824]
|}
Some networks e.g. sip carriers behave badly when receiving subscribes for presence/dialog-info, which cannot be handled, so there is an option added to block these.<br/><!--<br/>pbx_api.h<br/>pbx.cpp<br/>pbx_trunk.cpp<br/>pbx_trunk.h<br/>pbx_edit_trunk.xsl<br/>--><!---->
   
=== SIP: Debug information for problems with STUN ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=93233 93233]
|}
Added debug output to trace problems with STUN.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== Gateway: FAX interface User-User-Info error response in disconnect event ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=93455 93455]
|}
User-User-Info response of the FAX interface is not forwarded in the alerting state. The problem exists when call to Fax interface was routed through multiple PBX. Now the UUI response is sent in the disconnect event if the response is a error notification.<br/><br/><!--<br/>Changed files:<br/>dummysig.cpp<br/>webfax.h<br/>webfax.cpp<br/>--><br/><!---->
   
=== IP222 IP232 IP241: DSP code update ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=94916 94916]
|}
DSP code update to version 680<br/><br/><br/><!-- ac_494e.cpp phone_orchid.mak  ac_codec3.cpp phone-dsp.xsl new dsp file <br/><br/><br/><br/>Dear Carsten,<br/> <br/>We found the bug in our FW.<br/>You can try using the attached patch or, if you prefer, we should have a new official 680 release within two weeks.<br/>Anyway, I’ll update you once the new release is available.<br/> <br/>Regards,<br/>Eliezer <br/> <br/>--><!---->
   
=== SIP: New interop tweak "No Remote Hold Signaling" ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=94954 94954]
|}
New config option on gateway interfaces "No Remote Hold Signaling".<br/>Disables interworking of "inactive" into remoteHold.<br/><br/><!--<br/>voip.h<br/>sip.cpp<br/>gk.cpp/h<br/>relay_edit_voip.xsl<br/>--><!---->
   
=== IP6010: DSP trace options improved ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=95078 95078]
|}
T38 trace flag worked only in one direction.<br/>Changing trace options needed a reboot.<br/><br/><!-- ac_491.cpp  ac_dsp3.cpp --><br/><!---->
   
=== IPv6: Disable checkmark on ethernet interfaces added ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=95452 95452]
|}
Allows to disable sending/receiving of IPv6 packets on this interface. This may be desireable for security reasons<br/><!--<br/>ip6_ethif.cpp<br/>ip6_ethif.h<br/>ip6_ethif_ip.xsl<br/>--><!---->
   
=== IP222 IP232 IP241: DSP code update ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=95764 95764]
|}
Channel was muted during silent periods ( no CNG during NLP operation ).<br/><br/><!-- ac_494e.cpp  ac494e002ce4.680.10.pf.01. --><!---->
   
== Bug Fixes ==
 
 
   
=== PBX Mobility: Connected number from mobile phone was forwarded to caller ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90722 90722]
|}
If a call is answered on the mobile phone, it should look identical to the caller to the case that the call was answered locally. This means a connected number from the mobile phone must not be forwarded.<br/><!--<br/>pbx_mobility.cpp<br/>--><!---->
   
=== Potential Trap when rapidly switching local Media connections (Conferencing) ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=90933 90933]
|}
There was a race condition when switching local media channels (e.g. ISDN channels to conference interfaces), which could cause media not functioning or even a trap<br/><!--<br/>media.cpp<br/>medialib.h<br/>--><!---->
   
=== SIP: Better handling of incoming calls ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91153 91153]
|}
Msg sequence INVITE,CANCEL,INVITE may result into second call rejected.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
   
=== IP0010,3010,6010,1060,810: ethernet link down not detected when a cable was unplugged after boot ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91600 91600]
|}
The link state interrupt was triggered only once after boot, further link state changes were not indicated.<br/><br/><!-- mv78x00_drv.cpp --><!---->
   
=== myPBX: Rejecting an incoming broadcast call disconnected the call for all alerting endpoints ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91719 91719]
|}
The call sould continue alerting on the other endpoints.<br/><!--<br/>pbx_client.cpp<br/>--><!---->
   
=== phone: ip222, ip232:  reset/restart USB headset when a hang condition or a port disconnect condition is detected ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91732 91732]
|}
For unknown reasons some types of wireless headsets stop working after some hours or days. Either the port state changes to disabled or the device rejects control commands with a stall response. In both cases the device is reset and restarted now. If even this fails the complete USB host controller is reset and in most cases the device returns to operational state thereafter<br/><br/><!-- ohci.cpp ohci.h usb_hid.cpp --><br/><!---->
   
=== phone: ip222, ip232: Some USB headsets were not detected after a soft reset ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91734 91734]
|}
This was observed with Jabra BIZ 2400 USB at IP222/232 with hardware build 800 and newer.    <br/><br/><!-- ohci.cpp --><!---->
   
=== IP-DECT: Trap with data calls ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91807 91807]
|}
The IP-DECT Radio traps if a data call is released and the release includes a facility.<br/><br/><!--<br/>Changed file:<br/>dectradio_fty_call.cpp<br/>--><!---->
   
=== Kerberos administration: Increase maximum number of Kerberos users from 20 to 50 ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91847 91847]
|}
Increase limit on page General/Kerberos.<br/>Use POST for submitting form, instead of GET.<br/><!--<br/>kerberos_db.xsl<br/>--><!---->
   
=== Voicemail: &lt;pbx-getcallinfo out-leg2-name=".." out-leg2-orig-name=".."&gt; ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91874 91874]
|}
Passing name info of divertingLegInformation2 facility into the script.<br/><!--<br/>pbx/vm/*<br/>--><!---->
   
=== SRTP: Avoid one-way media with high start sequence numbers ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91892 91892]
|}
In some cases SRTP calls had one-way media because the RTP sequence number wrapped from 65535 to 0 at be beginning of the call before the receiver started receiving and processing packets.<br/><br/>The scope of start sequence numbers for RTP streams is changed from [0;65535] to [0;32767] to make sure that the receiver can always receive packets before the overflow happens.<br/><br/>The calculation of the roll-over counter (ROC) is also improved to be more reliable.<br/><br/><!--<br/>media.cpp<br/>cipher_api.h<br/>cipher_api.cpp<br/>--><!---->
   
=== H.323: Unnecessary re-initializing of rtp-channel on incoming calls to phone ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91898 91898]
|}
This did not create any problems except CPU load and together with another problem in RTP it caused no media on incoming SRTP calls approximately every 1000th call.<br/><!--<br/>h323ch.cpp<br/>phonesig.cpp<br/>phonesig.h<br/>--><!---->
   
=== IP22,IP24,IP28,IP302,IP305: RTP-DTMF not offered when using a/b interface ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91905 91905]
|}
For example:<br/><br/>\tv=0<br/>\to=- 14 1 IN IP4 10.17.1.91<br/>\ts=-<br/>\tt=0 0<br/>\tm=audio 16414 RTP/AVP 8 0 18 4 97<br/>\tc=IN IP4 10.17.1.91<br/>\ta=rtpmap:97 CLEARMODE/8000<br/>\ta=fmtp:18 annexa=yes<br/>\ta=fmtp:18 annexb=yes<br/>\ta=ptime:20<br/>\ta=silenceSupp:off - - - -<br/>\ta=sendrecv<br/><br/><!--<br/>ab_sig.cpp<br/>--><!---->
   
=== PBX: MOH URL parameters for parked calls should refer to the parked-to and not parking user ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91931 91931]
|}
The MOH URL Paramter (%l, %h, %n, ...) can be used to use different MOH Files based on the User who is holding the call. In case of a parked call this should refer to the object where the call is parked, not to the user who has initiated the parking.<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== IP-DECT: Wrong XML data in Radio endpoint ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91933 91933]
|}
The XML data in the Radio endpoint is wrong and fixed now. The data is only used for the command show-endpoints, not for the GUI.<br/><br/><!--<br/>Changed file:<br/>dectlocalusers.cpp<br/>--><!---->
   
=== PBX Mobility: Call from mobile endpoint to a user with mobility, but without PBX phone was disconnected ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91976 91976]
|}
This was a collateral damage of<br/><br/>fix: #86813: PBX Mobility: Take precaution against hanging calls because of misbehaving mobile phone or client<br/><br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== SIP/DNS: Wrong port used on secondary SIP server ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=91995 91995]
|}
If SRV query returns 2 hosts with different port, but no IP address in additional records,<br/>SIP starts two A queries for the two host names.<br/>Both resolved IP addresses are combined with the port of the most preferred host of the SRV answer.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
   
=== IP-DECT: Hold/Retrieve could result in no media for incoming SIP calls with SRTP ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92014 92014]
|}
This was in fact a H.323 media negotiation problem between master and slave<br/><!--<br/>h323ch.cpp<br/>--><!---->
   
=== SIP/WLAN: Keep local Contact-URI up-to-date on subscriptions ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92134 92134]
|}
Keep local Contact-URI up-to-date on subscriptions (e.g. message summary) when local ip address changes at runtime.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== IP-DECT: Hold and Retrieve with SIP and SRTP could result in no media ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92189 92189]
|}
Problem with forwarding changed SRTP Keys from master to radio<br/><!--<br/>h323ch.cpp<br/>--><!---->
   
=== phone: ip241: do headset volume control in digital domain ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92203 92203]
|}
<!-- ac_codec3.cpp --><!---->
   
=== IP232,IP222,IP241: Reduce flicker when opening details of call list entry ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92242 92242]
|}
Reduce flicker when opening details of call list entry.<br/><br/><!--<br/>phone_list_ui.cpp<br/>--><!---->
   
=== IP-DECT: Potential trap when switching to an from 3pty conference ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92262 92262]
|}
In case of a very unlikely collision of events, a message was sent to an already deleted object, which cause a restart<br/><!--<br/>comcertp_msp_drv.cpp<br/>+ many more dsp drivers<br/>--><!---->
   
=== IP232,IP222,IP241: Fix for display of international numbers on phone UI ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92335 92335]
|}
Show international numbers as +4930123456 instead of I4930123456.<br/><br/><!--<br/>forms_color.cpp<br/>--><!---->
   
=== PBX: Call Completion to a user without registration, but forking, caused immediate Callback ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92341 92341]
|}
Instead to call completion request should be rejected in this case<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== SIP: Huge SIP messages causes out-of-sync on TCP stream ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92373 92373]
|}
Reading SIP messages from TCP stream gets confused by huge SIP messages.<br/>Presence exchange with external UC was disordered.<br/>Increased size limit from 100KByte to 200KByte.<br/><br/><!--<br/>siptrans.cpp<br/>--><!---->
   
=== Gateway: Transmitting FAX documents to receiver with polling mode ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92388 92388]
|}
It is not possible to send fax documents to a receiver with polling capability. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>t30.cpp<br/>--><!---->
   
=== IP-DECT: Cipher key index table update function ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92467 92467]
|}
The cipher key index table is wrongly updated in the Crypto Master if a entry line yet exists. This is fixed now.<br/>The Crypto Master is needed for DECT Security Early Encryption.<br/><br/><!--<br/>Changed file:<br/>crypto_key_index_table.cpp<br/>--><!---->
   
=== IP222 IP232: Handset gains changed to avoid low microphone volume ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92577 92577]
|}
Low microphones levels were squelched.<br/><!-- ac_codec3.cpp <br/>Gain update nach Empfehlung von Audiocodes:<br/><br/>Dear Carsten,<br/> <br/>First, I can’t recall if there is a reason for you to use the 660.27 patch? Please remind me if there is a reason for not upgrading to version 680.<br/>In this patch version there is a bug in the EC debug mechanism (The Seq. # is not initialized properly). It would be better to upgrade.<br/> <br/>We’re suspecting the cutoffs are due to the new noise-tail remover, which you complained about (the dispersion phenomenon). It can be harm in low gains, so we would like the gain the the EC input to be higher.<br/>Please try another set of gains (In barracks – Current gains):<br/> <br/>Dig. Output gain = -5 dB (-8 dB)<br/>Anl. Output gain = -3 dB (0 dB)<br/> <br/>Anl. Input gain  = +37.5 dB (+31.5 dB)<br/>Dig. Input gain = +2 dB (+8 dB)<br/> <br/>Please let me know if you get any improvement from this above test.<br/> <br/>Thanks,<br/><br/>--><!---->
   
=== Voicemail: Emailing file names with '#'-characters failed ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92757 92757]
|}
An Escape mechanism wasn't applied<br/><!--<br/>pbx/vm_script_if.cpp<br/>--><!---->
   
=== Trap On Ldap DOS Attack ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92978 92978]
|}
Malign asn.1 content wasn't rejected<br/><!--<br/>ldapasn1.cpp<br/>--><!---->
   
=== IP800 IP6000 IP2000: duplicated DTMF digits in transcoding scenarios ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92981 92981]
|}
When tandeming VOIP links for trancoding or other purposes DTFM digits were sometimes duplicated.<br/>The RTP carried up to 25ms DTMF remaining DTMF, now its only 16ms.<br/><br/><!-- AC48xLo.c --><!---->
   
=== PBX: Trap if user object is deleted, which is used by other applications (e.g. myPBX) ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92985 92985]
|}
The applications need to cleanup in this case<br/><!--<br/>pbx_api.h<br/>pbx.cpp<br/>pbx.h<br/>pbx_client.h<br/>pbx_client.cpp<br/>--><!---->
   
=== IP-DECT: CSV user export ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=92991 92991]
|}
There are some missed users in the CSV user export file. It occurs if there are users with login rights. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dectusers.cpp<br/>-><!---->
   
=== Voicemail: Url Parameter To Control Number Processing ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=93043 93043]
|}
$_leg2tweak<br/>--<br/>Controls <pbx-getcallinfo out-leg2=".."/><br/>true(default): set leg2 to <ext-nr> from <vm-nr>+<ext-nr><br/>false: set leg2 according to received divertingInfoLeg2 facility<br/><br/>$_trailhash<br/>--<br/>Controls <pbx-getcallinfo out-cdpn=".."/><br/>true: pass trailing (en-bloc) '#' into cdpn<br/>false(default): don't pass trailing (en-bloc) '#' into cdpn<br/><br/>http://wiki.innovaphone.com/index.php?title=Howto:Configure_the_innovaphone_Voicemail#URL_Query_String_Variables<br/><br/><!--<br/>pbx_vm.cpp/.h<br/>--><!---->
   
=== IP0010,3010,6010,1060,810: limit number of ethernet packets processed per receive interrupt ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=93130 93130]
|}
this prevents that too much time is spent in ethernet driver in case of broadcast storms or DOS attacks.<br/><br/><!-- mv78x00_drv.cpp mv78x00_drv.h mv78x00_drv_s.xsl --> <!---->
   
=== IP-DECT: Call transfer with enbloc dailing ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=93185 93185]
|}
Call transfer with enbloc dailing fails. This is fixed now.<br/>This changes also the R-key handling: after dialling a digit for a consultation call the call must disconnect with R-1 like in ring-back state.<br/><br/><!--<br/>Changed files:<br/>dectradio_call.h<br/>dectradio_call.cpp<br/>dectradio_mastercall.cpp<br/>-><!---->
   
=== SRTP: Remove traces when packet authentication failed ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=93203 93203]
|}
SRTP and SRTCP software encryption produced traces when packet authentication failed. This is not needed, because an event is created anyway, when this happens frequently.<br/><!--<br/>cipher_api.cpp<br/>--><!---->
   
=== SNMP Walk udpTable, tcpConnTable Could fail ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=93225 93225]
|}
The table index returned wasn't always lexically ascending, causing a walk to stop.<br/><!--<br/>udp4.cpp<br/>tcp4.cpp<br/>--><!---->
   
=== phone: ip222,ip232: sporadic boot time trap when a Jabra LINK 280 adapter is plugged ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=93261 93261]
|}
the initialisation fails with CC=5 on first device descriptor read. after restart of host controller serial_irq() traps in reading the done list.<br/>Happens mostly with upload DRAM.<br/><!--<br/>ohci.h<br/>ohci.cpp<br/>--><!---->
   
=== IP-DECT/Relay: Blocked calls by hidden feature code *5/*7 ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=93272 93272]
|}
Outgoing calls with beginning number *5 or *7 are blocked by the feature codes module because of hidden new service codes for an OEM device (#79028). This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dtmffty.cpp<br/>--><!---->
   
=== PBX: Master Slave license update period 10s instead of 10min ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=93330 93330]
|}
For test purposes the period was reduced to 10s, but by accident this change ended up in version 9 as well<br/><!--<br/>pbx.h<br/>--><!---->
   
=== Gateway: CGPN-Maps executed even if the Route did not match in case of enbloc calls ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=93400 93400]
|}
For enbloc calls the CGPN of routes were executed even if the dialed number was incomplete.<br/><!--<br/>gk.cpp<br/>--><!---->
   
=== PBX: Port License counting wrong, when moving users ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=93477 93477]
|}
When moving a user from a master to a slave, the license count on the master was reduced only after reboot.<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>--><!---->
   
=== PBX: XML error on User Interface if slave name with non-ascii characters used and registration was redirected from such a slave ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=93483 93483]
|}
Conversion of UCS2 as received from the registration to UTF-8 on Web UI was wrong<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== PBX: Hide connected Endpoint did nor work for forward to other PBX or Gateway object ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=93513 93513]
|}
Diverting Leg Information facilities or Name Identification facilities were still forwarded in this case.<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== LDAP Client: SearchRequest.derefAliases Changed To neverDerefAliases(0) ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=94812 94812]
|}
Was derefInSearching(1)<br/><!--<br/>ldapdir.cpp<br/>--><!---->
   
=== Gateway: Header line of FAX documents with big endian devices ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=94824 94824]
|}
The header line of FAX documents with big endian devices was not correctly printed. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>t4.h<br/>t4.cpp<br/>--><br/><!---->
   
=== SIP: Port mapping with STUN failed since DNS resolvin of STUN server failed ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=94876 94876]
|}
Port mapping with STUN failed since DNS resolvin of STUN server failed.<br/>Wrong STUN server port was used (5060 instead of 3478).<br/><br/><!--<br/>sip.cpp/h<br/>--><br/><!---->
   
=== H.323: No media after blind transfer to Waiting Queue on other PBX, when operator connects ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=94961 94961]
|}
Happened when different coders where used on caller and called side.<br/><!--<br/>h323ch.cpp<br/>--><!---->
   
=== SIP: Display names need escaping of " and \\ according to RFC-3261 ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=94966 94966]
|}
Acc. to RFC-3261 characters " and \\ (%x22 and %x5C) are to be escaped as "quoted-pair".<br/><br/><!--<br/>sipmsg.cpp<br/>--><!---->
   
=== PBX Waiting: Presence set for operator was not cleared, on delete or editing of Waiting Queue object ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=95016 95016]
|}
When configuration of the Waitinng object is now changed, any presence set by the Waiting object is cleared.<br/><!--<br/>pbx_wait.cpp<br/>--><!---->
   
=== Gateway: FAX interface interop with non-conforming Fax devices improved ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=95029 95029]
|}
There are Fax devices sending wrong (too long) message initially after being called. In this case it is best handled by ignoring these message and wait for the retry instead of disconnecting the call.<br/><!--<br/>Changed file:<br/>t30.cpp<br/>--><!---->
   
=== SIP/UDP: Sending response to wrong address and port ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=95065 95065]
|}
Sending response to wrong address and port.<br/>But only if Via header of incoming request contains domain name.<br/><br/><!--<br/>siptrans.cpp<br/>--><!---->
   
=== IP241: Headset receiver muffled sound ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=95164 95164]
|}
..<br/><!-- ac_codec3.cpp  Headsetvolume wieder analog geregelt, mit der Digitaleinstellugng war der klang dumpf --><!---->
   
=== PBX Exec: Secretary availability monitoring did not work with multiple scretaries, with names starting identically ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=95197 95197]
|}
The availability state (secretary booked into the exec primary group) was not associated with the correct secretary. A compare of the names only covered the first half of the name.<br/><!--<br/>pbx_exec.cpp<br/>--><!---->
   
=== Click sounds at caller side when calling another port of same gateway ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=95436 95436]
|}
occured since V9hotfix5<!---->
   
=== PBX Broadcast: Memory leak when calling busy broadcast object with round-robin config ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=95439 95439]
|}
If all destinations of a broadcast object are busy a name-id facility generates a memory leak.<br/><!--<br/>pbx_bc.cpp<br/>--><!---->
   
=== IP222 IP232 IP241: Use fifo for DSP control channel ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=95763 95763]
|}
try to fix a trap with USB headset <br/><!-- ac_494e.cpp  <br/>The fifo was implemented in fix 63648.<br/>The fifo usage was replaced with a ASSSERT at 20.10.2011.<br/>Now it is enabled again and a DSP error event "Host transmit queue congestion" is logged when a packet cannot be sent immediately.  <br/>--><!---->
   
=== IP232,IP222,IP241: Fix for call status display ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=95769 95769]
|}
If a call was remotely disconneced during HOLD,<br/>the held phone was constantly displaying "held" instead of "disconnected".<br/><br/><!--<br/>forms_color.cpp<br/>--><!---->
   
=== SIP: Interworking issue with "LifeSize Passport/LS_PP1_4.11.9 (8)" ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=95785 95785]
|}
Problems decoding large and complex SDP offer from LifeSize.<br/><br/><!--<br/>sdpmsg.cpp/h<br/>--><!---->
   
=== 802.1x (EAPOL) did not work on interfaces configured for VLAN ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=95931 95931]
|}
802.1x (EAPOL) frames received without a VLAN tag must always be passed to the protocol module, even if the interface is confiured to use VLAN<!---->
   
= V9 Hotfix 22 (9061240) =
Changes included in Version 9 hotfix22
[http://mantis.innovaphone.com/view.php?id=96108 Definition]
 
== New Features ==
 
 
   
=== PBX: New option for RTP Proxy - proxy only if different registration address ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=88439 88439]
|}
To avaoid RTP Proxy for two endpoints located within same private network behind NAT<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_general.xsl<br/>--><!---->
   
=== PBX Executive: Allow monitoring of availability of secondary secretary, don't  treat Exec as secretary ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=95497 95497]
|}
With these two additions a configuration with two executives and two secretaries, each secretary being primary to one executive an secondary to other can be configured with a single group for each secretary and both executives can monitor the availability of both secretaries.<br/><!--<br/>pbx_exec.cpp<br/>---><!---->
   
=== SIP: Workaround for buggy registrar ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96313 96313]
|}
Workaround for buggy registrar.<br/>Different expirtes values in Contact header and Expires header.<br/><br/>\tSIP/2.0 200 OK<br/>\tVia: SIP/2.0/UDP x.x.x.x:2069;rport=2069;branch=z9hG4bK-CDDD130C<br/>\tTo: ;tag=5b8729d5-6f6353c4-cbsxz<br/>\tFrom: <sip:38795988@193.90.37.3>;tag=2109370043;epid=0090331e0bef<br/>\tCall-ID: 7ef63c56e909d311b3890090331e0bef@95.130.221.205<br/>\tCSeq: 1004 REGISTER<br/>\tExpires: 60<br/>\tContact: <sip:38795988@x.x.x.x:2069;transport=UDP>;expires=300<br/>\tUser-Agent: ZTE-SBC<br/>\tX-ZTE-Cause: "SBC-4721-2002"<br/>\tContent-Length: 0<br/><br/>Better apply the smaller expires value.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== PBX: Send forking calls as diverted calls ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96370 96370]
|}
So that on the called side, it will be displayed who forked the call the same way as a diverted call is displayed.<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== PBX: pbx_makecall.txt not only for mobility, but for Waiting Queue also ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96384 96384]
|}
Allows to initiate call with simple web request from a Waiting Queue<br/><!--<br/>pbx_admin.cpp<br/>--><!---->
   
=== IP-DECT: Physical location was wrong after logout/login on handset ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96392 96392]
|}
The physical location information is based on the redirection of the registration from the PBX at the physical location to the registration PBX. Some information was not cleared with the logout, so re-registration startet with the registration PBX right away.<br/><!--<br/>h323sig.cpp<br/>--><!---->
   
=== phone: if a number to be dialled contains a comma, the digits following the comma are sent as DTMF tones after connect ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96402 96402]
|}
This applies to all numbers dialed en bloc, i.e. numbers dialed via indirect dialing, a phone directory or a function key. The comma must not be the first character of the number. <br/><br/><!-- phonesig.cpp --><!---->
   
=== IP-DECT: OEM PBX type info in GUI data ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96723 96723]
|}
Now the GUI data includes the type info of an OEM PBX.<br/><br/><!--<br/>Changed file:<br/>dectusers.cpp<br/>--><!---->
   
=== IP22 IP24 Ip28 IP302: Don't complete media negotiation for ab-interfaces if no media can be sent ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96773 96773]
|}
For incoming calls to a phone media negotiation was already completed during ringing, so that when going off hook the media channel was already established. This causes interop problems, because there are endpoints which asssume there is inband info (e.g. ringback) if media negotiation is complete so local tones (e.g. ringback) were turned off.<br/><br/>In the past with slowstart this premature media negotiation was usefull to avoid delayed media after off-hook. With SIP or H.323 faststart there is no use anymore.<br/><br/>Needed to avoid that the DSP send CLIP and tones at the same time, which can cause sporadic DSP failures.<br/><br/><!-- ab_sig.cpp --><!---->
   
=== SIP: New interop tweak /register-interval ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97834 97834]
|}
New config file option /register-interval 60<br/>Problem is too weired to explain.<br/>This option can be used to set the REGISTER interval to a fixed value regardless of the negotiation.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
   
== Bug Fixes ==
 
 
   
=== PBX: URI dialing, should not be case sensitive and numbers should be possible ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=89326 89326]
|}
needed for federation<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== IP222 IP232 IP241: LCD display is after softreset sometimes out of sync ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=95866 95866]
|}
Displaycontroller needs to be stopped before restart.<br/><br/><!-- orchid_lcd.cpp  --><!---->
   
=== Httpclient: Problems with HTTPS URLs ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96099 96099]
|}
Shortcut to local file I/O did not work fot HTTPS URLs.<br/><br/><!--<br/>httpclient_i.cpp<br/>--><!---->
   
=== phone: do not report "No Media Data received" errors for connections to a recording device ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96102 96102]
|}
some recoding devices, for example ASC never send data on a recording connection<br/><br/><!-- phonesig.cpp,sip.cpp --><!---->
   
=== IP-DECT: Possible no media in case of media renegotiation after handover ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96124 96124]
|}
The problem happened in about 50% of the cases of a media renegotiation which results in a different coder after handover.<br/><!--<br/>dectradio_dectcall.cpp<br/>--><!---->
   
=== IP-DECT: Show release state in IP1202 ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96157 96157]
|}
The release state is not shown in the IP1202. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>config.h<br/>--><!---->
   
=== SIP: Must not answer "refresher=uac" if request contains "refresher=uas" ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96167 96167]
|}
Interop issue with Genband C20 PBX.<br/><br/>Must not answer<br/>    Session-Expires: 400;refresher=uac<br/>in 200/OK, if INVITE contains<br/>    Session-Expires: 400;refresher=uas<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== PBX Trunk: "Outgoing Calls restricted" did not work correctly, Presentation restricted was set, but number could be wrong ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96200 96200]
|}
For example if an analog Gateway was registered to a PBX user, and this Gateway did not send a Calling Party Number with the call, the call was sent with Presentation restricted, but without digits. This could affect Billing Applications which are based on CDRs from the Gateway.<br/><!--<br/>pbx_trunk.cpp<br/>--><!---->
   
=== phone: ip222,ip232: inbound calls automatically connected to Plantronics Savi W440/740/745 headsets with new firmware Versions ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96276 96276]
|}
reported for:<br/>- Savi W440 with firmware 0118 on USB/DECT Dongle D100<br/>- Savi W740/745 with firmware 0115<br/><br/>reason:<br/>the newer firmware versions reject truncated output reports (no trailing 0 bytes) with STALL. The error handling for this case was wrong and caused an autoconnect.    <br/><br/><!-- keys_ip222.cpp, usb_host_if.h, ohci.cpp, ohci.h, usb_hid.cpp, usb_hid.h,  usb_audio.cpp, usb_audio.h usb_msd.cpp, usb_msd.h, ip233_host.cpp, ip233_host.h --><!---->
   
=== IP-DECT: Wrong GK id of standby Master to Mobility Master ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96302 96302]
|}
The standby Master uses a wrong gatekeeper id to register to the Mobility Master. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>dectmaster.h<br/>dectmaster.cpp<br/>--><!---->
   
=== PBX-CDRs: Conference ID missing in CDRs created by Mobility data callback/callthru ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96399 96399]
|}
This caused these calls not to show up in the reporting<br/><!--<br/>pbx_mobility.cpp<br/>--><!---->
   
=== PBX: Unexpected behaviour if too many filter were configured ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96416 96416]
|}
No new filters were accepted without error message<br/><!--<br/>pbx_admin.cpp<br/>servlet.h<br/>--><!---->
   
=== IP22 IP24 IP28 IP302 IP305: Sporadic DSP host interface overruns ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96606 96606]
|}
On the small gateways the DSP hangs if control packet on host interface arrive too fast. Now the rate is limited.<br/><br/><!-- ac_494.cpp ac_494.h  Queue ähnlich wie bei AC494E --><!---->
   
=== License download not working ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96622 96622]
|}
License download stops.<br/>Licenses are not downloaded.<br/><br/><!--<br/>license.cpp<br/>--><!---->
   
=== PBX: Changing of PBX Object Name did not change the name to be used for registration ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96634 96634]
|}
So if the name of the PBX registering as slave was changed as well, it did not register anymore. The PBX object had to be deleted and created with new name.<br/><!--<br/>pbx_admin.cpp<br/>pbx_edit_loc.xsl<br/>--><!---->
   
=== Gateway: FAX interface on IPVA ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96660 96660]
|}
The FAX interface on IPVA can not connect to a remote device because of wrong protocol events. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>facsimile_def.h<br/>--><!---->
   
=== SRTP: One way audio after some minutes on IP6000 IP2000 IP6010 IP0010 ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96673 96673]
|}
Applies to v9hotfix21 on IP6000, IP2000, IP6010, and IP0010.<br/><br/>Collateral damage from fix #91892: SRTP: Avoid one-way media with high start sequence numbers.<br/><br/><!--<br/>cipher_api.cpp<br/>mv78x00_crypt.cpp<br/>--><!---->
   
=== Gateway: CGPN-Maps executed even if the Routing was already completed ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96685 96685]
|}
A CGPN map, which was configured in a route following the route, which was actually executed was executed as well.<br/><br/>This was a collateral damage from fix: #93400: Gateway: CGPN-Maps executed even if the Route did not match in case of enbloc calls<br/><!--<br/>gk.cpp<br/>--><!---->
   
=== IP22 IP24 IP28 IP302 IP305: Sporadic DSP host interface overruns - CLIP disabled, new trace option ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96716 96716]
|}
On the small gateways the DSP hangs in some conditions.<br/>Now the trace-stop is replaced with an Assert to recover from this situation.<br/><br/>To get a trace of this condition a new trace option is added at dsp.xsl, called txt-trace. This traces the DSP message as text, so that they can be read out after a trap.<br/>Typical usage is to enable DSP-trace, DSP control messages DSP data messages and DSP txt trace.<br/>DSP pcm trace and DSP T38 trace should be off to avoid excessive debug load.<br/><br/><br/>Also, the CLIP messages are disabled since they caused problems in the past.<br/><br/><br/><!-- ac_494.cpp ac_494.h ac_dsp3.cpp trace.xsl gk.cpp --><!---->
   
=== H.323: One-way-voice if SRTP call to a Waiting queue is forwarded via Waiting Queue Maps to a phone ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96721 96721]
|}
This is a problem with SRTP key exchange, which could happen in other traffic scenarios as well.<br/><!--<br/>h323ch.cpp<br/>--><!---->
   
=== IP-DECT: Web UI administrator user list removed on IP1202 ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96726 96726]
|}
The Web UI administrator user list is removed on the IP1202 now.<br/><br/><!--<br/>Changed files:<br/>httpfiles.mak<br/>_inn_asc_dect_users_right.xsl<br/>--><!---->
   
=== IP28: Click sounds at caller side when calling ip28 gateway ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96782 96782]
|}
<!--<br/>si3241_drv.cpp<br/>si3241_drv.h<br/>--><!---->
   
=== IP28: Sometime Ringing stopped working on an anlog port. Worked again only after reset. ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96787 96787]
|}
This happened due to low ringing volatge, the default value of "low" on the interface configuration for the ringing voltage caused this to happen.<br/><!-- si3241_drv.cpp --><!---->
   
=== IP22 IP24 IP28 IP302 IP305: Sporadic DSP host interface overruns - Updated DSP code ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96852 96852]
|}
Clip enabled on all channel.<br/><br/><!-- ac_dsp3.cpp ac_dsp3.h ac494.cpp ip24.mak --><!---->
   
=== phone: ip222,ip232,ip241: Local Network Coder default for User-2..6 was G711 instead of G722 as for User-1 ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96890 96890]
|}
WEB-Interface "Phone/User-2..6/General/Options/Local Network Coder" was preset to G711 instead to G722 as for User-1. <br/><br/><!-- phone_user_ui.cpp, phone_user_ui.h --><!---->
   
= V9 Hotfix 23 (9061252) =
Changes included in Version 9 hotfix23
[http://mantis.innovaphone.com/view.php?id=96632 Definition]
 
== New Features ==
 
 
   
=== PBX Waiting: timeout argument for pbx_makecall.txt URL ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97010 97010]
|}
A timeout argument was added to the PBX0/ADMIN/pbx_makecall.txt URL to cancel an outgoing call initiated with this URL.<br/><!--<br/>pbx_admin.cpp<br/>pbx_api.h<br/>pbx_wait.cpp<br/>pbx_wait.h<br/>pbx_xml.cpp<br/>pbx_xml.h<br/>pbx_bc_conf.cpp<br/>pbx_bc_conf.h<br/>--><!---->
   
=== ISDN: Send legacy Redirecting Number for 'old' Fax Servers ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97615 97615]
|}
The redirecting number is an old style information element, which contains part of the information as the diverting leg2 facility. Some Fax Servers do not understand the leg2 facility.<br/><!--<br/>q931.h<br/>q931.cpp<br/>nt_tbl.h<br/>signal.cpp<br/>signal.h<br/>relay.cpp<br/>--><!---->
   
=== PBX-CDRs: Better CDR for pickup ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97681 97681]
|}
A pickup was not indicated in the CDRs<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== PBX-SOAP: Status of Boolean object indicated as local number ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97738 97738]
|}
When monitoring a Boolean object with SOAP a call is indicated. The local number of this call is set based on the status of the boolean object (00 automatic-off, 01, automatic-on, 10 - manual-off, 11 - manual-on)<br/><!--<br/>pbx_bool.cpp<br/>--><!---->
   
=== PBX-SOAP: UserPark allows to park to another object ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97741 97741]
|}
The argument 'cn' was ignored in v9 and earlier, now it can be used to identify a destination for the park.<br/><!--<br/>pbx_xml.cpp<br/>--><!---->
   
== Bug Fixes ==
 
 
   
=== SUBSCRIBE for MWI not correctly handled after change of IP address ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96898 96898]
|}
SUBSCRIBE for MWI not correctly handled after change of IP address<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== phone: ip222,ip232: USB headset echo effects in call setup phase when a G722 call is started by a CTI application ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96974 96974]
|}
The caller hears the calling tones and it's own speech from the headset microphone but the callers speech is not transmitted to the called party.<br/><br/><!-- ac_phone3.cpp -->  <!---->
   
=== H.323: Offered packetization should be honored for SIP interoperability ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=96983 96983]
|}
Problem happend with calls from Samwin CBC<br/><!--<br/>h323ch.cpp<br/>h323ch.h<br/>--><!---->
   
=== RTP-DTMF: Must increase duration field when sending RTP-Event with END marker ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97001 97001]
|}
Must increase duration field when sending RTP-Event with END marker to comply with RFC.<br/><br/><!--<br/>media.cpp<br/>--><!---->
   
=== ASN.1 BER: Decoding of Sequence Member with indefinite length failed ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97072 97072]
|}
An Avaya supplementary service couldn't be decoded<br/><!--<br/>asn1_ber.cpp<br/>--><!---->
   
=== H.323: Channel Close sometimes not sent on hold ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97080 97080]
|}
Esspecially on the second hold within a call the Channel Close was not sent to the party, which put the other on hold. This caused the channel not beeing turned off on this side (the other side receives music on hold in this case)<br/><!--<br/>h323ch.cpp<br/>channel.h<br/>channel.cpp<br/>--><!---->
   
=== IP4 did not work anymore when IP6 was disabled  via WEB interface ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97111 97111]
|}
When the "IP6/ETNx/IP6/Options/Disabled" was checked the IP4 operations did stop after a while.<br/><br/><!-- ip6_ethif.cpp --><!---->
   
=== phone: ip222,ip232: Plantronics Savi W440 dosn't report Talk-Key events in a call established at phone or by a CTI application ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97115 97115]
|}
When a call via this headset was initiated/accepted by the Redial-Key, the Headset(Mode:Control) function key or a CTI application, the call could not be disconnected by pressing the Talk-Key at the headset because the Headset did not report this action.<br/><br/><!-- keys_ip222.cpp --><br/><!---->
   
=== phone: DTMF digits following a comma in a number to be dialed were not handled correctly in some cases ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97150 97150]
|}
- in the "Destination Number" configured under "Phone/Direct Dialing" in conjunction with a nonzero "Autodial Timeout":  the DTMF digits were sent as dial digits<br/>- with a nonzereo "Enblock Dialing Timeout" configured under "Phone/User x/General/Options": sending of DTMF digits was delayed by the configured timeout\t<!---->
   
=== Gateway: FAX interface on IP800/IP305/IP302 ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97571 97571]
|}
Fix for the last fix #96660.<br/>The FAX interface on the IP800/IP305/IP302 can not connect to a remote device because of wrong protocol events. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>facsimile_def.h<br/>--><br/><!---->
   
=== H.323: Fast Unregister/Register operations could lead to failed registrations, in case of fixed signaling ports ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97637 97637]
|}
A listening socket could still be in use. Only happened with IP-DECT and multi-master.<br/><!--<br/>h323sig.cpp<br/>h323.h<br/>--><!---->
   
=== PBX: Name-Id of busy destination was not forwarded to other PBX ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97646 97646]
|}
Display was different when calling a busy phone on local PBX or on another PBX<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== IP22 IP24 IP28 IP302 IP305: ASSERT on DSP queue overrun added ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97653 97653]
|}
<!--  ac494.cpp  --><!---->
   
=== phone: "Prepare Override" function key did not work since V9hotfix21, the overriding source address was ignored ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97665 97665]
|}
<!-- phonesig.cpp --><!---->
   
=== SIP: Trap when terminating a call while re-negotiation is ongoing ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97675 97675]
|}
Trap when terminating a call while re-INVITE is pending.<br/><br/><!--<br/>siptrans.cpp<br/>--><!---->
   
=== phone: ip230,ip240,ip241 :  unefined codes received from a DHSG Headset basestation were misinterpreted as Hookswitch indication ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97732 97732]
|}
sometimes DHSG Headset basestations send codes not defined for DHSG which are silently discarded now. <br/><br/><!-- keys_gen.cpp --><!---->
   
=== H.323: Potential Trap when reconfiguring an H.323 registration ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97820 97820]
|}
This is a collateral damage from<br/><br/>91815: IP-DECT: Static ports between Master and Radio<br/><!--<br/>h323sig.cpp<br/>--><!---->
   
=== PBX Executive: Calls with calling id restriction and without calling id, were sent to executive, even if secretary available ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97897 97897]
|}
This happend for calls coming in from public ISDN with calling id presentation restriction thru a trunk object without number<br/><!--<br/>pbx_exec.cpp<br/>--><!---->
   
=== SIP: Trap when using STUN ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97898 97898]
|}
Trap when using STUN.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== SIP: Dialog-Info was encoded with wrong state attribute ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=97926 97926]
|}
Dialog-Info was encoded with wrong state attribute "full".<br/>Must be "partial".<br/><br/><!--<br/>sip_dialog_info.cpp<br/>--><!---->
   
=== SIP: Logging was wrong ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98037 98037]
|}
Logging was wrong.<br/><br/><!--<br/>sip.cpp/h<br/>--><!---->
   
=== PBX: Potential trap when receiving unknown presence activity ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98043 98043]
|}
In the respective version unknown activities are mapped to "busy"<br/><!--<br/>fty.cpp - rollback of this change<br/>h450.cpp<br/>--><!---->
   
= V9 Hotfix 24 =
Changes included in Version 9 hotfix24
[http://mantis.innovaphone.com/view.php?id=98531 Definition]
 
== New Features ==
 
 
   
=== IPVA: ETHx Transmit Queue Size Limited To 1MB ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98187 98187]
|}
Was unlimited<br/><!--<br/>vlance_drv.cpp<br/>--><!---->
   
=== Linux: Shutdown warning message ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98438 98438]
|}
If Linux is running, a shutdown warning message is shown at the Linux General page now.<br/><br/><!--<br/>Changed file:<br/>linux.xsl<br/>--><!---->
   
=== SIP/TLS: Using domain name as fall back to proxy name when comparing to the certificat "subject" ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98643 98643]
|}
Using domain name as fall back to proxy name when comparing to the certificat "subject".<br/><br/><!--<br/>siptrans.cpp<br/>--><!---->
   
=== Voicemail: Allow SMTP Email Port Different Than tcp/25 ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98683 98683]
|}
eg in email.xml: ..smtp.foo.bar:888..<br/><!--<br/>smpt_mta.cpp<br/>--><!---->
   
=== phone: directory search highlights first matching entry if the search expression contains any non numeric digit ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98806 98806]
|}
To prevent unintended dialing of a directory entry starting with numeric digits the search expression was checked if it consists of dialable digits (0-9*#,) only.<br/>In this case the first matching entry was not automatically highlighted (activated) so that the input (number) could be dialled by going off-hook. <br/>Now the check includes the numeric digits (0-9) only.  <!---->
   
=== SIP: New interop tweak /accept_any_reg_interval ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98887 98887]
|}
For endpoints that do not read the expires value from 200/OK response to REGISTER.<br/>E.g. "User-Agent: TRBOnet.Enterprise"<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
== Bug Fixes ==
 
 
   
=== IPVA: Unused ETH1 Could Cause Out-Of-Memory Situation ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98179 98179]
|}
Outgoing packets could queue up without ever getting purged.<br/><!--<br/>vlance_drv.cpp<br/>vlance_drv_s.xsl<br/>--><!---->
   
=== SIP: Trap - not checking array boiundaries ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98219 98219]
|}
Trap - not checking array boiundaries when processing User-Agent header.<br/><br/><!--<br/>sipmsg.cpp<br/>--><!---->
   
=== SIP: STUN not working ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98223 98223]
|}
STUN not working if STUN server IP address is configured.<br/>No problem if if STUN domain name is configured.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== phone: ip222,ip232: USB connection sometimes lost until reboot ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98290 98290]
|}
Observerved with with headsets from various manufacturers. Most probably caused by strong electrostatic discharges to the USB connection cable. In such case also unplugging/plugging of the headset was not noticed by the driver anymore.<br/><br/><!-- ohci.cpp ohci.h --><!---->
   
=== Logging: "Alarm and Event Forward Server" address could not be changed anymore once configured ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98457 98457]
|}
<!-- logging.xsl --><!---->
   
=== LDAP Replication: ASN.1 Decoding Failed For More Than 100 Attributes Per Object ===
 
{|
|Status
|<font><font color="green">Closed</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98468 98468]
|}
Happend with 100 attributes in a PBX stored phone configuration.<br/><!--<br/>ldapsrv.cpp<br/>ldapdir.cpp<br/>--><!---->
   
=== Corrupt trap buffer when tracing UART messages ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98533 98533]
|}
<!-- uart_drv.cpp --><!---->
   
=== IP232,IP222: Confirming blind transfer with redial key did not work ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98591 98591]
|}
Short user guide:<br/><br/>  Press 'redial' before accepting the call, enter telephone<br/>  number for the call diversion and confirm by pressing 'redial' again.<br/><br/><!--<br/>app_ctl.cpp<br/>--><!---->
   
=== phone: ip222,ip232: USB headset media connection lost after a release received from a remote conference peer ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98600 98600]
|}
Happened only on a release of the call which was the active call when the conference was established. The remaining VOIP connection was OK but the media stream was not passed from/to headset anymore.<br/><br/><!-- ac_phone3.cpp ac_codec3.cpp --> <!---->
   
=== IP241: DHSG Headset messages are sometimes wrong ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98642 98642]
|}
<!-- orchid_lcd.cpp orchid_lcd.h config.h --><br/><br/>Beim Neuladen des Sequencers bei Änderung der LCD-Helligkeit kommt der UART-Takt gelegentlich zu schnell. Besser noch wäre 2 sequencerprogramme zu definiere, aber wie das geht ist nicht im Orchid/Titan Usermanual nicht offensichtlich --><!---->
   
=== SIP: SDP version not increased when answering an offer where only media-mode has changed ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98739 98739]
|}
If remote side changes from 'sendrecv' to 'inactive'<br/>the SDP answer follows this change of media-mode,<br/>but SDP version was not increased.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== Gatway: Configuration of Blockdial Timeout at Routes did not work ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98748 98748]
|}
Wrong value was calculated, if multiple maps were used in a single route blockdial timeout configuration of a map was lost, when another map was configured.<br/><!--<br/>gk.cpp<br/>--><!---->
   
=== enabled state of an external directory configured via a PBX config template was lost in some cases, ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98816 98816]
|}
This did happen for example when<br/>- a second phone was registered to the same PBX user (twin phone)<br/>- a "Phone/Reset/Reset User Specific Configuration" was done via the phones WEB GUI (but not when this was done via the PBX GUI)<br/><br/><!-- phone_edit.cpp phone_user.cpp phone_user_ui.cpp --><!---->
   
=== SIP: Trap when sending &lt;dialog-info&gt; ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98902 98902]
|}
Trap when interworking group-indications into dialog-info.<br/><br/><!--<br/>sip.cpp<br/>--><!---->
   
=== Memory Leak when deleting voicemails ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98929 98929]
|}
A list wasn't cleaned<br/><!--<br/>pbx_vm.cpp<br/>vm_script_if.cpp<br/>test\\9.00\\pbx\\vm\\vm-box-to-box\\vm\\cf-files\\vm-listfiles.xml<br/>--><!---->
   
=== IP222 IP232: Noise in the microphone at 6400Hz ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98941 98941]
|}
.<br/><!-- ac_codec.cpp platform/filter/ip222_mic_*.* --><!---->
   
=== IP222 IP232 IP241: Codec register debugs added ===
 
{|
|Status
|<font><font color="red">Assigned</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98972 98972]
|}
Codec register debugs are enabled with dsp  trace.<br/>Codec register are dumped after changes to analyse Manits 97903 <br/><!-- ac_codec3.cpp --><!---->
   
=== PBX: No config updates were sent to the phones, when selection of templates was changed ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98983 98983]
|}
When the config of the template itself was changed an update was sent, but if it was changed which templates were used on a user object, no update was sent.<br/><!--<br/>pbx.cpp<br/>--><!---->
   
=== Phone: Could not configure fkey labels containing single quotation mark ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98986 98986]
|}
Could not configure fkey labels containing single quotation mark.<br/><br/><!--<br/>phone_edit.cpp<br/>--><!---->
   
=== myPBX: Missed calls for mobility calls accepted somewhere else ===
 
{|
|Status
|<font><font color="orange">To-decide</font></font>
|-
|Id
|[http://mantis.innovaphone.com/view.php?id=98995 98995]
|}
A cause code was missing in CDRs generated by mobility<br/><!--<br/>pbx_mobility.cpp<br/>--><!---->

Revision as of 00:31, 19 April 2013

This is the Firmware V9 Roadmap Document.

The release date of the next Hotfix is planed for the second monday of a month. Please note that this a scheduled and no fix date.

This article is generated automatically. Do not edit! Please see the disclaimer before using the information presented here!


V9 Release

This is the Version 9 Release. It is expected to be released 2011, Week 17. Definition

New Features

Other new Features

57648 dde My.innovaphone: interface for downgrade/upgrade licenses
14925 gd LDAP over TLS
37992 gd Boolean to check for day/month/year also
52926 gd Use of SIP Dialog Info for Group Functions across PBXs
59130 msc HTTP-Client: Support for proxy server
61702 msc X.509: Certificates with domain names and IP addresses
61790 tac G.722 negotiation in SIP signaling
36001 gd IPv6
61837 gd SIP: Send Alert-Info in INVITE
8495 cmu ip21/22/24/302 TFE
50332 dde Reporting
26947 msc myPBX
52956 queue PBX: Password for unknown regs
45803 jfr DHCP POSIX timezone option id is 100 officially (since 2007), let server provide it if explicitely requested
59131 msc HTTP-Client: DNS resolution
59132 msc HTTP-Client: Support for HTTP redirect
54937 tac Support for multiple presences on phones
60088 tac SIP: Support for Event "talk"
60092 tac SIP: Support for Event "conference"
60537 tac SIP: Put "alias" parameter into Via header of REGISTER request (RFC-5923)
62741 tac SIP: Get display information from Call-Info header in connect response
63186 tac SIP: Get display information from Call-Info header in register response


Bug Fixes

V9 Hotfix 3 (90600.03)

Changes included in Version 9 hotfix3 Definition

New Features

SIP: Send OPTIONS at configurable interval

Status Closed
Id 67519

For keep alive pruposes on interfaces without registration.
Required for Lync interoperability.
(config change TSIP /options-interval 30) Status: sip.cpp/h
siptrans.cpp/h

SIP: Fast re-routing on gateway interface w/o registration

Status Closed
Id 67593

OPTIONS is used to check availability of remote peer.
If OPTIONS fails, not trying to send INVITE.

SIP: New config file option /no-ms-acceptedby

Status Closed
Id 67665

Some IP phones get confused by this Microsoft extension.
Cisco SPA 303 and Cisco SPA 962 don't stop ringing anymore

ip200a/230/240: handset conversations can be monitored in a directly connected headset

Status Closed
Id 67666

This feature is required for a special application and is supported only for ip200a/230/240 phones with a directly connected headset (non DHSG).
It is enabled via
config add INCA_DSP /handset-spy <volume>
whith <volume> in the range from 1..8

PBX: Translation of Cause "Call Rejected" to Cause "User Busy" for endpoint objects only

Status Closed
Id 67668

If the cause is received from a gateway, it is forwarded transparently now

myPBX: Show active sessions on the admin interface

Status Closed
Id 67708

The active myPBX sessions are shown on the page PBX/myPBX.

Phone: Added new function keys to Phone-UI (Toggle &amp; Prepare Override)

Status Closed
Id 67729

Description: Phone: Added new function keys to Phone-UI (Toggle & Prepare Override)

PBX: New presence activity "do not disturb"

Status Closed
Id 67777

If a users sets this presence activity no calls are passed.

PBX: Allow CFB on Gateway Type Objects

Status Closed
Id 67827

A CFB is triggered by a User Busy. If a CFB is used for example at a Trunk, the CFB is executed when the called remote user returns busy. Because this may be unexpected the CFB was not executed at a Gateway Type Object.

It is now enabled again, because it is useful when connecting external systems which return busy to indicate an out of channels situation

PBX Trunk/Gateway: Round robin within registrations to same device, different devices sequentially

Status Closed
Id 67835

This way both round-robin or sequential usage of gateways can be configured

improved test for Timeslot Switch Chip of ip6010 ip3010 ip0010 ip1060 and ip6000

Status Closed
Id 67840

intension is better analysis of hardware problems Status: idt72_drv.cpp, idt72_drv.h

ip22/24/28/302/305/6010/3010/1060/241/222/232: False DTMF detects

Status Closed
Id 67844

Seen on IP222 and IP6010 Status: ac_dsp3.cpp

phone_orchid: builtin color display test

Status Closed
Id 67845

After the builtin test function has been started the display test mode is entered when the 'Esc' key is pressed. Numeric keys trigger a full screen test display, all other keys stop the display test mode.
To the keys 0..9 the following patterns are assigned:
DarkGray, White, Grey, Black, Red, Green, Blue, Yellow, Cyan, Magenta

IP-DECT: Configuration XML data for OEM device

Status Closed
Id 67858

A new configuration XML attribute is added for changing a OEM GUI.

Phone: "Function keys not modifiable on the phone" mask should disable creation of new function keys of masked type

Status Closed
Id 67905

Description: Phone: "Function keys not modifiable on the phone" mask should disable creation of new function keys of masked type. Currently, only modification of preset function keys is disabled, but the creation of new ones enabled and possible.

Phone: Added command line option to hide Administration Menu and/or MAC/Serial completely

Status Closed
Id 67943

Description: Phone: Added command line option to hide Administration Menu and/or MAC/Serial completely. See /hide-mask option to PHONE ADMIN-UI in wiki for more information.

IP-DECT: OEM registration string

Status Closed
Id 68058

The Radio registration string of an OEM device is changed to check a OEM license in the DECT Master.

SIP: Diagnostics of transport error "Remote server certificate mismatch"

Status Closed
Id 68142

If remote sverer certificate does not match the destination domain name
signaling connection is refused by client.

myPBX: Support for WebEx meeting passwords

Status Closed
Id 68172

In WebEx a meeting can have a password that must be entered by the attendees when they join. Some WebEx accounts can only create meetings with passwords.

The possibility to configure a global meeting password is added to the PBX/Config/myPBX page.

Phone: Message function key

Status Closed
Id 68208

Description: Phone: Message function key. Multifunctional depending on number of unread messages. Stores one prepared message (with destination and message text) and presents the new message screen when invoked. If incoming messages pending, display the letter/message icon and jump to incoming-messages subscreen upon invocation.

phone: Finnish translations updated

Status Closed
Id 68395

no english placeholders anymore, Texts begin with upper case letter

IP150: dimming of key LEDs and LCD backlight

Status Closed
Id 68414

to save power in special environments the key LEDs can be dimmed by
config add KEYS0 /light-off
The lcd backlight can be configured the usual menu way on the phone.

HTTP-Client: Allow user names longer than 16 characters

Status Closed
Id 68499

Now user names with up to 64 characters are allowed.

Phone: Call forwarding (always, busy, no reply) destination now choosable from dial-menu

Status Closed
Id 68504

Phone: Call forwarding (always, busy, no reply) destination now choosable from dial-menu. Usage: enter number or search for phonebook entry, press menu-key, scroll down to choose call-forwarding (always, busy or no-reply) and acknowledge choice in CF-screen.

IP-DECT: Allow setting empty text for idle display

Status Closed
Id 68553

Some handsets will not show signal strength and battery symbols if idle display is defined, they must be set to empty string.

ac_dsp3: support echo canceller trace

Status Closed
Id 68649

required to analyze echo canceller problems

Phone: On IP240, make OK key a headset activation key (along with Space Key)

Status Closed
Id 68774

Phone: On IP240, make OK key a headset activation key (along with Space key).
Enabled from Phone/Preferences : Use Newline/OK Key as Headset Key.

Bug Fixes

IPxx10: error handling in sata driver

Status Closed
Id 67229

Old cards are producing DMA errors that were not handled properly. Try again read/write operation after error recovery.

DECT: IP6000/IP6010/... default config Master mode off

Status Closed
Id 67479

Now the Dect Master is in mode off by default for the IP6000/IP6010/...

SoftwarePhone: DTMF to voice mail

Status Closed
Id 67563

Now, sending DTMF with myPBX is correctly working. This fixes sending DTMF to voice mail.

VM: Trap while processing self-forwarded call

Status Closed
Id 67570

VM: Trap while processing self-forwarded call

SIP: Uninitialized data in SDP offer/answer

Status Closed
Id 67617

Applies to G.726 exclusive calls only.

Phone: Main menu scrolling below last item broken

Status Closed
Id 67618

Phone: Main menu scrolling below last item broken.
1st item hould be activated upon down arrow press (done) and screen focus moved up (not being done - bug).

cpld update not working on ip241

Status Closed
Id 67629

- Status: phone_orchid.cpp phone_orchid:config.h xilinx.cpp
and as consequence also ip6010:config.h

SIP: Interoperability with Lync and media-bypass

Status Closed
Id 67645

Ack contained wrong To-Tag when calling a lync client in media-bypass scenario.
Results into call drop after 30 seconds.

Phone: Automatic key repeat did not work on IP222/IP232

Status Closed
Id 67661

Automatic key repeat did not work on phone w/o alpha keyboard

Call to Voicemail did not work anymore with SRTP

Status Closed
Id 67672

This was a collateral damage from fix

http://wiki.innovaphone.com/index.php?title=Support:DVL-Roadmap_Firmware_V9#Allow_configuration_of_SRTP_crypto_suite.2C_to_be_used_for_media_proposals

Phone: long function key titles hide idle screen information

Status Closed
Id 67681

Description: Phone: long function key titles hide idle screen information.
Fixed: important idle screen information now shortens the amount of displayed function key name. Following information is now displayed over the function key text: a) crossed bell icon on do-not-disturb (lines 2+3) b) CFU + CFU-destination (lines 2+3) c) missed calls, unread messages and waiting callbacks (line 4)

PBX Waiting: Call forwarded with DTMF mapping was shown in myPBX for each registration

Status Closed
Id 67682

The call was not sent with the original conferenceID, so myPBX could not detect that it was in fact the same call

PBX: License accounting in centralized licensing scenario wrong if master not available

Status Closed
Id 67698

When the master is available the slave stores the licenses from the master including the usage. This stored usage included the licenses used by the slave itself, so if after a reset the master was not available the local usage just added to this.

Now from the stored usage the local usage is subtracted.

phone_orchid: microphone is not mute on a call intrusion in silent monitoring mode / microphone cannot be muted in a conference

Status Closed
Id 67704

On a call intrusion in silent monitoring mode the microphone of the intruding party must be mute. In a conference the micro should be muted when the micro key is pressed and unmuted when the micro key is pressed again.
Muting the microphone did work when only one call was active but not when two calls were active as in a intrusion/conference.

Phone: CLIR on text messaging did not work

Status Closed
Id 67710

CLIR on text messaging did not work

SIP: Interoperability with Lync: Handling of REFER from Lync

Status Closed
Id 67713

REFER for blind transfer was rejected with "406 Not Acceptable",
due to absence of user part in refer-to URI.

PBX Trunk: Problem with Forking to trunk if multiple GWs are registered to Trunk

Status Closed
Id 67720

If one of the gateways rejected the call (no channel, not connected, ...), the original call from which was forked was disconnected

Phone: Calls received with CLIR appear in call list with an empty entry

Status Closed
Id 67722

Description: Phone: Calls received with CLIR appear in call list with an empty entry; now fixed to display the CLIR text: anonymous/unbekannt/...

PBX: License accounting did not work with Unknown Registrations under some special conditions

Status Closed
Id 67731

It could happen that a registration to a user was not accounted for if the endpoint used for this already had an unknown registration at the time the user was created

User Interface: Alignment of tables fixed

Status Closed
Id 67734

Some strange alignments were introduced with the last hotfix

Gateway: Interface Name with multiple "Umlaute" did not work

Status Closed
Id 67737

There was a length limitation of the URL encoded output, which was already exceeded if three "Umlaute" (or any character which is encoded in more the one byte with utf-8), were used

IP-DECT: Packetization could change after handover

Status Closed
Id 67738

On the new radio the RTP should be sent using the same packetization as was negotiated with the original call

SIP: Fix for early media from Waitng Queue

Status Closed
Id 67775

PROGRESS after ALERT was not handled by SIP stack.
Now 183 Session Progress with SDP is send after 180 Ringing w/o SDP.

Leak checking improved

Status Closed
Id 67783

sometimes leaks were falsely detected. Problem if objects are about to be deleted, which were not owned by any module anymore. This happend esspecially with httpclient.

SIP: Generate/add SRTP key on media-relay interfaces

Status Closed
Id 67789

This enables even partial SRTP (SRTP on one side of media-relay).

Phone: Enable "Activate Registration" without user/password authentication if "Protect Configuration at Phone" set

Status Closed
Id 67791

Description: Phone: Enable "Activate Registration" without user/password authentication if "Protect Configuration at Phone" set. Activating a registration is a state change, and not a configuration modification, so allow this option.

H.323: A name_id of length 0 resulted in invalid H.450 coding

Status Closed
Id 67796

An empty name identification received was forwarded in H.323 as invalid H.450. Such a name is now forwarded as 'name not available'.

H.323 Malformed packet

Status Closed
Id 67803

The ASN.1 encoder had a bug under one special condition: For a constrained character string with a maximum length of more or equal to 16bits, with an effective length of zero, the padding for octett alignment was missing for the zero length bitfield containing the string.

In H.323 this only happens for the CallIdentity used for H.450 call transfer message in case of blind transfer without consultation.

This fix breaks compatibility with earlier versions, for this reason this fix is available for version 9,8,7 and 6.

If phones and PBX with versions containing and not containing this fix are mixed the following problems will occur:
- A blind transfer without consultation (initiated with the redial key) is not possible
- A call which was transfered without consultation is not displayed at the transfered-to phone as transfered

SIP: Unwanted media-relay sessions when using forking/broadcast/multi-reg

Status Closed
Id 67819

If in incoming SIP was routed to multiple destinations
the final session could be media-relay although not configured.

ip22/24/28/302/305: DSP affected by certain fax tones

Status Closed
Id 67821

Brother fax problem, the first fax is transfered, the next fax transfers fail.
Switch to fax from remote is now done without reopening the channel.
Closing the channel waits until t38 is switched off.
Status: ac_dsp3.cpp ac_dsp3.h

IP30x, IP1060, IP2010, IP6010: Fax did not work if rerouted from ISDN interface to a Voip destination

Status Closed
Id 67823

DSP was not configured correctly

IP2x2: Handset microphone gain increased

Status Closed
Id 67843

changed from 22.5db to 30db Status: ac_codec3.cpp

IP241: Remote party information truncated more than necessary

Status Closed
Id 67872

Line 3 of call ctrl

IP-DECT: Changed channel option SRTP crypto suite

Status Closed
Id 67904

Now DECT system channel configuration option 'Secure RTP' is a drop down box. The DECT Master correctly transmits the changed option. This feature was changed in V9 Hotfix 2, related case #66810.

SIP: DNS problem when SRV response provides no additional records

Status Closed
Id 67907

If 2-step resolving is required (SRV and A) the service port
of the SRV response got lost and default SI Pport 5060 was used.

SIP: Trap when configuring STUN server on a SIP/TCP or SIP/TLS interface

Status Closed
Id 67923

STUN is for SIP/UDP only.

myPBX: Always send an ID with CT_INITIATE

Status Closed
Id 67932

This is needed for interoperability with phones using v9hotfix2 or older. (See #67803)

SIP: Must answer every request - even unknown/unsupported methods

Status Closed
Id 67935

Lync sends proprietary NEGOTIATE request and waits for response.
Otherwise transport connection is blocked for any upcoming request.

PBX Waiting: A call rejected on a primary operator was never sent to all operators

Status Closed
Id 67945

if again and again rejected on all primary operators. Problem happend also if rejected by "Do not disturb" on operator phone.

PBX: Master/Slave compatibility problem with version 9 and version 8 and non-ascii characters in PBX name

Status Closed
Id 67956

In version 8 only latin1 characters were allowed, which means in unicode the high byte was always 0. So it could be ignored and when sending location information between master and slave sometimes the high byte contained 0xff.

In version 9 this non-ascii location information was not correct unicode at all.

The problem happened only if non-ascii characters were used when naming a PBX.

SIP: Generate new SRTP key on every incoming re-negotiation

Status Closed
Id 67958

Required to have different encryption after transfer.

PBX-CDR: Mobility calls to Trunk, external number was missing in CDR

Status Closed
Id 67978

only the number of the trunk itself was available

IP150: OEM specific WEB GUI modifications did not work

Status Closed
Id 67992

manufacturer specific stylesheets have to be be adjusted to the new GUI style

PBX: End of call intrusion was not signaled to the phone

Status Closed
Id 68007

The call intrusion tone was generated even if the intrusion was terminated

PBX: User Interface Inconsistency. Same thing was sometimes call 'Response Timeout'/' and 'No Response Time'

Status Closed
Id 68008

Only Response Timeout used anymore

PBX-SOAP: Support UserClear for pending outgoing calls on Waiting Queues

Status Closed
Id 68009

UserClear was ignored in this state

SIP: Fix for media negotiation with SRTP

Status Closed
Id 68067

Exchange of SRTP key may fail after hold/retrieve

phone_orchid: DTMF Tones detected in voice data from microphone were propagated to remote side

Status Closed
Id 68075

DTMF tones may be detected from audible feedback on pressing a dial key in connected state and also from some other source. It's better to propagate only tones requested explicitely via a dial key and not from some external source.

SIP: Trap handling 491 response on reliable transport

Status Closed
Id 68093

Trap with MAX_BUSY_TICKS

phone_inca: "ETH0/Isolate PC Link" checkmark could not be cleared via WEB UI once set

Status Closed
Id 68098

Only a WEB UI problem, a "config rem ETH0 /isolate-pc" did help.

SIP: Trap when outgoing SIP call is cancelled while DNS resolving is ongoing

Status Closed
Id 68099

Trap when outgoing SIP is cancelled while DNS resolving is ongoing

Gateway: Allow configuration of username and password for ENUM/SIP interfaces

Status Closed
Id 68147

For rare where remote destination server asks for authentication.
(And all remote destination servers ask for same auth or remote destination server s always the same.)

SIP: Interoperability with LinkSys SPA3102

Status Closed
Id 68174

LinkSys SPA3102 gives "g729a" as RTP payload type mapping:

v=0
o=- 510843041 510843041 IN IP4 192.168.10.20
s=-
c=IN IP4 192.168.10.20
t=0 0
m=audio 16404 RTP/AVP 18 100 101
a=rtpmap:18 G729a/8000
a=fmtp:18 annexb=no
...

Needs to be handled.

IP241: Fix for display rendering

Status Closed
Id 68181

Display of diversion destination was corrupt.
Pickup fkey labeling was wrong when party information needed truncation.

Gerneral/Admin page was broken if too many authentication servers were configured

Status Closed
Id 68231

The number of authentication servers is now restricted to 10.

SIP: Interoperability with Lync

Status Closed
Id 68232

Media negotiation problem on calls coming from on Lync client
and getting forwarded to another Lync client.

IP241: Updated some display text information

Status Closed
Id 68234

No need to strongly abbreviate disconnect cause text.

Phone: Added loud note to web-ui that bool funnction key can only toggle if boolean object addressed by number

Status Closed
Id 68240

Description: Phone: Added loud note to web-ui that bool funnction key can only toggle if boolean object addressed by number

phone: intrusion call started in handset mode is not terminated when going on hook when TAPI or operator run on PBX

Status Closed
Id 68249

With TAPI or operator running on the PBX the the signaling of a busy condition is changed such that a disconnect instead of a release is sent. The disconnect was not handled correctly, the hookswitch state was lost and the next on-hook signal was ignored. TThus teh call could be terminated with the disc-key only.

Phone: Switch presence fkey light on if presence activity is set

Status Closed
Id 68256

Switch presence fkey light on if presence activity is set

myPBX: Wrong charset was used when authenticating against reporting or changing user password

Status Closed
Id 68258

Javascript uses Unicode. The PBX and the reporting use UTF8. Therefore strings have to be converted to UTF8 before encrypting them using RC4.

IP241: New fkey backgrounds

Status Closed
Id 68304

Less transparency to avoid interference with personalized background images

myPBX: Chunked response from application sharing provider did not work

Status Closed
Id 68311

After receiving HTTPCLIENT_RECV_RESULT, pbx_client did not send another HTTPCLIENT_RECV.

phone_orchid: call pickup via partner/pickup key fails when the key is pressed immediately after audible signal

Status Closed
Id 68313

When "Audible Signal after alerting" was configured on a partner/pickup key and the key was pressed while or a short time after the audible signal was played then the picked call was mute. Status: files: ac_dsp3.cpp

IP241: Activate external background image from phone menu

Status Closed
Id 68351

External background image source can be configured on web ui.
Background image can be selected on phone menu.
Now also external background image can be selected.

myPBX: Hide passwords for application sharing and reporting in config

Status Closed
Id 68352

The passwords are moved from the config line to the VARS.

PBX0/MY-A - Application sharing password
PBX0/MY-R - Reporting password

SIP: Failed to register on dynamic PBX

Status Closed
Id 68362

Attempt to register on a dyn PBX with SIP protocol was rejected with "301 Moved Permanently".

IP-DECT: Adding OEM radios to Kerberos realm did not work with passwords containing special characters

Status Closed
Id 68377

The password was not URL-decoded when reading it from the UI.

SIP: Pending control calls on gateway

Status Closed
Id 68378

Incoming unsolicited NOTIFY(message-summary) may cause pending control call on Gateway.
Control calls are calls (signaling connections) without media channel.
These calls are now released.

WEB GUI page cannot be scrolled completely when height of left hand logo is too big

Status Closed
Id 68382

Height of FHF logo is bigger than that of the default logo, this must be considered when computing size for iframe below tab lines.

DTMF user configuration with invalid checkbox check for presence setting

Status Closed
Id 68383

The check of the checkmark of the presence setting was wrong.

IP222: Alpha input using the num block

Status Closed
Id 68398

Automatic switching back to numeric mode when leaving input ctrl.

SIP: Trap when using TLS as transport

Status Closed
Id 68410

Only if remote side closes transport connection while requests are pending.

phone_orchid: begin of voice mail prompt was cut off sometimes

Status Closed
Id 68416

Connection of Voip to DSP channel was delayed and thus the first packets of the RTP stream were lost.

body onload attribute can be extended and tab_active method doesn't crash anymore

Status Closed
Id 68430

Custom PBX object XSL had no method anymore to set the onload attribute of the body. This can be now extended with a XSL template parameter.
Additionally the tab_active method has been called by default and the default value caused the method to crash.

X509: Fix for reading innovaphone info from flash

Status Closed
Id 68435

Parsing the innovaphone info text was incorrect

License: Be safe against factory reset during license invalidation

Status Closed
Id 68447

If factory reset is done before license invalidation procedure is complete,
will keep you from completing the license invalidation.
Now the procedure can be completed even after factory reset.

IP241 - handsfree speaker volume to low

Status Closed
Id 68451

The handsfree speaker volume was too low even when configured to maximum. Now the general output volume is increased by 3 dB. In case of problems the general output volume can be changed by
config add AC-DSP0 RINGER /VoiceOutputGain n
with n = 1..63 -> (-32 + n)db, n = 32 -> 0dB, n = 0 -> mute

wrong link to PPP Interface State Info Help

Status Closed
Id 68463

Link out of V9 namespace

phone: DHSG headset not reset to idle after a hookswitch signal in idle state

Status Closed
Id 68567

most DHSG headsets generate a hookswich signal and enter voice mode when taken out of basestation. This hookswitch signal was simply ignored.
Now the voice mode is cleared after one second if there is no other DHSG event before.

SIP/TCP: Transport error when connection is closed by client

Status Closed
Id 68578

If transaction client closes connection before final response has been sent,
the server tries to open a new connection toward ephemeral port of closed connection.

SIP: Fix for Dialog-Info notification

Status Closed
Id 68581

Send an empty dialig-info XML after inbound subscription.
Required for interop with Grandstream GXP2010.

SIP: Use secondary proxy address on interfaces without registration

Status Closed
Id 68635

When secondary proxy address resolved by DNS it should be used to send INVITE to.

SIP: Handling of 491 response on reliable transport

Status Closed
Id 68652

Re-try of re-INVITE was missing.

SIP: Treat domain part of SIP URI case-insensitive

Status Closed
Id 68653

According to RFC3261(19.1.4)

SIP: Fix for REFER handling

Status Closed
Id 68663

Support for attended transfer between two calls from/to different SIP endpoints.
As long as both calls belong to the same signaling interface.

SIP: Problem decoding INFO(application/dtmf-relay)

Status Closed
Id 68667

DTMF digit was not decoded from message body if whitespace between EQUAL and DIGIT.
E.g. Signal= 5

Phone: Changing config option /sip-hold does not call for reset

Status Closed
Id 68691

Reset is required and 'reset required" must be displayed.

SIP: Web UI for cause code mapping (fix)

Status Closed
Id 68772

Did not work as expected

SIP: Out-Of-Memory trap if Group Indications are enabled

Status Closed
Id 68804

If 'Group Indications' are enabled on a PBX object where a SIP client registers,
an OOM trap may occur in case of heavy call activity on PBX.

Phone: "Keep Calling Party Info on Pickup Key" option from Phone->Preferences not working, fixed

Status Closed
Id 68820

Phone: "Keep Calling Party Info on Pickup Key" option from Phone->Preferences not working, fixed

Phone: DnD Absence Message containing newline dumped to config-file breaks phone upon config-file upload

Status Closed
Id 68898

Solution: All flashdir entries cotaining newline now dumped as binary.

IP30x V9 hf2 media problem: Echo canceller not reliable

Status Closed
Id 68901

Sometimes voice from ISDN/Analog to the IP is muted after some time.
Seems to be a problem in the latetest echocanceller.
Change to old DSP code until fixed DSP code is available.
Status: ip24.mak

Denial of Service filter in ethernet library did not work

Status Closed
Id 68907

this filter can be useful to prevent DOS attacks on non routing devices

SIP: Trap when handling SUBSCRIBE on federation interfaces

Status Closed
Id 68976

SIP client object may be deleted while DNS query is pending.
But only under critical timing conditions.

myPBX: Use display name for sending meeting urls and conference numbers

Status Closed
Id 68992

For normal chat messages the DN is used as the sender name. For sending the links for application sharing or audio conferences the CN was used.

Phone: Hide calling party on Pickup key fixed

Status Closed
Id 68995

Phone: Hide calling party on Pickup key fixed

SoftwarePhone: No CT setup with remote connected calls

Status Closed
Id 68996

The call transfer setup facility is removed in the call setup if the call is a by remote control connected call used in case of outgoing calls with myPBX. This fixes an empty diverting party number information element in the PBX. Now it can be used with a trunk PBX object with the enabled option 'Set Calling=Diverting No', otherwise the calling party number was removed within this object.

PBX: Conference trap

Status Closed
Id 68998

A zero pointer trap in the broadcast conference PBX object is fixed.

IP222/IP232: Hookswitch on some PCB not working

Status Closed
Id 69001

Reduce LED frequency from 12kHz to 6 kHz Status: orchid_lcd.cpp

phone_orchid - builtin speaker test did not work

Status Closed
Id 69026

works only over channel 0

myPBX: Login did not work with system names containing special characters

Status Closed
Id 69033

Missing URL encoding when sending the connect-request message.

PBX Mobility: Mobility object hanging if call establishment with SOAP/myPBX canceled

Status Closed
Id 69052

If a call for a mobile phone is initiated by SOAP or myPBX, a call is first sent to the mobile phone. If the mobile phone accepts the outgoing call to the destination is initiated. If the mobile phone did not accept the initial call, no other calls could be done from then on.

IP-DECT: Configuration of Media preferences did not work anymore

Status Closed
Id 69056

Media preference were ignored. This was a collateral damage introduced with fixes from the last hotfix

V9 Hotfix 4 (90600.04)

Changes included in Version 9 hotfix4 Definition

New Features

Improved protection against Denial of Service attacks

Status Closed
Id 69166

flooding a box with different kinds of packets may lead to out of memory conditions. The Denial of Service filter in the ethernet layer is activated where required. TCP listening sockets have a backlog limit now. The http service restricts the number of half-open sessions and limits the number of concurrent sessions according to the total memory available on a box.

ISDN interop issue with SecuGATE LI 30 from Sirrix

Status Closed
Id 69168

The SecuGATE LI30 is sending/receiving ISDN INFO messages in Call Proceeding State (State 3 and state 9), which was not supported

PBX Mobility: Support of transfer on mobile side, not using mobility

Status Closed
Id 69275

In case that the mobile phone transfers the call to another destination, this call must be removed from the mobility function, so that the mobility function is available for another call

USB Driver merge from v10 to v9

Status Closed
Id 69288

First step to support USB headsets. This is not functional yet.

SIP: New config file option /hold-notify-as-inactive

Status Closed
Id 69293

If set, holdNotific is interworked into "a=inactive".
If not, holdNotific is interworked into "a=sendonly".

SIP: New config file option /prefer-pai2

Status Closed
Id 69459

Interoperability with Telepo:
When receiving INVITE, get calling party id from second P-Asserted-Id header.

new: DHCP manufacturer specific option 'boot-cfg' provides an URL to read config from after any reset except 'creset'

Status Closed
Id 69472

This option permits to boot boxes with a fresh config provided via TFTP/HTTP without storing the config on the device.
It is intended to be used as follows:

1. the box is started with DHCP enabled (no initial configuration)
2. the box contacts the DHCP server and gets the ip-address and also the Vendor Specific Information in option 43.
Suboption 249 of the Vendor Specific Information specifies the URL of the boot config file.
3. the box polls the TFTP/HTTP server for the config file.
4. the box reads the config file and executes the commands provided in the file

The URL may contain the same meta-character strings an Update Server URL, for example #m (mac-address)
The length of the URL in the DCHCP suboption is restricted to 127 characters.

The URL is polled in 5 second intervals.
The config file is read and executed by the update process in the usual way.
A 'creset' commmand as last command of the file will restart the box with the new configuration without writing any 'config' command options to the flash.
After a restart by the 'creset' commmand the boot-cfg URL is ignored.
After a restart by any other of the 'reset' commands or by a power cycle the boot-cfg URL is processed again.

On an Innovaphone DHCP-Server configuration of a boot-cfg URL and providing it to clients via suboption 249 must be explicitely enabled by
config add DHCP0 /boot-cfg
config write
config activate
Once enabled the URL may be entered under "IP4/ETH0/DHCP-Server/Boot Config URL" and is provided to all clients then.

If an Innovaphone DHCP-Client receivess a boot-cfg URL it is displayed under "IP4/ETH0/DHCP/Boot Config URL".

auto complete dtmf feature codes with '#' after 2 seconds

Status Closed
Id 69561

Optional feature for phones, which are not able to send a '#', e.g. the iPhone.
They dial a feature like a cfu with a destination number and after two seconds, the feature code is automatically completed with a '#'.

PBX: HTTP request to initiate call for mobile phone

Status Closed
Id 69570

To improve GSM client functionality

PBX: WSDL for SOAP API readable from box

Status Closed
Id 69688

Simplifies Application development in some environments

SIP: Get display information from Call-Info header in register refresh response

Status Closed
Id 69758

Get display information from Call-Info header in 200/OK

IP-DECT: Display update

Status Closed
Id 69770

Now, the endpoint's display is updated if the registration forwards a display update.

Voicemail <pbx-query-obj> queries new properties

Status Closed
Id 69781

"user_type": returns "1"(endpoint) or "2"(other object like gateway)

"pseudo_type": returns types alike "vm", "bc_conf", "trunk"

"pseudo_text": returns friendly texts alike "Voicemail", "Bc Conference", "Trunk Line"

Bug Fixes

Kerberos: Protect against ping pong attacks

Status Closed
Id 68822

Do not answer with an error message to unexpected or malformed messages.

This protects against the "Kerberos Server Spoofed Packet Amplification DoS" attack. The attack causes two Kerberos servers to send each other error messages in a ping pong style.

RTP-DTMF: Start handling of RTP-DTMF on reception of END event

Status Closed
Id 68938

Workaround for Bug in MS Lync / Mediation server.
Mediation server changes destination port while sending RTP-DTMF redundancy retransmissions.

SIP: Presence interoperability with ESTOS UC server

Status Closed
Id 69050

'Do Not Disturb' signaling without presence/tuple/contact element.

SIP: Support for Mediation Server Cluster

Status Closed
Id 69051

Load balancing and fail-over acc. to spec.

Memory leak with each new telnet session

Status Closed
Id 69157

Memory leak with each new telnet session (last packet upon exit not cleaned up)

SIP: Media negotiation problem

Status Closed
Id 69159

On media-relay with exclusive codec we can answer an incoming SDP offer right away without passing to app.

PPP connection fails after LCP renegotiation with different authentication methods (PAP -> CHAP)

Status Closed
Id 69167

the active authentication entity was not stopped when another authentication was started and and signalled layer down after some timeout

PBX Twinning: Blind transfer to other phone in a twin phone configuration was not possible

Status Closed
Id 69170

Only normal call to other phone or consulation call was supported

Web-UI: Fixed layout of DynPBX configuration

Status Closed
Id 69197

Use min-width for fieldset.left

IPVA didn't run on vSphere5

Status Closed
Id 69232

IPVA didn't run on vSphere5

SIP: Interoperability of INFO(application/dtmf-relay) with Polycom

Status Closed
Id 69247

Fail to decode DTMF signal, since "application/dtmf-relay" body does not contain any CRLF.
While CRLF is required according to "SIP INFO Package for DTMF".

IP222/IP232: residual echo in handset mode

Status Closed
Id 69249

Change Codec config Status: ac_codec3.cpp

ip22/24/28/302/305: DSP affected by certain fax tones causing a trap

Status Closed
Id 69250

Assert removed for testing
Status: ac_dsp3.cpp

PBX: Phone config was not sent to phone, if phone was power cycled shorty after registration

Status Closed
Id 69280

The new registration after the power cycle was not detected as new registration but as re-transmission of the previous registration, so it was not reported to the PBX and no phone config was sent

CX0-Wave-Encoding sometimes produced white noise

Status Closed
Id 69286

CX0-Wave-Encoding sometimes produced white noise.
Also a memory leak was eliminated.

IP-DECT: Hanging call after call transfer

Status Closed
Id 69309

The hanging call in dectradio is fixed which occurred if a user does an unattended call transfer to an unassigned number and the transferred call is not disconnected.

SRTP: Bad key exchange on H.323/SIP interworking

Status Closed
Id 69353

MKI was added in SDP due to uninitialized data object.

Pass DNS-TTL to SIP

Status Closed
Id 69374

In order to do load-balancing SIP needs to ge hold on the TTL of DNS resource records.

SIP: Support of maddr parameter in redirect response for REGISTER

Status Closed
Id 69386

Set when redirecting REGISTER.
Read when processing redirect response.

G726 codec obsolete

Status Closed
Id 69388

The G726 codec was rarely used (if ever) in real life. In addition there are signaling problems specially with DECT peers when G726 is selected. Thus G726 is removed from the list of supported coders in all products.

IP222: Input ip addresses does not work

Status Closed
Id 69393

Entering an ip address on the phone's menu does not work.
All digits are doubled.

IP222: Call reroute did not work

Status Closed
Id 69455

Using redial key on in incoming not-connected call did not work.

PBX: Call Diversion/Forward to '-' did not always show expected result

Status Closed
Id 69457

A call diversion to a destination'-' can be used to explicitly no execute a diversion of this type. So if a user has an CFU to '-' and this diversion is valid for a given call (Filter, Boolean), the phone should ring.

In fact the call was rejected.

There was also a problem with CFB in case of "busy on ... calls"

IP-DECT: New radio BMC firmware PCS05Ak

Status Closed
Id 69468

The new radio BMC firmware PCS05Ak for the IP1200 fixes a trap by the DECT system if more than 255 DECT users without an endpoint subscription are sent to it.

PBX: No CLIR on internal calls did not work for SOAP

Status Closed
Id 69474

If the features "No CLIR on internal Calls" is activated on a PBX a CLI is sent to the called phone even if the call was sent with "CLI presentation restricted". The same should be case on SOAP/TAPI when monitoring this user.

Now when "No CLIR on internal Calls" is enabled all number information available is provided on SOAP.

PBX: Reject calls without media, if no known facility

Status Closed
Id 69477

Fixes compatibility issues between versions. For example presence subscription sessions from v8 phones being forwarded to voicemail

PBX Waiting: Not possible to send DTMF to Waiting Queue from myPBX Numeric Keyboard

Status Closed
Id 69482

The Waiting Queue prohibited the facilities used for this to be sent to calling phone

PBX: Filter for internal or external calls at CFs did not work CFB or CFNR if call already diverted

Status Closed
Id 69483

Problem:

User A has CFU to User B
User B has CFNR for ext. Calls only to User C

An internal call to A was diverted to B (ok) and after no response diverted to C (nok)

PBX Waiting: In case of "Announcemen w/o Connect" together with Alert-Timeout 0, DTMF dialing was not possible

Status Closed
Id 69496

In this case as only response to the incoming SETUP a PROGRESS was sent. This meant, that the caller was still in overlap dialing state, so a phone does not send DTMF, but translates input keys to INFO dialing messages.

A CALL-PROC is now sent before PROGRESS, which terminates the dialing.

SIP: Send BYE with Reason header with "Q.850 Recovery on timer expiry"

Status Closed
Id 69500

If session refresh is outstanding the call is released with BYE with "Reason: Q.850;cause=16".
Better send BYE with "Reason: Q.850;cause=102"

PBX Waiting: No ringback when doing two-stage dialing to a Gateway/Trunk object

Status Closed
Id 69531

A local ringback is now switched on, when receiving ALERT from called party

SIP: Bug when decoding SRTP keys from SDP

Status Closed
Id 69545

Bug in decode base64.

Web-UI: Input field to small for SIP-URI

Status Closed
Id 69578

Input control on "UC" tab of External UC properties dialog was too small.
Only part of SIP-URI was visible.

phone: assume an outbound call to be an external call if connected number info is missing in connect event

Status Closed
Id 69581

In certain ISDN configurations the PBX can not provide the connected number info in the connect event for an outbound call. In this case the the call was assumed to be an internal call and consequently was not recorded when transparent recording of external calls was configured.
Now an external call is assumed in this case.

IP-DECT: Reset link

Status Closed
Id 69584

The reset link of DECT System Config GUI page is fixed.

SIP: Do not check remote certificate name when calling PBX client with TLS

Status Closed
Id 69598

Check is reqired only when calling an unregistered client or when giving INVITE to registrar.

phone: VLAN signaling priority could not be configured via phone menu

Status Closed
Id 69633

Under "Menu/Administration/IP Settings/VLAN" there was only a "VLAN Priority" menu item. This menu item did override the 'Priority RTP Data' value but not the 'Priority Signaling' value as entered via WEB configuration.
Now the items "Prio. RTP Data" and "Prio. Signaling" replace the "VLAN Priority" item.

IPxx10-sata: trap after config /trace /track activation

Status Closed
Id 69642

Instruccion was accessing uninitialized pointer.

IP222: Alpha input using the num block

Status Closed
Id 69646

Input mode changes back to numeric in screen "Wahlvorbereitung".
Should stay in alpha mode.

IP200: Manufacturing test fails is hwbuild is reprogrammed

Status Closed
Id 69662

Old hwbuild was not cleared, because on INCA devices the hwbuild is located on a higher address was not cleared. Status: flash_firmware.cpp

H.323: Media Negotiation problem with Lync interop and SRTP

Status Closed
Id 69687

With a retrieve from the lync after hold (which is signaled as a reinvite with sendrevc) new media parameters were sent, containing new SRTP keys. These new media parameters cannot be used, on the PBX which is initiating new end to end media negotiation at the same time. These media parameter were not ignored properly.

phone_orchid: wrong volume setting when monitor mode is entered

Status Closed
Id 69734

when monitor mode ise entered by pressing the speaker key in a handset conversation the handsfree speaker is enabled in addition to the handset speaker.
the volume was reconfigured with the wrong value.

phone_orchid: Calls received with CLIR appear in call list with an empty entry

Status Closed
Id 69737

Calls received with CLIR or without a number/name appeared in call list with an empty entry; now either "anonymous" (CLIR) or "unknown" is displayed instead of a name

SIP: Take SDP id and version as 64bit integer

Status Closed
Id 69738

o-line of SDP offer/answer is defined as 64bit integer
Interoperability with MX-ONE

Relay: Forward facilities to local destinations

Status Closed
Id 69764

This fix is related to the previous fix #66629 for V9 hotfix2. Now, facilities are only forwarded, if the destination is a physical interface, not e.g. a SIP provider.

SIP: Trap when handling NOTIFY(application/qsig)

Status Closed
Id 69771

Traps if no progress indicator present in tunneled DISCONNECT message.

WEB-UI Config Upload screen blank after upload of a file with a lot of failing lines

Status Closed
Id 69812

When uploading a file with about 50 failing lines the screen was left blank without any info about the failing operation.

phone: "Keep Calling Party Info on Pickup Key" option from Phone->Preferences did not work with very long numbers/names

Status Closed
Id 69816

speciall when using this option in conjunction with the "Display Name on Pickup/Partner Key" option the calling party info was not correctly displayed

IP6010: SRTP using AES-192 and AES-256 did not work

Status Closed
Id 69828

Due to a bug in the encryption driver of the IP6010, only AES-128 worked on this platform.

IP-DECT: Location update (OEM)

Status Closed
Id 69863

Location update with message waiting information is fixed if the endpoint roams. This is only used by OEM devices.

Logging: Threshold for error event "SRTP authentication failed"

Status Closed
Id 69920

Error event was triggered at the very first decrypt failure.
Some decrypt failure are expected during media re-negotiation.
Trigger this error event after 10 decrypt failures in line.

PBX Calls Page/SOAP wrong number

Status Closed
Id 69921

In a configuration with escapes for calls from a slave and a node not the root node and the call forwarded to the master, because the number could not be resolved locally, wrong escapes were added to the called number

phone ip222: MWI LED not working

Status Closed
Id 69924

The MWI LED was not switched on for pending voice mails and/or pickups

Trap during fax transfer

Status Closed
Id 69950

Seen on Ip28, can happen on AC_DSP3 ( IP22/24/28/302/305/1060/301/6010 ) Status: ac_dsp3.h

phone_orchid: checkmark "Phone/User-x/General/Options/No DTMF Detection" has no effect

Status Closed
Id 70114

If this checkmark is set DTMF digits entered via keyboard in a connected call shall be sent in-band as voice data, not encoded in RTP-DTMF packets as usual.

V9 Hotfix 5 (90600.05)

Changes included in Version 9 hotfix5 Definition

New Features

IP-DECT: Default frame size to 30ms

Status Closed
Id 70140

Now the default frame size is 30ms.

SIP: Support for transparent message headers on transfer

Status Closed
Id 70163

Interoperability with Telepo.

added support for USB tracing with Wireshark

Status Closed
Id 70373

USB traces are now supported by the debug class and can be opened with Wireshark and a new innovaphone.dll.

SIP: Prefer P-Asserted-Identity with tel-URI

Status Closed
Id 70417

Prefer P-Asserted-Identity with tel-URI over that with sip-URI.
In case there are two P-Asserted-Identity headers in INVITE.

Gateway: Forward Display Info received from ISDN Setup to H.323

Status Closed
Id 70562

needed for compatibility with SecuGATE LI30

<pbx-getcallinfo> returning diversion reason

Status Closed
Id 70697

as
<pbx-getcallinfo out-leg2-reason="..."/>

myPBX: Interface for IM provider

Status Closed
Id 70867

Interface between the JavaScript client and the myPBX launcher. Status: IM to JS:
prepare_dial_name(value)
prepare_im(value)
ep_request(name,number)

JS to IM:
innovaphone_updateXml(xml)
innovaphone_sessionInfo(domain,name,number,cn,dn)

SIP: Diagnostics of transport error "SIP Overload"

Status Closed
Id 70883

Limitation of buffer allocation Status: siptrans.cpp/h
sipmsg.h

IP-DECT: License restriction for OEM device

Status Closed
Id 70887

A license restriction for an OEM device is added. It is not used in the IP1200.

SIP: Diagnostics of error "Registration expired"

Status Closed
Id 70938

Missing registration refresh Status: sip.cpp/h

Possibility to clear the call list in mypbx

Status Closed
Id 71202

A new button is now available in mypbx to clear the call list.

phone: LED mode of Join Group function key can be set both for idle and for active state

Status Closed
Id 71247

sometimes the "not in group" state must be signaled as the exception

myPBX: Show implicit visibility in groups

Status Closed
Id 71250

Active members of groups can see the presence and the calls of other group members. In order to make that clear to the user, now the visibility settings of myPBX show in what groups the user is visible.

IP-DECT: CSS changes for new OEM device

Status Closed
Id 71282

For a new OEM device with a changed style some CSS classes are added.

PBX: New User property 'Do not Disturb'

Status Closed
Id 71439

No calls are sent to the user if set.

Can be set by 'External UC' applications (e.g. Estos Procall)

PBX: Support up to six devices for a user

Status Closed
Id 71506

Sometimes 4 devices are to little.
A little bit of cleanup on the user interface was done as well with this change

phone: Mic Off/On controllable via Soap:UserRc(<call>,14/15)

Status Closed
Id 71721

To allow Soap app's control of the mute key

SoftwarePhone: New Tray Icon

Status Closed
Id 71969

The SoftwarePhone has got a new tray icon differs from the myPbx one, now.

Bug Fixes

SIP: Message decoding error

Status Closed
Id 70116

Only concerns Message headers whose value starts and with quotes, but are not quoted.
E.g.
Referred-By: "Huvudnummer"<sip:400@abcdef.ghi;fnrid=1759>;from-tag=5decdf1a;to-tag=2515833546;org-cid="6afa95ede909d311906f00013e11cdb3@192.168.2.115"

Linux: VLAN GUI removed

Status Closed
Id 70131

It is not necessary to configure a VLAN to Linux. Now the GUI page Linux VLAN is removed.

ip24/ ip6010/phone_orchid: in-band ring back tone generation into voice channel did not work

Status Closed
Id 70146

sometimes a ring back tone must be sent in-band to an ISDN channel. the tone was generated but passed to the wrong destination.

SIP: Problems with CLEARMODE

Status Closed
Id 70149

CLEARMODE was not offered as prefered codec outgoing INVITE.
Destination accepted G711a instead of CLEARMODE:

Offer:
v=0
o=- 18 1 IN IP4 172.16.66.77
s=-
c=IN IP4 172.16.66.77
t=0 0
m=audio 16422 RTP/AVP 4 18 8 0 96 97 101 13
a=rtpmap:96 G726-32/8000
a=rtpmap:97 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexa=yes
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20

Answer:
v=0
o=cp10 131833822144 131833822144 IN IP4 172.16.66.175
s=SIP Call
c=IN IP4 172.16.67.132
t=0 0
m=audio 33526 RTP/AVP 8 0 97
b=AS:75
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:97 CLEARMODE/8000/1
a=ptime:30

IP-DECT: Master radio list sync highlighting

Status Closed
Id 70150

In the IP-DECT Master radio list the active sync highlighting is fixed, now.

SIP: No update of name presentation after transfer on QSIG interworking

Status Closed
Id 70155

Fix for QSIG/SIP interworking.
Display name of callTransferComplete was not passed to SIP.

IP222/IP232: Handset volume with 90772 andHF4 too low

Status Closed
Id 70226

This is a side effect of the echo canceller setting, volume in IP222/232 increased. IP241 uses old settings.

SIP: Cannot use SIP phone with myPBX

Status Closed
Id 70252

Blind transfer (sending REFER) did not work since v9hotfix3.

IP241: Use menu-key to enter tftp mode

Status Closed
Id 70263

tftp was activated with alt-key, on on IP240 menu-key is used Status: platform_orchid.c
boot241.y

IP222/IP232: Sporadic Problem with the LCD Display after soft-reset

Status Closed
Id 70269

It seem to be problematic to reset all orchid modules, e.g. the DMA module during software reset.
Now only USB and ENET modules are reset, the display gets also reset.
The display reset is released in the firmware. Status: start_orchid.S
platform_orchid.c
phone_orchid.cpp
boot222.y
boot232.y
boot241.y

IP241: Missing fkey icons

Status Closed
Id 70353

Icons types list, face, mask-white and mask-black are available now.

PBX: Conference trap

Status Closed
Id 70362

Some reworks of the PBX conference object. Fixes traps with call transfers of conference calls and conference calls to other PBX objects or mobility. Object update is also possible without call and chat clearing, now. Set maximum call number takes effect for maximum incoming calls, now.

IP241: Status icon for conditional call diversion

Status Closed
Id 70363

Now having 2 different status icons for diversion.
One for unconditional and one for conditional forwarding.

IP241: Moving focus on screen "Call Diversion" activates diversion

Status Closed
Id 70377

Using UP/DOWN/LEFT/RIGHT keys on the "Call Diversion" screen makes ON/OFF changing it state.

Webmedia: Recording of G.722 did not work

Status Closed
Id 70382

Empty file was created.

myPBX: Closing a chat window sometimes causes a Java Script error

Status Closed
Id 70439

client.js, line 2033

The Problem is a collision of closing the window and incoming messages for that window.

SIP: DNS resolving _stun._udp.xten.com did not work

Status Closed
Id 70449

Because of unusual DNS response.
Target attribute of SRV answer records did not contain host domain name, but ip address.

<result val="0" title="SUCCESS">

<answer_rrs title="Answer Records">
<rr rr-type="33" rr-name="SRV" name="_stun._udp.xten.com" ttl="3407" priority="10" weight="0" port="3478" target="216.93.246.16"/>
<rr rr-type="33" rr-name="SRV" name="_stun._udp.xten.com" ttl="3407" priority="10" weight="0" port="3478" target="216.93.246.14"/>
</answer_rrs>

<ns_rrs title="Authoritative Nameserver Records">
</ns_rrs>

<ar_rrs title="Additional Records">
</ar_rrs>

</result>

H.323: Renegotiation to Fax did not work under some conditions

Status Closed
Id 70462

Problem happened
- If switch to fax was done right after connect. This is typically done by IP Fax Servers
- If multiple signaling hops (e.g. multiple PBXs) were used
- If connect to a tone interface happened during dialing

myPBX: Norwegian translation updated

Status Closed
Id 70477

Sent => Send
Logginn => Pålogging
Permanent logginn => Forbli pålogget

ethernet broadcasts bearing unicast IP packets with an arbitrary destination address were passed to the local IP-stack

Status Closed
Id 70503

some load-balancing implementations send unicast IP packets (specially TCP-SYN) as ethernet broadcast packets.
Such packets must be silently discarded if the IP destination address is not the address of one of the local interfaces.

IP222: Cannot leave menu screen with ESC when IP address ctrl was active

Status Closed
Id 70569

ESC was taken as BS (backspace).
ESC must be handled as ESC when BS is no longer possible (empty input).

PBX: Presence update for PBX objects

Status Closed
Id 70577

Now the presence is updated in myPBX for other PBX objects if the access is changed to allowed.

PBX: Conference presence info

Status Closed
Id 70581

Now the broadcast conference object updates the presence info and is shown as callable in myPBX.

myPBX launcher: Hotkey does not work with some applications

Status Closed
Id 70623

Try to get the selected phone number using WM_COPY before sending CTRL-C. This should help in some of the cases where the hotkey didn't work before.

SIP: Do not try to map local listen port with STUN when SIP/TCP is used

Status Closed
Id 70654

Mapping of local listen port is only required fur SIP/UDP

IP222: Use R-key as BACKSPACE on edit fields

Status Closed
Id 70664

Not ESC-key.
ESC-key is for leaving screen only.

Trap: When Dectmaster registers user at PBX using SIP protocol

Status Closed
Id 70675

After closing regstration Dectmaster starts another call.
Call is rejected, but signaling enity is deleted before call object.

IP-DECT: Hanging call after failed call transfer

Status Closed
Id 70756

The hanging call in dectradio is fixed which occurred if a user does an unattended call transfer and the call transfer fails.

SIP: Allow STUN to be used to map local media ports on SIP/TCP interfaces

Status Closed
Id 70809

STUN cannot map signaling TCP port, but UDP media ports. Status: medialib.h
media.cpp
h323ch.h/cpp
sip.h/cpp
siptrans.h/cpp

IPVA, Keyboard Console, Credentials With Special Chars Couldn't Be Entered

Status Closed
Id 70873

The Scancode table wasn't finished

SIP: Reduce memory footprint of SIP stack

Status Closed
Id 70886

Free INVITE request buffer when receiving ACK. No need to keep any longer.
Delete INVITE client transaction when cancelling.

SIP: No route processing if neither Record-Route header nor Contact header is present

Status Closed
Id 70971

Misleading trace message:
sip_call::process_routing(0xA8) Unsupported transport protocol: sip:user@domain.com;user=phone

IP6010: Wrong timer under high load

Status Closed
Id 71001

-Clear IRQ in handle-interrupt after os_interrupt is too late, since IRQ´s a enabled again and e.g. the timer irq is called again if a lower level IRQ like the enet occurs.
-The IRQ needs to be cleared in the serial-irq handler, in all case. After the serial-irq other interrupts are enabled.
Status: ip6010.cpp
ip6010.h

SIP: Media negotiation problem

Status Closed
Id 71009

Internal re-negotiation during early media on incoming SIP call.
Provide received offer to app again.

SIP: Offer CLEARMODE only if bearer capabilities are "Unrestricted Digital Information"

Status Closed
Id 71162

On "Unrestricted Digital Information" only CLEARMODE is offered (no audio codecs).
On other bearer capabilities no CLEARMOE is offered (only audio codecs).

phone: dialog and presence subscriptions sometimes got lost after PBX restart when phone config was stored on PBX

Status Closed
Id 71198

This happened specially when both "Store Phone Config" and "Discard Config on Phone" was checked in the user object because of a unsubscribe/subscribe race condition.\t

when editing a phone config template the dialing location inherited from a predecessor template was stored in the edited templat

Status Closed
Id 71246

after a template has been edited unchanged information units inherited from predecessor templates must be removed from the edited template. this did not work for the dialing location and thus a later change in a predecessor template had no effect.

SIP: Endpoints behind NAT could not register at public PBX

Status Closed
Id 71266

PBX must send response to the (public) IP address where request was received from.
Not to the (private) IP address in Via header or Contact-URI.

ipva, update to _platform_tracing.xsl

Status Closed
Id 71274

ip6 missing

SIP: No media after accepting a waiting call

Status Closed
Id 71288

Call waiting on a phone.
Going onhock while another call is waiting starts ringer.
After going offhook again the waiting call is accepted, but no media in both directions.

CX0 Wave-Encoding Not Working If Fact-Chunk Present In Header

Status Closed
Id 71290

CX0 Wave-Encoding Not Working If Fact-Chunk Present In Header

DHCP Server Identifier could not be cleared via WEB interface

Status Closed
Id 71305

When the field 'Server Identifier' was cleared and OK was pressed the just cleared value reappeared but after a reset 'Server Identifier' was clear.

IP241: Show ISDN display information in full length

Status Closed
Id 71330

Use scrolling if required

NAT: Mapping to different internal UDP port did not work

Status Closed
Id 71339

Configured destination port got lost after configuration

phone: send config to PBX only when the config was edited on phone

Status Closed
Id 71387

A config from an older PBX may contain duplicate elements which are stripped by the phone. I such a stripped config is sent back to the PBX the PBX will return the old config again.

SIP: Send first NOTIFY(dialog) after sending 200/OK for SUBSCRIBE(dialog)

Status Closed
Id 71413

NOTIFY(dialog) was sent before 200/OK for SUBSCRIBE(dialog)

\tRequest: SUBSCRIBE sip:150@192.168.200.14
\tStatus: 401 Unauthorized
\tRequest: SUBSCRIBE sip:150@192.168.200.14
\tStatus: 100 Trying
\tRequest: NOTIFY sip:158@192.168.200.214:5060
\tStatus: 200 OK
\tStatus: 481 Subscription Does Not Exist

SIP: Interop with Nortel CS1000 SIPLine GW

Status Closed
Id 71426

Nortel sends 183/Progress with 'sendrecv' answer
followed by UPDATE with 'inactive' offer
followed by UPDATE with 'sendrecv' offer.

Innovaphone SIP stack remains in 'inactive' state.

H.323 Signaling Timeout event was generated for 'normal' RAS re-transmissions

Status Closed
Id 71434

A single retransmission is normal under heavy load, so this is no reason for an event. Signaling Timeout events are now generated only if they cause a state change.

H.323 re-negotiation: Don't reuse media proposals if a select was already sent

Status Closed
Id 71435

This fixes compatibility issues with SIP, especially when SRTP is used.

IP241: Show both parties of each call on screen "Pickup List"

Status Closed
Id 71455

Currently only calling party infomation is displayed.

phone: when a menu item was selected by the adjacent key sometimes the function assigned to this key was also executed

Status Closed
Id 71469

The function assigned to the menu item (for example pickup) was executed first and then the key-function (for example dial).

PBX: Wrong web page when submitting an object and an error happens

Status Closed
Id 71470

If for example a dupicate number is detected, the same web page should be displayed including the error message for the duplicate number. But not the same page was displayed but a page which could contain information not related to the object.

v8 to v9 upgrade problem with gateway registration names containing non-ASCII characters

Status Closed
Id 71474

In general this was a problem with config line arguments seperated by ':'. This happened with the <number>:<name> argument within gateway definitions. The ':' was url-encoded and <name> interpreted as <number>

PBX: CFU was executed on PRESENCE_PUBLISH/SUBSCRIBE calls

Status Closed
Id 71479

This was unexpected behaviour. You want to see the presence status of the configured user and not the presence status of the destination to which this user has configured a call forwarding

SIP: Interoperability with MX-ONE

Status Closed
Id 71480

A semi-attended transfer fails if MX-ONE sends INVITE(Replaces)
instead of 200/OK when connecting a call.

Phone: Presence-Fkey did not always show presence set by myPBX

Status Closed
Id 71487

Problem:
1) Set presence A with IP phone (fkey shows A)
2) Set presence B with myPBX (fkey shows B)
3) Delete presence with IP phone (fkey shows no presence)

Now Fkey shows presence B.

PBX: Tooltip on "PBX/Config/Log Calls" checkmark wrong

Status Closed
Id 71507

It read "If not checked PBX calls are logged", should be "If checked PBX calls are logged"

IP241: Added flag to hide background-image configuration from phone menu

Status Closed
Id 71513

See http://wiki.innovaphone.com/index.php?title=Reference9:Concept_Fine_grained_function_hiding#More_Information
for more information

Media: Discard RTP packets from wrong source

Status Closed
Id 71515

Packets arriving at RTP port must be discarded if the source if not the expected one.
To be save against DOS attack and for interop with Lync.
In some scenarios Lync starts sending RTP packets while having the call set to 'inactive'.

PBX: Blind transfer with consultation to BC-Conference failed

Status Closed
Id 71540

The call was disconnected

H.323: No Alarm/Event should be generated by shutting down registration due to reset

Status Closed
Id 71545

no event is sent to the application about this kind of unregistration

phone-orchid: micro-speaker loop of embedded test did not work

Status Closed
Id 71566

Der Test war im Treiber nicht freigeschaltet

myPBX: Send existing invitation links to users that join the chat session later

Status Closed
Id 71608

Users A and B have a chat session.

1. A starts application sharing or audio conference
-> A and B receive an invitation link

2. A adds a third user C to the chat and C accepts
-> C should also receive the invitation link

This worked only for audio conference but not for application sharing.

IP-DECT: Ring back tone after transfer

Status Closed
Id 71610

If in-band ring back tone is sent and the call is transferred to a new destination with no in-band ring back tone, a local ring back tone must be played to the DECT handset. This is fixed now.

Ring Back tone missing after transfer when in-band tone was provided before but not after transfer

Status Closed
Id 71612

The initial local ring back tone must be restarted when no data is received after transfer.

PBX: Send Name Identification with CLIR calls if "No CLIR on internal Calls"

Status Closed
Id 71643

The feature "No CLIR on internal Calls" did not work completetly. The number was sent, but the Name Id was still suppressed

phone_orchid: Ring Back tone missing when Silence Compression is enabled

Status Closed
Id 71670

When Silence Compression was enabled the tone generation was not triggered

IP232: Fix for touch handling

Status Closed
Id 71674

Do not open touch keyboard on controls with CTRL_READONLY.
Do not open touch keyboard on controls without CTRL_ACTIVATE.
Using KEY_SHIFT has modified key to uppercase permanently.
Cursor positioning on text controls did not work.
Multi-line editor control was not displayed after hiding touch keyboard.
Hide overlay keyboard after next touched key.
Move and resize editor control when activating touch keyboard.

SIP: Trap on timer expiration during call release

Status Closed
Id 71699

Media negotiation watchdog timer expired after final SIG_REL went to app.
But before app deleted the call object.

phone: display info provided by SETUP or CONNECT was ignored

Status Closed
Id 71727

only the display info provided by an INFO event was handled

phone_orchid: continuous dialing tone stops after 10 seconds and is not restarted anymore

Status Closed
Id 71753

the continuous dialing tone as used in most countries must be played as long as no digit is entered.

IP230/IP240: Sporadic traps during manufacturing programming

Status Closed
Id 71778

Inefficient loop during mac address programming Status: flash_firmware.cpp

IPVA, Reset didn't work on VMware Player 4.0

Status Closed
Id 71818

IPVA, Reset didn't work on VMware Player 4.0

IP241: Slow screen update when changing fkey type on fkey configuration screen

Status Closed
Id 71831

Removing and adding config controls is very slow.

ip6010 - calling &amp; dtmf tone timing did not work

Status Closed
Id 71966

the tone time was calculated much too short

IP232: Redesign of touch keyboard

Status Closed
Id 71968

Redesign of touch keyboard

IPVA, Trap After Failed FW Upload

Status Closed
Id 71978

If a firmware upload failed with "wrong checksum" the box trapped occasionally, leaving the boot disk in an inconsistent state.

V9 Hotfix 6 (90600.06)

Changes included in Version 9 hotfix6 Definition

New Features

IP232: Presence fkey shows current activity, but not current note

Status Closed
Id 71611

Show current note in second line of fkey label.

myPBX: Cause codes for instant messaging

Status Closed
Id 71878

Display causes why a chat session has been terminated.

For example: Unassigned number, rejected, no response...

myPBX: Make call state icons clickable

Status Closed
Id 71886

Clicking a call state icon should have the same effect as clicking the text.

myPBX: Notification on disconnected chat sessions

Status Closed
Id 71918

Display a text that the chat has been terminated.

Debug information on assertion

Status Closed
Id 71961

More debug information on default event handler.

myPBX: Display reason why a browser is not supported

Status Closed
Id 71972

Currently it only says "Browser version not supported".

It should also tell what feature isn't supported:
- XmlHttpRequest
- WebStorage
- PostMessage

myPBX: Forward invitation links to incoming chat calls that are added to a chat conference

Status Closed
Id 72280

The same way like the invitation links (for application sharing and audio conferences) are forwarded to additional outgoing chat calls.

myPBX: Display invitation links in the chat window of the organizer

Status Closed
Id 72282

The organizer of an application sharing session or a audio conference wants to see all corresponding links in the chat window.

SoftwarePhone: Password encryption tool

Status Closed
Id 72431

A password encryption tool was added.

SoftwarePhone: Crash dump

Status Closed
Id 72433

Now, the SoftwarePhone writes a crash dump file.

myPBX: Display shortened links in chat messages

Status Closed
Id 72445

Shorten the displayed part of the URL in order to avoid text overflow.
Cut overflowing texts at the right boundary of the chat box.

PBX: New presence access flag 'online'

Status Closed
Id 72538

With this flag a user allows access to his online status

myPBX: Configure visibility of online state and presence separately

Status Closed
Id 72632

Phone status and IM status are now called online status.
Activity and note are now called presence.

The visibility of both is now configured separately.

PBX Mobility: Better support for Opticaller

Status Closed
Id 72701

HTTP request to initiate call for mobile phone

myPBX: New translations

Status Closed
Id 72768

For the other v9hotfix6 items regarding myPBX.

Support for more automated Fax Tests

Status Closed
Id 72826

Switching from Audio to Fax to Audio to Fax. Special case which could happen if switch to Fax happened before connect.

myPBX: Highlight incoming alerting calls

Status Closed
Id 73093

Highlight the alerting icon using an orange border.

SIP: Interworking with KIRK Wireless Server 300

Status Closed
Id 73110

If "KIRK Wireless Server 300 PCS10__ r3327"
calls into PBX and is connected with Voicemail,
Voicemail may send re-INVITE with SRTP key.
Instead of accepting or ignoring the SRTP option,
KIRK Wireless Server 300 rejects the whole SDP offer.

Now we retry the re-INVITE w/o offering SRTP key.

PBX: Forward original received ISDN display element to picking up or forwarded call

Status Closed
Id 73278

In the display element from ISDN there could be vital information from equipment like crypto gateways. This should be available also if the call was picked or forwarded.

SoftwarePhone: Support for Jabra SPEAK 410 USB with product id 0x0410

Status Closed
Id 73424

Now, the SoftwarePhone supports Jabra SPEAK 410 USB with product id 0x0410.

SIP: Overriding cause code mappings did not work

Status Closed
Id 73477

Mapping of SIP response codes into Q.931 cause codes could not be overwritten.

Linux: Empty file check

Status Closed
Id 73554

Now, an empty file is not accepted as a successful file upload.

Bug Fixes

myPBX: Make more obvious that closing the chat window terminates the chat

Status Closed
Id 71893

Display warning message when the user tries to close the chat window.
Show disconnect icon instead of close icon in the window.

IPVA, Optimize Flash-Related I/O Operations

Status Closed
Id 71954

Specifically to reduce boot-time. Utilizing 64KB I/O accesses where possible.

H.323 RAS: Registration with authentication to account without authentication failed in a strange way

Status Closed
Id 71958

Information was missing that no authentication was supported, so it was continued to retry instead of giving up right away

TCP: Roundtrip measurement wrong in case of packet loss

Status Closed
Id 71985

In case of packet loss, way to high round trip values were measured. If the packet-loss was to high, this could result in a constantly increasing re-transmission timeout value.

SIP: Workaround for delays in media negotiation

Status Closed
Id 72189

Workaround for delays in media negotiation caused by delayed Admission on H.323 calls.

SIP: Trap on IP-DECT when re-configuring PBX link

Status Closed
Id 72190

85:2195:425:7 - REG_PRI.4 default(8102be48): serial_timeout
85:2195:425:7 - Assertion failed line 748 in common/os/os.cpp, object deleted
Status: Merged to 09-80500

IP232: Input of capital Latin Extended-A

Status Closed
Id 72226

Input of capital Latin Extended-A

myPBX: Queue chat messages for im calls that are not yet alerting

Status Closed
Id 72237

The following fix did not work across PBXes:
#71608: myPBX: Send existing invitation links to users that join the chat session later

The problem was that messages were only queued in alerting state. When the call is made across PBXes this state isn't reached, yet.

Scheduling improved to avoid processes not being scheduled during long flashman operations

Status Closed
Id 72243

In version 7 it could happen, that IP and other processes were not scheduled any more during periods of long flashman operations (e.g. bootcode update or reorganizing flash).

In version 8 and higher there was already a fix for this problem, but this included special handling of the flashman priority level, which was not a good solution even if it worked.

IP232: Touch keyboard key highlighting

Status Closed
Id 72268

Added highlighting for key like SHIFT, BACKSPACE, etc.

PBX: H.323 Names in some places (e.g. Waiting Queue Maps) containing non-ASCII charcters could break the user interface

Status Closed
Id 72277

This happened esspecially for Names as destinations for Waiting Queue maps, but it could happen at other places as well.

SIP: Cleanup failed (resources leaking)

Status Closed
Id 72284

Call and channel objects were not freed sometimes
when INVITE was followed by CANCEL very fast.

IP241: Disable Link LED not working

Status Closed
Id 72338

Timing problem with access to the paged MDIO registers Status: orchid_drv.cpp

Trap: When accessing web interface

Status Closed
Id 72362

Seen once.
Not reproducable yet.

Ip6010 DSP Disconnect timeout after fax session

Status Closed
Id 72403

Debugs added
* enable with http://addr/debug.xml DSP trace and DSP control message trace to printout all packets to the DSP with a descriptive string. That allows to analyse the message flow to the DSP after a trap.
* for further testing old fax disconnect procedure can be enabled with http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl with "t38 skip fax close". Status: ac_dsp3.cpp
ac_dsp3.h
ac_491.h
dsp.xsl

IP2x2: Fixed fkey handling of builtin test

Status Closed
Id 72442

Fkey test did not work for touchkeys, key symbols were misplaced on screen

myPBX: Improve presentation of chat conferences

Status Closed
Id 72444

Chat conferences should not be displayed as multiple calls but as a single item in the main window.

IP232: Parameters for touch sensor fixed for latest sensor

Status Closed
Id 72455

latest touch was treated like the very old version that needed other parameters.
Old touch is not supported any more ( only 3 were build) Status: edt_touch.cpp

IP241,IP222,IP232: Make password configuration more convenient

Status Closed
Id 72456

When entering a passwort on the phone's ui content is displayed as "****".
Show last enered character in plain text.

Ip6010 DSP Allow coder change from T38 to voice and back to T38

Status Closed
Id 72457

.. Status: ac_dsp3.cpp

IP222: Take center key of 4-way-navigation control an OK key

Status Closed
Id 72477

Center key should enter selected menu item.
Not leaving the current menu screen.
ESC key can be used to leave current menu screen.
Phone app will ask whether to save changes.

myPBX: Name and Number Display not correct on IM sessions across PBXs

Status Closed
Id 72491

for local sessions (same PBX) the number and the Long Name was displayed for the remote party and this should be the same for remote session.

Sometimes the number was missing, sometimes the Name was displayed instead of Long Name.

The number was not adjusted correctly if different Nodes were involved

SoftwarePhone: Trap and media channel after conference

Status Closed
Id 72511

If a call is disconnected during conference mode, the SoftwarePhone traps or there is no voice channel to the remaining party. This is fixed now.

HTTP: Accept authentication with both Latin1 and UTF-8 coding

Status Closed
Id 72512

The server does not know what encoding is used in Basic- or Digest-Authentication. Therefore both encodings shall be tried.

ip28 codec hang when TEL port configured to '600Ohms'

Status Closed
Id 72521

this option experienced problems in the past and should therefore not be selected. If selected the firmware will switch to default CTR21.

PBX: Cause for release/reject of a IM session not signaled

Status Closed
Id 72533

Better feedback to the User

myPBX: New icons for unknown phone and im status

Status Closed
Id 72536

The difference between open, closed and unknown was not understood well.

SoftwarePhone: Dialing outbound calls temporarily failed

Status Closed
Id 72628

When there was a held call and a consultation call and the consultation call was released by the remote peer the SoftwarePhone did not accept further outbound calls until the held call had been released. This is fixed now.

IP232: Fix for touch keyboard handling

Status Closed
Id 72651

When editing call diversions, one of the on/off controls may render across touch keyboard.
Content of multi-line-edit-control was mis-placed when touch-keyboard was activated.
Display of first matching directory entriy on indirect dialing screen.

ip241 - monitor mode (handset + speaker) did not work in V9hotfix5

Status Closed
Id 72702

monitor mode (handset + speaker) was displayed but the speaker was mute

update - scfg command could hang when the HTTP session was broken or prematurely closed by the server

Status Closed
Id 72708

in consequence update script processing was stopped until reboot

Web-UI: Wider input fields for domain names

Status Closed
Id 72723

When configuring
- Remote Domain
- Local Domain
- Proxy
- STUN Server
on a Gateway interface there's need for more space.

Trap: When Dectmaster registers user at PBX using SIP protocol

Status Closed
Id 72729

When Dectmaster registers user at PBX using SIP protocol

SIP: Add payload type for RTP-DTMF in case of media-relay

Status Closed
Id 72732

Add payload type for RTP-DTMF (telephone-event) to SDP offer in case of media-relay.
Should help on Mobility scenarios.

PBX: Called Name displayed when calling an object with forking was wrong

Status Closed
Id 72735

The name of the forking destination was displayed instead of the name of the called object

IP4001/IP6000 -UART driver did not work

Status Closed
Id 72743

the hardware requires the registers to be written 32-bit wise which was not respected by a fix one year before.

PBX: No Audio if call thru Waiting Queue DTMF destination, was transfered to BC-Conf

Status Closed
Id 72746

Problem caused by call state management error in PBX for calls connected without alert if alert was received later

IP241,IP222,IP232: Wrong call state displayed

Status Closed
Id 72748

"Destination reached" displayed instead of "subscriber busy" if consultation call was rejected.

PBX Waiting: Ringback missing when using DTMF to dial from one WQ to another which is alerting

Status Closed
Id 72766

This was a collateral damage for a fix for Waiting Queue announcements from a Boolean Object

IP222,IP232: Long key press on numeric block while being offhook does not open directory search

Status Closed
Id 72831

Long key press during offhook is expected to open directory search.

IP232: New touch control in menu header for screen exit

Status Closed
Id 72833

Re-design

IP241,IP222,IP232: Support for unicode 0x308 (diaeresis/umlaut)

Status Closed
Id 72859

When ä,ö,ü are encoded as a,o,u followed by diaeresis from unicode block 'Combining Diacritical Marks' only a,o,u where displayed.
Now ä,ö,ü are displayed.

IP241,IP222,IP232: Support for hebrew and arabic presence notes

Status Closed
Id 72905

Hebrew and arabic text passages are rendered frmo right to left.

PBX: Conference no media

Status Closed
Id 72915

There is sometimes no media for a conference member. This is fixed, now.

SIP: Memory leak during transfer

Status Closed
Id 73003

Occured on internal testing only (002-conf-with-bcast.xml)

IP241,IP222,IP232: Update for CFU indication on idle screen

Status Closed
Id 73013

Not displaying "cfu:241" on header of idle screen
but "Diverted to 241"
or "Umgeleitet zu 241"
or "Renvoyé sur 241"
etc.

myPBX: Simplify adding multiple users to a chat session

Status Closed
Id 73016

Do not unselect the chat session, after a person has been added. Replace "start chat" buttons by "add to chat" buttons when a chat session is selected.

RTP-DTMF: Digit may get lost during media re-negotiation

Status Closed
Id 73037

Receiver starts handling on END event.
Sender may stop sending before END event was sent.

myPBX: Do not show bubble for outgoing chat calls and messages

Status Closed
Id 73039

Notifications are only needed for incoming calls and messages.

myPBX: Skript errors in chat window

Status Closed
Id 73040

Closing the chat window caused different script errors in Internet Explorers and Firefox.

LDAP replication from v7 could stop

Status Closed
Id 73052

LDAP replication from v7 could stop

SIP: Don't write SRTP key into T.38 part of media description

Status Closed
Id 73112

v=0
o=- 1295 1 IN IP4 10.120.55.3
s=-
c=IN IP4 10.120.55.6
t=0 0
m=audio 20026 RTP/SAVP 8 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -
a=inactive
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9xHxSg836505XOXwdIHfQ8Cm2ZYezNPpjvHNPCvb
m=image 0 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9xHxSg836505XOXwdIHfQ8Cm2ZYezNPpjvHNPCvb

PBX External UC: Support of multisite setup

Status Closed
Id 73115

This requires all calls to the External UC to be routed to one interface. Before this fix the calls were routed within the local PBX only.

Now a Name (not Long Name as before) has to be configured for the External UC interface.

IP241,IP222,IP232: Show number of missed calls in status bar

Status Closed
Id 73140

New extended symbol is more eye-catching.

PBX: Called party number wrong at original called phone with forking

Status Closed
Id 73145

If the forking destination and the original called phone are assigned to different nodes

Media Negotiation: v9 XPARENT not compatible to v8 or earlier XPARENT

Status Closed
Id 73153

In v9 a dynamic payload type is used for XPARENT to be compatible to SIP, whereas in v8 an earlier payload type 0 was used. Within the media negotiation this should be detected and switched back to payload type 0.

Phones: Update of phone text data

Status Closed
Id 73211

Some translations missing (Polski,Eesti).
Support for language typical letters (Polski).
Some abbreviation removed (for color phones).

PBX: Conference GUI update with IE

Status Closed
Id 73256

The PBX broadcast conference object window is not closed if IE is used. This fixed now.

SIP: support for "Content-Type: multipart/alternative"

Status Closed
Id 73260

Lync interoperability

IP2x2: Keyboard handling on call reroute

Status Closed
Id 73268

Redial key launches "Transfer" screen.
After entering the destination number, Redial key should cause execution of call reroute.

PBX Waiting: When using a waiting queue for outgoing dialing, the announcement was cut off at the beginning

Status Closed
Id 73274

This was because the announcement was started already when the call was sent out and not when the call was connected

myPBX: Allow spaces in last LDAP attribute

Status Closed
Id 73276

Currently "j doe innovaphone" finds all internal users starting with j. It should only find internal users that start with "j doe innovaphone".

IP232: Cursor positioning when touching ip address control

Status Closed
Id 73407

Make sure input cursor is at right side of touched octet.

SIP: Configuration of SIP response code mappings did not work on IPxx10 and IPVA

Status Closed
Id 73453

http://wiki.innovaphone.com/index.php?title=Howto:Tweak_SIP_Response_Code_to_Q.931_Cause_Code_Mapping

PBX: Reporting licenses counting could be wrong if config templates were used

Status Closed
Id 73466

It could happen that a reporting license was used up by a config template object, if reporting was checked and other config templates where referenced

myPBX: Allow overriding implicit allows

Status Closed
Id 73471

It was not possible to define an allow with the same name. This is needed to override allows that are inherited from templates or group memberships.

IP241,IP222,IP232: Trap when scrolling down long call list

Status Closed
Id 73494

Out of memory trap when scrolling down long call list.

Trap: When using Diagnostics/Ping

Status Closed
Id 73536

When leaving Diagnostics/Ping or Diagnostics/Traceroute web screen a trap may occur.

PBX: Navigating in locations and groups if non-ascii charaters were used, failed with internet explorer

Status Closed
Id 73567

Internet explorer buggy concerning url encoding of href attributes

IPVA, crash-dump without backtrace, if tracing was switched off internally

Status Closed
Id 73574

IPVA, crash-dump without backtrace, if tracing was switched off internally

ISDN: Do not provide tones and no signaling of inband info for unrestricted digital information calls

Status Closed
Id 73603

There is equipment, which is doing unrestricted digital information ISDN calls, which gets confused if there is an ALERT message indicating inband tones (ringback).

IP241,IP222,IP232: Symbol "new messages" and symbol "headset" do overlap in status bar

Status Closed
Id 73608

Moved "headset" symbol to right side of status bar.

IP-DECT: DECT endpoint update for multicast identities

Status Closed
Id 73613

The update information to DECT is changed if the user logs out a handset. This is used to update the multicast identities in some OEM handsets.

Ip6010 DSP Allow calls to start with T.38

Status Closed
Id 73626

Needed for some T38 fax applications Status: ac_dsp3.cpp

IP241,IP222,IP232: Missing "Recall possible" display text on call completion

Status Closed
Id 73628

Missing display text on 'recall possible' notification.

SIP: Media negotiation problem on outbound call from BC Conference object

Status Closed
Id 73652

INVITE was sent without SDP offer.
test/9.00/pbx/conf/SIP/002-conf-with-bcast.xml

IP241,IP222,IP232: Extra digits dialed when using RIGHT key on screnn 'indrect dialling'

Status Closed
Id 73694

'123123' instead of '123' was dialed.

ISDN Trunk: Transfer to ISDN Trunk with TONE interface failed

Status Closed
Id 73695

There was not media after the transfer

H.323: Renegotiation to SRTP after dialtone failed

Status Closed
Id 73698

Happens with Escape Dialtones configured in PBX and calls with SRTP enabled

Option added to wait for the DSP to disconnect fax, needed only for some fax devices.

Status Closed
Id 73748

Use the prodedure of
Mantis 67821: ip22/24/28/302/305: DSP affected by certain fax tones
only if the option /t38-wait-fax-close set.

This avoids traps caused by disconnect timeouts if the DSP response is not received in time.
Status: ac_dsp3.cpp
ac_dsp3.h
dsp.xsl

SIP: Using wrong remote port when registering

Status Closed
Id 73784

Only affects IP-DECT when handset is switched OFF and ON and if the SIP runs on non-standard port.

IP232: Memory leak in display rendering

Status Closed
Id 73864

Memory leak in display rendering

V9 Hotfix 7 (90600.08)

Changes included in Version 9 hotfix7 Definition

New Features

IP241,IP222,IP232: Some diagnostics to check phone's ui memory consumption

Status Closed
Id 74049

Where have all the memory gone.

SIP: Support for "Content-ID" in "multipart/alternative" bodies

Status Closed
Id 74161

Support for "Content-ID" in "multipart/alternative" bodies

IP241,IP222,IP232: Added drop shadow to lower end of menu screens

Status Closed
Id 74278

To make end of list of menu entries more discernible.

myPBX: Support static URLs for application sharing

Status Closed
Id 74389

In some application sharing solutions a fixed link can be used to create and join meetings (GoMeetNow, BeamYourScreen).

There are two URLs configured, one for the presenter and one for attendees. When the user clicks the aplication sharing button the links are sent using chat messages.

PBX: Tracing flag turns on tracing in all dyn PBX's as well

Status Closed
Id 74390

Helpfull to debug dyn PBX setups

debug method raw_ethernet

Status Closed
Id 74897

A new debug method to trace raw ethernet data.

Support for new hardware

Status Closed
Id 74990

.

possible use of push-pull drivers for new hardware

Status Closed
Id 75051

not visible to customers

PBX/IP-DECT: User password length

Status Closed
Id 75453

The maximum user password length in the PBX and IP-DECT is increased to 23.

H.323: Support for switch back to local media on endpoint

Status Closed
Id 75522

needed to resume normal call after a conference using external conference unit

IP232: Touch gesture to open directory-search/indirect-dialing screen

Status Closed
Id 75549

Vertical upwards on idle screen opens directory-search/indirect-dialing screen.
Vertical downwards on idle screen opens main-menu screen.

PBX-SOAP: Wsdl Versioning mechanism fixed

Status Closed
Id 75552

Better mechanism, to avoid constantly adding arguments to the Version function with new Versions.

different port naming convention for new hardware

Status Closed
Id 75575

BRI1..5 instead of TEL1..4+PPP

PBX-SOAP: UserHold without MOH to local User

Status Closed
Id 75577

UserHold was sending MOH to the local and the remote User. With the argument remote=true, the MOH is sent to the remote user only

myPBX launcher: Use default configuration from local machine registry hive

Status Closed
Id 75634

If no user configuration is given, the following values are copied from the local machine hive.

"Software\\innovaphone\\myPBX\\URL"
"Software\\innovaphone\\myPBX\\secondaryURL"

Also the autostart checkmark is disabled, if autostart is activated, globally. Autostart is controlled with the following registry key.

"Software\\Microsoft\\Windows\\CurrentVersion\\Run\\innovaphone myPBX"

SIP: support for proprietary message header

Status Closed
Id 75864

Support for proprietary message header (CAL)

H.323: Call to SIP Trunk with MediaRelay and exclusive coder renegotiation from TONE to early media did not work

Status Closed
Id 75959

No ringback and no audio after connect

SIP: Added Call-Info header to re-INVITE for hold/retrieve

Status Closed
Id 76011

Call-Info: <urn:X-cisco-remotecc:hold>
Call-Info: <urn:X-cisco-remotecc:resume>

SIP: Support for active call pickup acc. to RFC-3891

Status Closed
Id 76035

Sending INVITE with Replaces header to do pickup at SIP PBX's

phone ip222, ip232 : USB headset support (beta)

Status Closed
Id 76463

general support for USB headsets added, some headsets are tested and working, others will follow soon

phone - Headset Function Key can be configured now as enable/disable or as call control key

Status Closed
Id 76468

a headset key mode can be configured to use the key either to enable/disable the headset (Mode: Enable) or to start/accept/clear calls via headset (Mode: Control)

Bug Fixes

Ip6010 DSP Allow coder change from T38 to voice and back to T38 with local DSP

Status Closed
Id 73193

.. Status: ac_dsp3.cpp

ISDN: Enable fax detection only after connect

Status Closed
Id 73213

On ISDN networks it can happen that the Connect message is delayed. This way fax tones are forwarded to the caller before the caller has received this Connect.

This way a renegotiation on voip to fax could happen before the connect, which is not supported by sip.

Phones: Update of phone text data

Status Closed
Id 73775

External input

SIP: Handling of collision of transfer and release

Status Closed
Id 73936

If one end releases a call while the other initiates an attended transfer, a "ghost call" may remain.
Resource leak.

IP241,IP222,IP232: Limitation of background image size

Status Closed
Id 73957

Limit is 512 kBytes now.
To keep phone from out of memory trap.

H.323: Renegotiation after PBX Node dialtone failed sometimes

Status Closed
Id 74020

This depended on timing. This happened if the call was sent after the dialtone to a master over a slow link.

IP232: Open phone menu by touch gesture

Status Closed
Id 74046

Touching header bar and moving vertically down opens the phone menu.
No need to use center key of 4-way control.

Dect User Gui: AD-replicated objects weren't listed under certain conditions

Status Closed
Id 74051

Deleted object thate were re-replicated werent't visible in the Dect user GUI. A mandatory Ldap attribute was missing under such circumstances.

IP222: Handset Microphone level too low

Status Closed
Id 74073

Handset micro increased by 6db
Handsfree micro unchanged
Status: ac_codec3.cpp

IP222,IP232: Language configuration not saved

Status Closed
Id 74087

Language can be selected on phone menu, but change is neither saved nor applied.

SIP: Don't take NOTIFY(application/simple-message-summary) with "Do-Not-Disturb: no" as FTY_MWI_DEACTIVATE

Status Closed
Id 74162

Interop with Nortel CS1k PBX

Phones: Fine grained function hiding did not disable Fkey configuration

Status Closed
Id 74174

Value 0x00000800 (PHONE_HIDE_US_FUNCTION_KEYS) hides function keys entry from user-list,
but fkeys can still be (re)configured by long fkey press.

For more information see:
http://wiki.innovaphone.com/index.php?title=Reference9:Phone/Protect

myPBX: Allow asterisk as a wildcard in LDAP search strings

Status Closed
Id 74200

As it is done on our phones.

IP232: Pixel noise with some color values

Status Closed
Id 74221

- changed to lower pixel clock Status: orchid_lcd.cpp

PBX: Conference to VM

Status Closed
Id 74222

Now a call to the VM as a conference member works again.

RTP-DTMF: Digit may get lost during media re-negotiation

Status Closed
Id 74232

Receiver starts handling on END event.
Sender may stop sending before END event was sent.

No media for calls from SIP provider forwarded back to SIP provider in case of NAT and MediaRelay

Status Closed
Id 74290

NAT traversal depends on a packet being sent from inside the NAT to outside, to fix the RTP destination of the outside endpoint. This does not happen if both endpoints are outside.

Dummy packets are sent from the Media Relay function in this case to achieve this.

ISDN: Send HLC with mobility calls

Status Closed
Id 74296

Some ISDN networks refuse the forwarding of a call to a mobile network if no HLC (High Layer Compatibility) Information Element indicating Telephony is included in the call.

IP241,IP222,IP232: Improved phone screen rendering

Status Closed
Id 74301

Reduce map copy operations.

SIP: Accept INVITE's from alternative proxy

Status Closed
Id 74310

If alternative proxy is configured, calls should accepted from alternative proxy,
even if primary proxy is alive and kicking.

Flash Directory: Substring match could miss results

Status Closed
Id 74320

Searching by substring worked case sensitive in some cases, instead of functioning case insensitive.

PBX: CF at Gateway Type objects - additional dialed digits should be added to the destination

Status Closed
Id 74348

This way a CFNR at a trunk object can be used to reroute the call to another trunk.

Gateway: Deleting of routes could result in duplication of routes

Status Closed
Id 74356

This happened if an interface registration was disabled, for which automatic routes have been generated and then a route was deleted. The last route was duplicated.

IP241/222/232: Monitormode (Lauthören): Level too low

Status Closed
Id 74362

Use independent analog codec channels for speaker and headset receiver.
Speaker volume in monitor mode is configured as in handsfree mode.

To change to speaker level in monitor mode the gain of the speaker can be configured with
config change AC-DSP0 RINGER /DualOutputModeGain level
config activate

level is from 0..63 0
0 -> -32dm
32 -> 0db
63 -> 31db

Status: ac_codec3.cpp
ac_codec3.h

IP241,IP222,IP232: Could not activate first builtin background image from phone menu

Status Closed
Id 74367

Could not activate first builtin background image from phone menu if external background image is currently active.

IP241,IP222,IP232: Show multiple diversions on incoming calls

Status Closed
Id 74371

If a call was diverted more than once, the phone shows first diverting party (original called number) and last diverting party.
(Not only the last diverting party)

myPBX launcher: Unhandled exception when accessing browser object

Status Closed
Id 74380

The myPBX launcher could crash when accessing the browser document because of an unhandled exception.

IP232: Pickup fkey displays no information in active state

Status Closed
Id 74392

Pickup fkey displays no information in active state

Phones: Only one diverting party displayed on incoming calls

Status Closed
Id 74406

Only original called party displayed, not the last diverting party.
Phones with non-color display only (IP240,IP230,IP200,IP11,IP150)

NAT: Permanent UDP forwarding did not work

Status Closed
Id 74549

Permanent UDP forwarding was unusable after first forwarded datagram.

PBX Waiting: If used as outgoing dialing device, coder list sent with the call was not correct

Status Closed
Id 74593

Should be based on the coders paramter in the configured URL

Phone_orchid: Residual echo in handset ( sporadic)

Status Closed
Id 74671

Echo canceller NLP sensitivity Mode changed to 2 Status: ac_dsp3.cpp

IP222,IP232: Function key "Hotdesk" did not work

Status Closed
Id 74723

Function key "Hotdesk" did not work

IP232: Fix for long touch on fkey

Status Closed
Id 74743

Opens fkey config, but mist not toggle control on touch-off.

Web-UI: Configuration of "Sync Server" does not return with HTML page

Status Closed
Id 74778

Configuration of "Sync Server" does not return with HTML page

IP222,IP232: changes not saved when leaving menu screen with LEFT key

Status Closed
Id 74792

Even after confirming to save

IPxx10: Flashdir Segments Default to 129(was 51)

Status Closed
Id 74826

IPxx10: Flashdir Segments Default to 129(was 51)

Interop: Don't add an error log "SRTCP authentication failed" on non-RTCP packets

Status Closed
Id 74903

Microsoft Lync send invalid RTP and RTCP packets at the beginning of each call.
In case of encrypted media, decryption of these packets fails.

phone_orchid: ethernet statics counters sometimes displayed as negative numbers

Status Closed
Id 74913

the unsigned counters were printed as signed integers

IP241,IP222,IP232: Volume control not displayed when already at maximum

Status Closed
Id 74927

Volume control is not displayed when trying to increase while being at maximum.

IP6000 crypto driver: Trap when buffers are depleted

Status Closed
Id 74935

Avoid the trap and log an Event when the buffers are depleted.

IP241,IP222,IP232: Mute status indication does not work

Status Closed
Id 74964

"MICROPHONE OFF" is displayed on status bar now.

SIP: Rare problem with blind transfer

Status Closed
Id 74991

callIdentity must be zero when giving ctInitiate to PBX on blind transfer.

TLS: Flow control for incoming data

Status Closed
Id 75004

The TLS socket has to wait for the application to process incoming data before sending the next RECV.

H.323: Renegotiating to Fax was rejected, if PCM switch and local networks were used

Status Closed
Id 75006

If the side which initiated a switch to T.38 has configured PCM and the media address was classified as local due to local network configuration, the T.38 was rejected.

SIP: Interworking of divertingLegInformation1 improved

Status Closed
Id 75025

Honour subscriptionOption

IP222,IP232: Changes not saved in some cases

Status Closed
Id 75088

Some config screens did not write changed settings directly after "Save Changes" dialog.
If menu was left with DISC key (instead of ESC key) the changes have been discarded.

Also affects other phones: IP241,IP240,IP230,IP110
Changes are saved immediately when leaving the current screen.
Not when leaving "User Settings" or "Phone Setting" screen.

PBX Waiting: Diverting leg1 info not correct when diverting to a Waiting Queue

Status Closed
Id 75091

If Waiting queue not defined in root node and escapes are used

PBX: Adjusting received leg1 info wrong

Status Closed
Id 75097

In context of nodes with escapes

VM: <pbx-upd-obj type="cfu"..> without effect when invoked multiple times

Status Closed
Id 75121

Statement <pbx-upd-obj type="cfu"..> failed to work properly after being used for diversion manipulation multiple times within a single script session.

IP28 Watchdog leads to endless interrupt and trap

Status Closed
Id 75186

.

IP800/6000: Problem mit FAX und CNG detect: Option to disable CNG detect added

Status Closed
Id 75278


config change AC-DSP0 /t38-cng-detect-disable 1
config write
config activate

Status: ac_dsp3.cpp/h
ac_fax3.cpp/h

SIP: Decrement Max-Forwards and interwork to H.323

Status Closed
Id 75288

Instead of sending always "Max-Forwards: 70"

H.323: Renegotiation to T.38 did not work for slowstart call thru media-relay to efc endpoint

Status Closed
Id 75305

A common scenario when this happend is an XCAPI (slowstart) calling thru a PBX with media-relay configured to an ISDN interface

Fix for MIPS counter

Status Closed
Id 75310

MIPS counter was incorrect

PBX: Allow 'Max Calls' configuration of 0

Status Closed
Id 75362

So that no call at all is sent to the object but a CFB (if configured) is executed

SIP: Send "305 Use Proxy" if INVITE is received from unexpected source

Status Closed
Id 75380

Applies to registered interfaces only (e.g. phones).

TLS: Possible trap when restoring sessions

Status Closed
Id 75394

Only in v9, depending on the certificate.

PBX Exec Object: Leg2 information send to secretary not adjusted correctly for nodes

Status Closed
Id 75431

The leg2 information is used to display at the secretary the number of the exec which was called. This number was not correct if nodes with escapes were used

Gateway Interface Maps: Should be applied to leg1 info also

Status Closed
Id 75437

The same rules used for a Calling Party Number are applied to leg1 info

SoftwarePhone: Timer tick resolution

Status Closed
Id 75447

The timer tick resolution is increased. This fixes the call RTCP calculation transmitted to the PBX.

IP241,IP222,IP232: Indirect dialing of names not possible

Status Closed
Id 75451

Could not dial a name from 'indirect dialing' screen
by moving the focus to input field and going offhook.
(Only fkey worked)

PBX: CFNR Loop check detected loops that weren't

Status Closed
Id 75465

Esspecially a CFNR at a Waiting Queue was not executed if the destination of the CFNR has transfered the call to the Waiting Queue

Gateway: Event "No Media Data Received" was created for each SWITCH-PCM call

Status Closed
Id 75483

There should be no event in this case, this is normal.

PBX: Conference id prefix/suffix configuration

Status Closed
Id 75507

The broadcast conference PBX configuration is changed: now, the third party conference unit option is saved and must be enabled to use the configured id prefix and suffix. Otherwise they are ignored now and default values for the innovaphone conference interface of the current device firmware version are used. This fixes the configuration if the firmware is updated from V8 to V9 and the innovaphone conference interface is used.
Disabling the "Create Dynamic Conference Id" option in firmware V9 hotfix 5 and 6 is also fixed now.

IP241,IP222,IP232: Call duration display wraps after 100 minutes

Status Closed
Id 75508

Now after 60 minutes the display changes from mode [mm:ss] to [hh:mm].

TLS: Duplicate alert message on malformed ClientHelloV2

Status Closed
Id 75509

Only one alert should be sent per session.

TLS: Improved negotiation of protocol version

Status Closed
Id 75510

TLS server unnecessarily rejected ClientHello messages with TLS 1.1 and higher. Instead of rejecting it should tell the client that it wants to use TLS 1.0.

TLS: Skip empty records

Status Closed
Id 75511

TLS record layer should ignore records with zero length without doing anything.

myPBX: Script error after waking the PC up from hibernation

Status Closed
Id 75532

Sometimes window.open throws an unhandled exception in this case.

SIP: Check proxy availability did not work in any scenario

Status Closed
Id 75542

OPTIONS can be used to poll remote proxy's availablity
to avoid TCP timeout when INVITE is to be sent.
Signaling interface is marked as down and not used anymore.

IP232: Hiding touch keyboard by touching a control

Status Closed
Id 75576

Touching a text control activates the touch keyboard.
Touching the text control again now deactivates the touch keyboard.

IP241,IP222,IP232: CFU information in header bar

Status Closed
Id 75654

CFU information in header bar is now displayed even if there's not is enough space between name and number.
Either name or number is omitted is required.

IP241,IP222,IP232: Replace triangle by arrow to display diversion/transfer information

Status Closed
Id 75689

Replace quite heavy 'BLACK RIGHT-POINTING POINTER'
by much lighter 'RIGHTWARDS ARROW'
to display diversion/transfer information
on call control, fkeys and call lists.

H.323: A forwarded HopCount>32 could result in a very small HopCount

Status Closed
Id 75765

There are only 5 bits for transmitting a HopCount in H.323. A HopCount from SIP is typically 70 and this value was not reduced to 32 but only the 5 lower bits were transmitted, which resulted in a HopCount of 6

IP: Minor memory management change

Status Closed
Id 75770

no use of malloc in ipproc

IP241,IP222,IP232: Two waiting calls are now displayed (instead of one)

Status Closed
Id 75789

Waiting calls are displayed below the two main calls.
Only two lines of information and grey background.

SIP: Handling of some more proprietary Alert-Info and Call-Info in INVITE

Status Closed
Id 75824

Alert-Info: <Bellcore-dr3>;info=alert-recall
and
Alert-Info: <http://not_used.com>;info=alert-autoanswer
and
Call-Info: <sip:127.0.0.1>;answer-after=0

PBX Waiting: Leg2 information sent with calls to operators not adjusted correctly for nodes

Status Closed
Id 75830

If a Waiting Queue was configured in a Node not the root node, the leg2 info was not adjusted corrcectly. The leg2 information is used to signal to the operator which Waiting Queue is forwarding the call

SIP: Decoding of RFC-4412 definitions

Status Closed
Id 75848

The 'Resource-Priority' Header Field
The 'Accept-Resource-Priority' Header Field
The 'resource-priority' Option Tag
417 Unknown Resource-Priority response

SIP: Handling of 422 Session Interval Too Small

Status Closed
Id 75873

Get Min-SE and re-try INVITE

SIP: Send b=TIAS attribute in media description

Status Closed
Id 75881

Send b=TIAS attribute in media description

remove indirect calls to os_mem_alloc() via malloc()

Status Closed
Id 75886

malloc() shall not be used anymore in sources dedicated to run on innovaphone hardware

IPv6: Memory leak if packets received, which are not handled locally

Status Closed
Id 75920

E.g. an IPv6 multicast UDP packet, for some other application was not deleted.

SIP: Fix for video negotiation

Status Closed
Id 75954

Fix for video negotiation

phone_orchid: after leaving a conference the active call was mute sometimes

Status Closed
Id 75955

The conference was not stopped internally in all cases.

Flashman: Show meaningful result after OEM certificate upload in production

Status Closed
Id 75958

"Certificate" instead of "-unknown-"

SIP: Fix for Supported header

Status Closed
Id 76016

Re-implemented to easily add more tags

IP222,IP232: Alpha mode did not work on multi-line edit controls

Status Closed
Id 76165

Could no enter alpha chars using the num block

SIP: Trap when parsing very large History-Info header

Status Closed
Id 76198

Trap when parsing very large History-Info header

SIP: SIP message may exceed available buffer

Status Closed
Id 76203

Trace message like this can occur:
ERROR: SIP message buffer (1025) exceeded! (9,14,801d781c)
SIP: CSeq missing
SIP: Message encoding failed!

SIP: Contact not allowed in BYE/CANCEL/PRACK request

Status Closed
Id 76210

Acc. to RFC-3261 there must not be Contact header line in BYE/CANCEL request
Acc. to RFC-3262 there must not be Contact header line in PRACK request

reset could fail, if issued during firmware update

Status Closed
Id 76266

and even worse any following reset command were not accepted as well

IP241,IP222,IP232: Show "New message from ..." on active phone screen

Status Closed
Id 76283

Show "New message from ..." on phone screen for 3 seconds when active with calls (non-idle).
Like on b/w phones.

IP241,IP222,IP232: Toggle of display-name/name-alias/number was buggy

Status Closed
Id 76290

String termination was missing.
Garbage data was displayed.

IP232: Open call details when touching entry in call list

Status Closed
Id 76292

Instead of initiating call to remote party in call list entry.

H.323: Merges from v10 for Media Negotiation fixes and special OEM protocol features

Status Closed
Id 76301

- Problem with transparent forwarding of SDP
- special SRTP key exchange
- Video fixes
- Unit Testing support

IP-DECT: Memory leak with rejected radio registration

Status Closed
Id 76308

Memory leak occurred with a rejected radio registration is fixed, now.

PBX-SOAP: When monitoring Waiting queue, the peer number was not indicated

Status Closed
Id 76315

The number was considered presentation restricted, because of some bit manipulation error

ip22/24/28/302/305/6010/3010/1060: Fax failed if tones are sent during the fax call

Status Closed
Id 76342

In some cases the Fax messages were missing due to tones send during the fxx call Status: ac_dsp3.cpp

Gatway: Transfer handling may cause re-routing after regular hang-up

Status Closed
Id 76357

Re-routing is expected to take place on non-connected calls only.

PBX Number Map: Not possible to use overlap dialing to Number map with incomplete destination

Status Closed
Id 76377

If a Number Object with incomplete destination was called and the number was to be completed with overlap dialing a wrong number was called.

This is a usefull feature to use Number Maps as quick dial to other nodes. In this case Number Maps are used with a destination of the remote node, so the number is incomplete, the number within this node has to be dialed in addition to the Number of the Number Map object.

SIP: Support for media recording

Status Closed
Id 76396

Media recording to HTTP URL

Gateway: Conference interface, no voice

Status Closed
Id 76419

The ADSP firmware is changed to version 122. This fixes a bug in the conference interface of IP6000/IP6010/... which results in conference calls without voice in one direction for a single member.

phone_orchid: spurious trap in long conference calls

Status Closed
Id 76445

in long confernce calls the phone may trap because the DSP delivers an unexpected zero size packet.

IP241,IP222,IP232: Call held indication

Status Closed
Id 76454

If remote side has put the call on hold
the user should get a visible indication
(in addition to hearing music on hold).

New call status "held" is displayed.

LDAP-Expert, Edit Object Dialog: Could show more than one object

Status Closed
Id 76464

LDAP-Expert, Edit Object Dialog: Could show more than one object

Qsig: Connected Number could be encoded wrong

Status Closed
Id 76472

Qsig: Connected Number could be encoded wrong

Gateway: Better handling of call-reroute requests

Status Closed
Id 76515

Apply interface maps to numbers in reroute request

PBX-SOAP: Better error handling when using a Waiting Queue for outgoing calls

Status Closed
Id 76544

- call was hanging if an invalid number or user without registration was called
- trap if outgoing call was done on operator connect

Spurious leaks in leak checking

Status Closed
Id 76589

A problem which mostly shows up in automated release testing and then requires manual investigation.

SoftwarePhone: Auto start configuration

Status Closed
Id 76595

The duplicate back slash occurred with the install directory is fixed now. Used by the run mode configuration 'At login'.

myPBX launcher: Window position was not always remembered

Status Closed
Id 76626

When the user just moved the window without resizing it, the window position was not remembered for the next session.

V9 Hotfix 9 (90600.11, withdrawn)

Changes included in Version 9 hotfix9 Definition

New Features

PBX-SOAP: UserCall new option diversion override

Status Closed
Id 75773

To initiate calls, which are not diverted by a configured CFU.

Use Info[] element with type="fty-no-cf"

SIP: Support for "Allow-Events: ccnr" and "Allow-Events: ccbs"

Status Closed
Id 76658

For interoperability:
Support for "Allow-Events: ccnr" and "Allow-Events: ccbs"

Phone->PBX LDAP Search returns normalised number to be dialled by phone

Status Closed
Id 77261

1)Phone->PBX LDAP Search returns normalised number to be dialled by phone.
2)Phone receives info about escape digits when registering at its PBX.

With 1) the PBX includes an object's normalized number into the LDAP search result.
With 1) the phone is able to dial that normalized number.
With 2) the phone is able to prefix required escape digits to the received normalized number. Status: checked in to 10.00, 9.00, 90600

PBX soap method SetPresence

Status Closed
Id 78092

PBX soap method SetPresence added to set the presence of a PBX user.

phone ip222, ip232 : USB headsets need not to be configured anymore

Status Closed
Id 78114

If an USB headset with a known signature (vendor/product id) is plugged it is automatically enabled. This is indicated by the headset symbol in the status line.
"Phone/Preferences/Start Outbound Call on Electronic Hook Switch (EHS) Signal" is implied in this case because some headsets will loose state if a hoook signal is ignored.
Status: checked in to 10.00, 9.00, 90600

phone: ip222, ip232: USB headset support - Plantronics C420, GN2000 USB - MS OC Version

Status Closed
Id 78201

. Status: checked in to 9.00, 90600

PBX Mobility: No-Alert checkmark as workaround for provider which do not provide Alerting

Status Closed
Id 78410

Some SIP provider do not provide an Alerting signal when a mobile phone is called. This could result in no ringback signal to the caller or the min/max-alert feature not working.

This new checkmark provides a fake Alerting in case Progress is received

show linux shutdown warning on firmware reset page

Status Closed
Id 78665

If linux is running, a warning is now shown, that linux should be shutdowned before a firmware reset is performed.

Linux: Enable/Disable support link

Status Closed
Id 78786

Now the Linux menu is always shown and a link is provided to enable or disable the Linux support (RAM reservation). The support state is also saved in the downloaded configuration file and restored with the upload.
Update: The support state is only saved in the downloaded configuration with password. The state is not saved in the configuration file with standard password or if downloaded by the update server. Please use the next or a later hotfix instead, see also fix #78836.

Bug Fixes

H.323: Media Negotiation problem with conferences on IP-DECT

Status Closed
Id 76314

A channel was not switched to the conference after a hold/retrieve cycle

PBX Mobility: Dialed digits could get lost, when using Opticaller

Status Closed
Id 76598

There could be a collision of a dialed digit with media renegotiation. For example if with the first digit a media was switched to inband information from a carrier.

phone: Ring Tone Titles containing apostrophes garble phone configuration

Status Closed
Id 76745

When under "Phone/Ring Tones/Add Ring Tone" a title containing apostrophes is entered the page "Phone/User-x/Preferences" cannot be edited anymore because of a XML-Error. Status: checked in to 10.00, 9.00, 90600

SIP: Record-Route handling on outbound subscriptions

Status Closed
Id 76831

Processing of Record-Route in SUBSCRIBE response.
Simply missing.

phone: Pickup fuction key is not displayed while all alerting calls are displayed on a Partner key

Status Closed
Id 77393

alerting calls displayed on a Partner key are not displayed on the Pickup key.
if nothing has to be displayed on the pickup key the key should not disappear but display the 'idle' label Status: checked in to 10.00, 9.00, 90600

H.323: Media Negotiation problem with transfer in Gateway (not PBX)

Status Closed
Id 77683

Under special conditions a blind transfer happend in the Gateway could result in a call without media. This only happened if the call was transfered twice and the destination of the first transfer was a physical interface.

PBX Twinning: When calling another (twin) phone, the call was sent to the original phone also

Status Closed
Id 77829

It still works, but this waiting call could be confusing

PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX

Status Closed
Id 77874

A call from a object within a node with escapes on a slave PBX was not routed to the master if the destination was within the same node and not known on this slave but was sent to the node-extern destination directly

PBX Waiting: Name Id missing in calls initiated with SOAP

Status Closed
Id 77911

When a Waiting Queue is used by applications to initiate outgoing calls, the name of the waiting queue should be sent with these calls as calling name. This name id was missing

RTP: Potential random trap when closing channels

Status Closed
Id 77918

Happens if there is a collision with a received packet and closing of the channel. Window for this is very small, so it should happen very rarely. Probability can increase with high load.

Web-UI: Visually separated commands

Status Closed
Id 77954

Visually separated commands 'Clear' and 'Save' on Maintenance/diagnostics/Events.
Also 'download' and 'delete' on General/License.

IP241,IP222,IP232: Show special symbol for call completion entries in call list

Status Closed
Id 77966

Show special symbol for call completion entries in call list to separate from usual missed calls.

IP241,IP222,IP232: Status message "Recording" not displayed

Status Closed
Id 77992

Status message "Recording" needs to be displayed during recording.

Gateway: Wrong Media info sent for calls forwarded from a VOIP interface to another VOIP interface

Status Closed
Id 78024

This could cause wrong information to be displayed on a PBX calls page. Esspecially SRTP was not indicated.

Better to not send any Media Info in this case.

IP241,IP222,IP232: Wrong display name in call list

Status Closed
Id 78047

Name identification of remote party is to be displayed,
but name identification of diverting party was displayed instead
on incoming calls list.

phone: the call which was the active call at start of a conference was not automatically cleared on a release from remote

Status Closed
Id 78086

The call which was the active call at start of a conference had to be cleared manually but the call which was on hold at start of conference was cleared automatically.
Now any call will be automatically cleared when relesed from remote. Status: checked in to 10.00, 9.00, 90600

phone_orchid: remaining call mute after remote relase for the call which was the active call at start of a conference

Status Closed
Id 78091

did not happen after a remote release for the call which was on hold at start of the conference. Status: checked in to 10.00, 9.00

phone: ip222,ip232: Muting USB headset micro via microphone key did not work but MICROPHONE OFF was displayed

Status Closed
Id 78159

. Status: checked in to 10.00, 9.00, 90600

IP241,IP222,IP232: Trap in font rendering

Status Closed
Id 78210

Some characters at beginning of line may cause trap.

Gateway: Trap when interworking Call Completion

Status Closed
Id 78228

Trap when interworking Call Completion.

LOG CALL 6 A:Call -> / PRI2::->*::
R_CALL free error c18a59b8

phone: ip241: memory leak when ip230x extension module is attached and used

Status Closed
Id 78311

one packet lost per keystroke on extension module

PBX: Dynamic group function keys did not work with non-ascii characters in group name

Status Closed
Id 78322

Name was not correctly converted to internal utf-8 representation

phone: ip222,ip232: trace USB media data only if explicitely requested

Status Closed
Id 78326

use "config add USB-HOST AUDIO /trace" if data is needed

H.323: Potential Trap in case of special malformed RAS registration

Status Closed
Id 78368

A NULL pointer access could happen

SIP: Removed b=TIAS attribute from media description

Status Closed
Id 78379

Causes trouble on some SIP providers (neotel.at)

DHCP server leases with hostnames containing non-ascii latin1 characters (for example Umlauts) could not be displayed

Status Closed
Id 78382

In pre-V9 firmware hostnames were stored latin1-encoded. Names contaning non-ascii latin1 charaters must be converted to UTF8 before display. Status: checked in to 10.00, 9.00, 90600

MyPBX: Changing password did not work any more

Status Closed
Id 78391

Collateral damage of fix #75453: PBX/IP-DECT: User password length

phone: ip222, ip232: USB headset sometimes mute after disconnect

Status Closed
Id 78414

depending on isochronous transfer state at disconnect time. seems to happen more often with relatively cheap headsets.

SIP: Be save against sudden death of SIP caller

Status Closed
Id 78460

Lifetime of an INVITE trasnaction is not limited by any timeout
after provisional response has been send/received.
Sudden death of a caller make calls hang forever.
Now overall lifetime of an INVITE server transaction is limited to 3 minutes.
After expiration fimnal reject response is sent and call is released.

IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax failure after transfer

Status Closed
Id 78487

channel was reconfigured on remote switch to T38, without a new activate fax relay command Status: ac_dsp3.cpp

PBX Boolean: Avoid unnecessary load by boolean monitoring

Status Closed
Id 78504

A call was sent to the monitoring endpoint every 10s

IP241,IP222,IP232: No RTP send on outgoing call in some cases

Status Closed
Id 78550

No RTP send on outgoing call in some cases

IP6000: Traps in DSP driver under high load

Status Closed
Id 78591

under high load timing may change. Checks in driver relaxed to take this into account.

PBX-SOAP: Limit on size for Admin call too small

Status Closed
Id 78592

Only 4k was allowed

SoftwarePhone: Default ToS value for RTP

Status Closed
Id 78593

Now the default ToS value for RTP packets is 0xB8.

PBX Filter: Maximum length increased from 13 to 17 digits

Status Closed
Id 78594

13 digits could be too little if an international number including a prefix of an external line was to be checked

IP241,IP222,IP232: Wrong call direction indication on PARTNER fkey

Status Closed
Id 78642

When partner calls someone else it my displayed as if partner is called.
(Only if option "Show connected Party in busy State" is enabled)

IP222 IP232 IP241: Sporadic noisy sidetone

Status Closed
Id 78649

Analog and digital sidetonegain were enabled,now only the analog sidetonegain is used. Status: ac_codec3.cpp

Gateway: Handling of blind transfer gets stuck in routing

Status Closed
Id 78709

Routing of transfer call stops with: reason='incomplete'
But during transfer no dialing digits will follow.

V9 Hotfix 10 (90600.12)

Changes included in Version 9 hotfix10 Definition

New Features

Linux: Enable/Disable configuration

Status Closed
Id 78836

With the new feature #78786 the configuration is only saved in the downloaded configuration file with password. Now the information is also included in the configuration file with standard password and in the file downloaded by the update server.

Bug Fixes

PBX: Trap if enabling Unknown Registrations

Status Closed
Id 78904

Collateral Damage from

fix: #77261: Phone->PBX LDAP Search returns normalised number to be dialled by phone

V9 Hotfix 11 (90600.14)

Changes included in Version 9 hotfix11 Definition

New Features

HTTP-Client: MD5-sess authentication

Status Closed
Id 77773

HTTP Digest Authentication with alogrithm=MD5-sess.
Choose the first supported "WWW-Authenticate" line from 401 response headers.

Needed for new versions of IIS.

Status: http://wiki.innovaphone.com/index.php?title=Support:DVL-Feature_Requests#HTTP_Client

Phone: Possibility to reject incoming SIP calls with customized reason phrase

Status Closed
Id 77928

E.g. "SIP/2.0 480 Do not disturb" instead of "SIP/2.0 480 Temporarily unavailable"

Active Directory Replication: Editfield for Poll Timer added

Status Closed
Id 78631

If change notifications cannot be received from an AD, a poll timer can be specified. A re-replication is going to take place after the poll timer expired.

X.509: Support for PKCS#12 files

Status Closed
Id 78820

Support for certificate import using password encrypted PKCS#12 files.

Currently the following encryption types are supported:

pbeWithSHAAnd128BitRC4 (1.2.840.113549.1.12.1.1)
pbeWithSHAAnd40BitRC4 (1.2.840.113549.1.12.1.2)
pbeWithSHAAnd3-KeyTripleDES-CBC (1.2.840.113549.1.12.1.3)
pbeWithSHAAnd128BitRC2-CBC (1.2.840.113549.1.12.1.5)
pbewithSHAAnd40BitRC2-CBC (1.2.840.113549.1.12.1.6)

myPBX: Pass language code to reporting

Status Closed
Id 78874

For localization of the call list.

Phone: Show number of voice messages on label of MWI fkey

Status Closed
Id 78894

Use "%u Message(s)" as label text for active state of MWI fkey.

IP-DECT: Cisco SIP conferencing

Status Closed
Id 78988

Cisco proprietary softkey events implemeneted to start a 3pty conference

IP-DECT: Feature Code for calling predefined service URIs

Status Closed
Id 79028

User for Cisco compatibility

Cisco Softkey Features, Status updates and Park

Status Closed
Id 79030

suported_mask needs more flags
Softkey Feature Park

IP-DECT: Cisco Park

Status Closed
Id 79032

Send Park softkey event

IP-DECT: Forward Cisco Status Line Updates related to a call

Status Closed
Id 79036

.

IP-DECT: SRTP for an OEM device

Status Closed
Id 79061

Now SRTP is configurable for an OEM device.

IP-DECT: Forward Cisco Status Line Updates not related to a call

Status Closed
Id 79104

.

Merge of v10 changes: Dynamic Timer Management/oem serial no

Status Closed
Id 79127

Better power management on some platforms
Display of oem serial no
stanard behaviour of delete on NULL

Merge of v10 changes: Memory Management

Status Closed
Id 79129

for OEM needs

Merge of v10 changes: Firmware File check when updating improved

Status Closed
Id 79130

for OEM needs

Merge of v10 changes: More ciphers

Status Closed
Id 79134

for OEM SRTP and other Certificate formats

Merge of v10 changes: Internal interfaces enhanced

Status Closed
Id 79135

for OEM support

Merge of v10 changes: ZipZip Tone definition updates

Status Closed
Id 79136

should overlay normal voice

Merge of v10 changes: Library update

Status Closed
Id 79140

.

Merge of v10 changes: permit DHCP mode change between client/disabled without reset

Status Closed
Id 79152

for OEM support

ASN.1 BER: Support for indefinite-length encoding

Status Closed
Id 79260

needed for PKCS#12 import
only decoding

ASN.1 BER: Support for constructed octet strings

Status Closed
Id 79261

needed for PKCS#12
only decoding, only one level of nesting

IP810: Config Wizard

Status Closed
Id 79284

Mechanism needed to add platform specific files (IP810 uses same firmware as IP6010, IP3010, ...)

Merge of v10 changes: Support for SIP specific phone features

Status Closed
Id 79306

for OEM support Status: checked in to 9.00

phone: permit to select the directories to be searched on indirect dialing

Status Closed
Id 79307

if there is for example a huge external directory used for inbound name resolution and dialing is restricted to internal partners it may be hard to find internal numbers via combined directory search.

config add PHONE DIR-UI /dial-mask <selection>

selection:
0x02 - local directory
0x04 - PBX directory
0x08 - external directory

0x06 = local + PBX
0x0E = local + PBX + external

IP-DECT: Signal waiting calls to handset

Status Closed
Id 79349

Was only acoustic information.

IP-DECT: LDAP phonebook (IP1202)

Status Closed
Id 79483

The new feature LDAP phone book for the IP1202 is added now.

IP-DECT: Three party conference with innovaphone PBX

Status Closed
Id 79503

Now it is possible to make a three party conference with DECT handsets with an innovaphone PBX (an innovaphone device with the CONF interface). The conferencing unit must be configured in the DECT master. The conference call is established with the feature code 'R' + '3'.
This fix also includes a rework of the DECT radio module. It can handle more than one waiting or hold call now.

Gateway: 'cn' attribute for test interfaces

Status Closed
Id 79506

The 'cn' attribute is included in the test interfaces SIG0/1 for internal tests now.

Bug Fixes

IP222/IP232/IP241: Ethernet link configuration

Status Closed
Id 78067

In some case ethernet link configuration and display didnt work.

Now the autoneg status is shown if available, and the phy status if no autoneg didnt complete.

An option to use autoneg with fixed speeds and a link trace option (/ltrace) are added. Status: orchid_drv.cpp
orchid_drv.h
orchid_drv.xsl

Media Relay: Don't send dummy RTP data on incoming calls

Status Closed
Id 78252

Dummy RTP data is sent just in case a NAT router is within the media path to set a UDP mapping in case both legs of the call contain a NAT router.

An example for such a situation is a call coming from a SIP provider thru a NAT router to the PBX, which forwards the call back out to the SIP provider. The NAT router won't get RTP data from inside to set the mappings.

The dummy RTP was sent to all legs of the call, but it is better to send it to outgoing call legs only, because endpoints calling in may turn off a local ringback tone when receiving dummy RTP

H.323/SIP: Avoid delayed SDP within outgoing calls as far as possible

Status Closed
Id 78466

If media renegotiation is needed, to one side of the call an request for a media proposal (in SIP terms, this is an INVITE without SDP) is sent. The media proposal (in SIP terms SDP offer) is then forwarded to the other side.

The request for an offer should if possible not sent with the initial call, because there is equipment which does not handle 'delayed SDP'

Media Relay: Don't terminate T.38 protocol in media relay, forward transparently

Status Closed
Id 78610

This should add robustness and reduce CPU load

Flash Directory: Display Error Messages

Status Closed
Id 78656

Flash Directory: Display Error Messages
Shedding light on replication problems.

currently only in 10.00

RTP: No check for changed media destination because of received multicast

Status Closed
Id 78698

When receiving RTP normally a check is done if the source of the RTP is the same as we are sending to. If this is not the case, we assume the destination of the RTP is behind a NAT router and we change the destination address to the source address of the received RTP.

This does not make sense for received multicast.

IP22 IP24 IP28 IP305: Faxempfang gelegentlich gestört

Status Closed
Id 78803

DSP erhält zum falschen Zeitpunkt ein close-rtp. Status: ac_dsp3.cpp

IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax receive max packet size increased to 1024

Status Closed
Id 78811

For internet fax devices Status: ac_fax3.cpp

phone: ip222, ip232: USB controller sometimes hangs

Status Closed
Id 78824

happened with certain headsets after fast connect/disconnect sequences Status: checked in to 10.00, 9.00
merged to 90600

SIP: SDP answer sometimes contains 2 media descriptions for audio

Status Closed
Id 78833

... instead of one for audio and one for video.

Offer:

v=0
o=- 1 2 IN IP4 130.30.1.111
s=CounterPath Bria
c=IN IP4 130.30.1.111
t=0 0
m=audio 1902 RTP/AVP 0 8 18
a=alt:1 1 : 1A98X9/s 31UwK5IA 130.30.1. 111 1902
a=fmtp:18 annexb=yes
a=rtpmap:18 G729/8000
a=sendrecv
a=x-rtp-session-id:ACE43888C33A4AC8895C7F8F69380914
m=video 28212 RTP/AVP 115 34 123 124
a=alt:1 1 : yjz/mbyQ aWGuD8xJ 130.30.1. 111 28212
a=fmtp:115 QCIF=1;CIF=1;I=1;J=1;T=1
a=fmtp:34 QCIF=1;CIF=1
a=fmtp:123 profile-level-id=42801e; packetization-mode=0; max-mbps=48600
a=fmtp:124 profile-level-id=42801e; packetization-mode=1; max-mbps=48600
a=rtpmap:115 H263-1998/90000
a=rtpmap:34 H263/90000
a=rtpmap:123 H264/90000
a=rtpmap:124 H264/90000
a=sendrecv
a=x-rtp-session-id:694677A9F65D419FB7B9E451FFCA02CC


Answer:

v=0
o=- 3141 1 IN IP4 10.230.5.1
s=-
c=IN IP4 10.230.5.1
t=0 0
m=audio 16410 RTP/AVP 0 13
b=TIAS:64000
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
m=audio 16410 RTP/AVP 0 13
b=TIAS:64000
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv

Conference: Permanent noise in conference channel

Status Closed
Id 78838

Permanent noise in conference channel.
Increasing with every new participant.

PBX: When doing a show users, unnecessary data was sent to the browser

Status Closed
Id 78878

The data could be reduced and thus speeding up the display of big users lists.

PBX Admin UI: Editing profiles was not possible if the user had too many contacts

Status Closed
Id 78883

The form is now submitted using POST instead of GET.

SIP: Wrong number of waiting messages (MWI)

Status Closed
Id 78890

MWI: Number of voice messages not decoded from incoming NOTIFY(application/simple-message-summary).
Was either 1 or 0.

SIP: Send even anonymous Diversion header

Status Closed
Id 78954

If diverting party's identity is unknown/hidden.
Diversion: ;reason=user-busy

SIP: Failed to decode presence XML from CUCM

Status Closed
Id 78967

Support for namespace "urn:ietf:params:xml:ns:pidf:status:rpid" added.

IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax failure after transfer #2

Status Closed
Id 78969

tone flag was not disabled Status: ac_dsp3.cpp

H.323: Media problem with SIP call-completion on IP-DECT

Status Closed
Id 78970

There was no media on the successful completion call (SDP recived in Progress was not forwarded)

SIP: Send 'Connected Number' in P-Asserted-Identity header of 200/OK

Status Closed
Id 78974

Send 'Connected Number' in P-Asserted-Identity header of 200/OK
if different from original called number.

SIP: Re-transmission of ACK with SDP answer

Status Closed
Id 78975

Handling of re-transmissions of 200/OK with SDP offer.
ACK must contain SDP answer.

IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: DSP packet debug didnt show some packets, version endian was wrong

Status Closed
Id 78985

cleanup Status: ac_491.cpp
ac_494.cpp

IP-DECT: Radio/Master calls switched to slowstart

Status Closed
Id 78987

If first media answer was received with a PROGRESS message, the call leg from the radio to the master was switched to progress (no EFC features are defined for the Progress message).

This could cause media problems later in the call with hold/retrieve/transfer.

SIP: REFER does not work as expected on Gateway interfaces without registration

Status Closed
Id 79008

Bad side effect of a previous fix in HF8:
http://mantis.innovaphone.com/view.php?id=76309

Introducing new SIP interop tweak "Take Refer-To URI as Remote Target URI"
to be found on the Gateway interface config (GWx).
See http://mantis.innovaphone.com/file_download.php?file_id=66299&type=bug

IP-DECT: Better handling when receiving a call, while a previous call is being released

Status Closed
Id 79022

Call to service number, which cause a callback can create problems.

Edss1 Interworking: Sending divertingLegInformation2 as PublicPartyNumber(was UnknownPartyNumber)

Status Closed
Id 79025

Such that the type of number resulting from an interface cgpn-map can also be asn.1 encoded.

Gateway: Mapping of Message Center Number of MWI

Status Closed
Id 79039

Mapping of Message Center Number of MWI according to CGPN mapping.

Qsig Interworking: Apply interface CGPN map to DGPN

Status Closed
Id 79060

Qsig Interworking: Apply interface CGPN map to DGPN

PBX: A CFB at a PBX object was executed on call to busy user

Status Closed
Id 79199

It should only be executed, if the calls exceeded the Busy On number.

PBX: Mapping of Message Center Number of MWI

Status Closed
Id 79232

Mapping of Message Center Number of MWI.

PBX-SOAP: Conference Id was missing in call-info for parked calls

Status Closed
Id 79243

This caused problems with the innovaphone operator

PBX-SOAP: UserClear cause should not be used for remote party

Status Closed
Id 79249

This can result in strange call clearing without tones

PBX: Include error in resulting XML if submit-object with insufficient rights was attempted

Status Closed
Id 79289

Problem with innovaphone operator.

phone: ip222,ip232,ip241: wrong AM/PM time display on status line

Status Closed
Id 79302

15:19 PM was displayed instead of 03:19 PM Status: checked in to 10.00, 9.00, 90600

Voicemail: Default for mwi.basicService now Speech(1) (was allServices(0))

Status Closed
Id 79303

Voicemail: Default for mwi.basicService now Speech(1) (was allServices(0))

May from now on be overriden by variable "$_pbxmwiservice".

PBX: Local objects could not be called from Nodes with escapes as expected

Status Closed
Id 79317

Was not possible to call at all, or the number was wrong

PBX-Trunk: Handling for Incomplete, Invalid, Busy, No Anser destinations for Media calls only

Status Closed
Id 79319

For example this should not be done for presence/dialog subscriptions

Refactored some ASN.1 BER Handling

Status Closed
Id 79357

Became necessary in the run of: #79260: ASN.1 BER: Support for indefinite-length encoding

PBX: Conference &amp; VM

Status Closed
Id 79359

The last connected user is not disconnected although it is configured. It occurs if a VM PBX object forwards the call to the BC Conference object (like the innovaphone conferencing script). It is fixed now.

IP-DECT: Subscription could get lost randomly with logout/login cycle

Status Closed
Id 79397

Problem with user database handling

H.323: Media problem if PBX rtp-proxy is activated for CFNR to external destination

Status Closed
Id 79430

no voice

HTTP: Could not have sockets with same ports but different remote addresses

Status Closed
Id 79458

This was a general problem with the new TCP stack, used together with IP6. Problem only happened for HTTP because only for HTTP this new stack is currently used.

ISDN: Fix for call completion interworking

Status Closed
Id 79498

CCBS/CCNR does not work in some cases



SIP: From-URI may got lost after call transfer

Status Closed
Id 79536

UPDATE may be send out missing From-URI.

AD Replication stuck after connection loss

Status Closed
Id 79541

Didn't reconnect. Side effect of after-hf10 fix

PBX Waiting/Broadcast: Incomplete CDRs if CFNR configured on object

Status Closed
Id 79549

If the CFNR was executed, the CDR for the call to the Waiting Queue/Broadcast object ended without release.

PBX: Web User interface problem with IE and non-ascii PBX names

Status Closed
Id 79568

For example groups at a user assigned to a PBX with non-ascii characters could not be edited.
The problem is a bug in IE XSL translation which does special handling of href attributes. Same thing with onclick attribute works.

ip3010 TEL port not working in NT mode

Status Closed
Id 79570

TEL port of ip3010 gateways configured in NT mode do not get Physical Link up indication. This problem applies to V9hotfix7 up to V9hotfix10. Status: ip6010.cpp

SIP: Max forward value of 32 could be too small for some provider

Status Closed
Id 79578

For a starting value of max-forwards a value of 32 was used, because this is the maximum value in H.323. This was too small for some sip providers. Starting value now increased to 64 and on H.323 the half value is transmitted.

SIP: Re-negotiation for T38 did not work in media-relay scenarios

Status Closed
Id 79583

Bad SDP answer was generated.

Status: Introduced by
#77277: SIP: SDP answer must have the same number of media descriptions as received offer

V9 Hotfix 12 (9061009)

Changes included in Version 9 hotfix12 Definition

New Features

myPBX: Syslog for sessions and application sharing

Status Closed
Id 77289

Syslog entries should be useful for debugging session timeouts and problems with the external application sharing solution.

SIP: Pass display names of <dialog-info> to phoneapp

Status Closed
Id 79543

Pass display names of <dialog-info> to phoneapp.

Gateway: Overlap dialing timeout configurable

Status Closed
Id 79639

The default of 4s is not good for all applications. Now configurable between 500ms and 6000ms.

IP-DECT: OEM Configuration of Cisco Features changed

Status Closed
Id 79684

should depend on installed license

myPBX launcher: Automatically move main window into visible area of screen

Status Closed
Id 79697

The window might be outside the visible area if the user changed the screen resolution or disconnectes one of the screens.

If the main window is not visible on any screen, it is now moved to the center of the primary screen, when the user clicks the tray icon.

PBX-SOAP: FindUser optional argument 'nohide'

Status Closed
Id 79734

Needed if the FindUser is used to look for users which have the 'Hide from LDAP property' set. Status:

PBX Mobility: Pick mobile call on fixed phone

Status Closed
Id 79794

by dialing number of Mobility object on fixed phone

Gateway: Allow setting of system time from ISDN time

Status Closed
Id 79889

In ISDN connect messages a public network is sending a local date and time. By a configuration option this can now be used to update the system time

IP-DECT: New Master/Radio behavior for license incompatibility

Status Closed
Id 79914

The behavior of the DECT Master is changed, if an unlicensed Radio tries to register in. This is only used for an OEM license model.

PBX: Better handling of presence information without activity

Status Closed
Id 80006

This type of presence information could contain a note which refers to the future

RPCAP trace: Indicate transmit/received packets by setting the remote mac address to 00-90-33-00-00-00

Status Closed
Id 80152

When reading a trace it is currently not obvious if a packet is sent or received, we need to find out the devices IP address, e.g. by reading the config. If the devices on MAC adress is used a source only if a packet is sent and as destination only if a packet is received this process is simplified.

This is now done for UDP/TCP traces!

SIP: Support for Alert-info:<Bellcore-dr1>

Status Closed
Id 80174

Proprietary tagging of internal calls

phone: ip222,ip232: more USB headsets supported

Status Closed
Id 80224

for a complete list see http://wiki.innovaphone.com/index.php?title=Reference9:Concept_USB_Headset

Status: checked in to 10.00, 9.00

Phones: Switch for phoneapp to disable auto-answer

Status Closed
Id 80233

Disable/enable auto-answer support on phoneapp level.

PBX: DECT attributes for DECT security

Status Closed
Id 80300

In preparation for the new DECT feature DECT security there will be new attributes for the endpoint data which must be taken over. With this fix the innovaphone PBX supports the new attributes if the user is edited.


phone: ip222, ip232: Jabra USB Headset feature "Reject incoming call" supported now

Status Closed
Id 80401

A double tap on talk button rejects a ringing call with 'busy'

IP-DECT: Signal waiting calls to handset

Status Closed
Id 80408

New event to the DECT system to add a waiting call to the call list.
This is step two of the implementation and related to the case #79349.


IP-DECT: Anonymous endpoint information

Status Closed
Id 80421

Information about anonymous endpoint is added to the event to the DECT system. In preparation for the new feature "easy subscription".

IP-DECT: DECT security

Status Closed
Id 80424

Some changes in preparation for the new feature "DECT security". This feature will be available later.

Phone: Accept MWI from Exchange Server

Status Closed
Id 80446

MS Exchange Server sends unsolicited NOTIFY(message-summary) to served user
with served user's number as destination and origin.
But phones expect to receive MWI message center number as origin.
MWI fkey would not light up.

SIP: New config option /take-zero-addr-for-hold

Status Closed
Id 80516

From now on "c=IN IP4 0.0.0.0" is no longer accepted as hold signaling.
Config option /take-zero-addr-for-hold is introduced to get back old handling.

IP-DECT: User log in, endpoint data added

Status Closed
Id 80620

During the user log in with a change of the IP-DECT Master the endpoint data (product id/software version) is sent to this new one now.

SIP: Pass display information to application

Status Closed
Id 80632

Pass display information received in Call-Info header in 200/OK for BYE to app.

IP-DECT: Idle display update

Status Closed
Id 80654

Handling of idle display update message in the call release message is added. Used with an OEM PBX.

myPBX launcher: Configurable hotkey action

Status Closed
Id 80684

Options are:
- Copy selected phone number to myPBX
- Show myPBX

myPBX launcher: Autocomplete configured URLs

Status Closed
Id 80689

If the user configures just an IP address, it shall be replaced by the full myPBX URL.

For example 192.168.0.10 will be replaced by http://192.168.0.10/PBX0/MY/client.htm.

Bug Fixes

PBX: CFNR at Gateway object with incomplete destination

Status Closed
Id 79605

A CFNR at a gateway object is executed if there is no registration. Any additional digits dialed should be added to the CFNR destination. This did not work if the original CFNR destination was incomplete and only completed with additional digits dialed.

PBX Waiting: Filter did not work anymore for CFNR

Status Closed
Id 79610

Collateral damage from

fix: #75465: PBX: CFNR Loop check detected loops that weren't

LDAP Server caused crash on port-scanner attack

Status Closed
Id 79634

LDAP Server caused crash on port-scanner attack

myPBX: Quotes in connected names caused java script errors

Status Closed
Id 79636

When the connected name of a call contained the ' character there was a script error.

PBX: Status displayed on boolean function key could be wrong

Status Closed
Id 79643

In case of severe network problems, it could happen that the status displayed on a Boolean function key was wrong and was only corrected when the boolean status changed.

DHCP Server Identifier was cleared after editing the DHCP-Server page

Status Closed
Id 79651

The value of "IP4/ETHx/DHCP/Server Identifier" was cleared when the OK or Renew button was pressed on the "IP4/ETHx/DHCP-Server" page. This bug was introduced with V9hotfix5.



SIP: Wrong Contact-URI in outgoing NOTIFY

Status Closed
Id 79700

Contact-URI should match the Request-URI of the SUBSCRIBE.
Also the Message-Account URI in "simple-message-summary" was wrong as result of the wrong Contact-URI.

Config: Could not dynamically set or reset /trace on the LICENSE module

Status Closed
Id 79709

Missing handling of MODULE_UPDATE.

PBX: Allow call completion only if recall is not prohibited by CFU

Status Closed
Id 79739

A CFU loop results in a rejection with busy. A subsequent call completion attempt was allowed and a recall possible was signaled right away. This was very confusing.

SIP: Handling of Call Pickup

Status Closed
Id 79741

Failed to handle INVITE with Replaces as Call Pickup.
Must redirect Pickup call to alerting party.

myPBX: Problem with checking browser capabilities in Firefox

Status Closed
Id 79757

When checking the supported browser features, Firefox thows an uncaught exception if cookies are deactivatd by the user. Therefore myPBX is stuck in the "Loading" screen instead of displaying a configuration hint.

IP241,IP222,IP232: Show lengthy number information on Partner fkey

Status Closed
Id 79779

Toggle between 'show head' and 'show tail' presentation.

SIP: DNS resolving of STUN server failed

Status Closed
Id 79788

Wrong STUN server port used (5060 instead of real port).

Fax: Channel memory leak

Status Closed
Id 79815

Memory leak is fixed in fax channel now.

PBX: Export to CSV, utf-8 byte order mark was missing

Status Closed
Id 79844

Without the utf-8 BOM the file was not correctly interpreted as utf-8 by Microsoft Excel and other applications.

PBX: For registrations containing name and number, use name to identify the device

Status Closed
Id 79846

It did depend on the sequence of the name and the number. If the number was first, the first device was selected and the name was ignored. Now the name is used to select the device regardless of sequence.

This is a problem with endpoints which always send name and number for registration.

SIP: CLEARMODE does not work since Hotfix10

Status Closed
Id 79858

Invalid SDP answer is send:

v=0
o=- 4 3 IN IP4 10.28.108.8
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 0 RTP/AVP 125 101
a=rtpmap:125 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000

memory violation in http client when processing an URI > 8000 byte

Status Closed
Id 79860

happens when the digest authentication header is constructed




Status: checked in to 10.00, 9.00

PBX Broadcast: Forking/Mobility configured at Broadcast object did not work

Status Closed
Id 79863

Mobility can be used now for forking with delay

Potential trap in packet libraray

Status Closed
Id 79866

Ther was a not obvious size limit for packet::put_head and packet::put_tail, which caused a trap if the size was exceeded. This could happen if a CDR exceeded a certain size.

SIP: Domain Name System (DNS) names compared case sensitive

Status Closed
Id 79887

Domain Name System (DNS) names are "case insensitive"

SIP: Interworking with "AUDC-IPPhone"

Status Closed
Id 79892

RFC-3551 4.5.2
Even though the actual sampling rate for G.722 audio is 16,000 Hz,
the RTP clock rate for the G722 payload format is 8,000 Hz because
that value was erroneously assigned in RFC 1890 and must remain
unchanged for backward compatibility. The octet rate or sample-pair
rate is 8,000 Hz.

Codec description must be: a=rtpmap:9 g722/8000
but "AUDC-IPPhone" sends: a=rtpmap:9 G722/16000
in SDP offer.

SIP: Problems with DNS resolving of proxy adresses

Status Closed
Id 79907

If resolving of the primary proxy failes, the backup proxy is never resolved.

IP-DECT: Reset required if Radio password changed

Status Closed
Id 79929

Now reset required is shown if the Radio password for the Master registration is changed.

Phone: PBX (operator) initiated outbound call was connected but mute when another call was ringing on phone already

Status Closed
Id 79943

the phone was not switched from ring mode to handsfree/headset mode

SIP: Interworking with HD audio client

Status Closed
Id 79953

Support for unknown codecs.
Passing transparentley.

Phone: Out-Of-Memory-Trap

Status Closed
Id 79980

If LOG server is configured but not reachable
the device will buffer arising LOG entries until a limit of 300kB.
This limit was to high for old black/white telephones.
Now not more then 1% of DRAM size is used for LOG buffer.

PBX: When editing a Node object it was changed to a PBX object

Status Closed
Id 79982

This was a collateral damage of fix: #78878: PBX: When doing a show users, unnecessary data was sent to the browser

ISDN: Problem configuring negative volume levels

Status Closed
Id 79992

Collateral Damage from
79028: IP-DECT: Feature Code for calling predefined service URIs

Dect: AD replication makes IPEI disappearing on all users

Status Closed
Id 80015

Side effect of a previous fix #72672 aiming at ad-replicated and deleted objects coming back to life.

myPBX: Setting CFx with boolean did not work in IE8

Status Closed
Id 80018

Using Internet Explorer 8 the selected Boolean object was not saved.
Internet Explorer 8 requires option tags to have a value attribute.

IP-DECT: Handover with IP1202

Status Closed
Id 80022

Only one handover is possible since the last fix with the IP1202. It is fixed now.

Name and Number configured for a registration must be completely replaced by registration result

Status Closed
Id 80039

If for example Number and Name is configured but the registration result provides the Number only, the configured Name must not be used in further signaling operations (diversion queries ...)

Status: checked in to 10.00, 9.00

PBX: Possible trap on calls from misconfigured nodes/PBXs (node parent loop)

Status Closed
Id 80093

If a node or PBX is configured with a parent node configured to itself in the most simple case, a call from an endpoint configured for this node to a destination which cannot be found in this node, will cause a trap.

This is a collateral damage of fix: #79317: PBX: Local objects could not be called from Nodes with escapes as expected

IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305 IP222 IP232 IP241: receveid RTP packets limited to 480bytes

Status Closed
Id 80111

increased to 640 Bytes to allow G711 80ms.


DHCP Survivability mode doesn't work after a DHCP restart

Status Closed
Id 80112

``Survivability´´ mode is used by WLAN phones. In this mode the settings of the 'saved lease' (IP addr, mask, ...) are used until a fresh lease is received. The 'saved lease' is the last lease received from a server, it is kept over a reboot.

When WLAN coverage is lost for a while and then regained a DHCP restart is requested to get a fresh lease from a server in a possibly different network. If this happened while using the 'saved lease' the phone lost it's (saved) IP address. Status: checked in to 10.00, 9.00

AC-DSP3: Switch trace off if the DSP Host interface shows an error

Status Closed
Id 80130

SIP: Keep backup registration while calls are active

Status Closed
Id 80137

Terminating backup registration will active calls (through backup system) get disconnected.

SIP: Content of Allow-Events header must be treated case-insensitive

Status Closed
Id 80173

Event names are case insensitive.

IP-DECT/Analog Features: Call Park should be done on the last active call

Status Closed
Id 80209

Szenario is an active call, then a waiting call comes in, which is accepted, then call park is executed. This call park should be done on the accepted waiting call and not the original.

IP-DECT/Analog Features: Call completion

Status Closed
Id 80212

The type of the call is changed back to normal state if the call completion is executed, and facility conversion is added for the call completion state.
This fixes the reusing of features for a call completion callback call, used if IP-DECT/analog features are enabled. This also fixes missed remote hold and retrieve events to the gatekeeper.

SIP: Handling of 488 for encrypted media

Status Closed
Id 80246

If SRTP offer is rejected by remote endpoint with "SIP/2.0 488 Not Acceptable Here"
we should re-try offer without encryption keys.

Phones: Show 'tel' presence on configuration screen

Status Closed
Id 80274

Show latest presence on 'Presence' fkey,
but show 'tel' presence on phone config menu.

PBX-Waiting: Calls to a Waiting queue object in altering state could not be cleared with SOAP

Status Closed
Id 80291

A SOAP UserClear issued for the call monitored at the Waiting Queue object didn't do anything

Phones: Sometime fkey could not be deleted

Status Closed
Id 80389

Sometime fkey could not be deleted when using the fast edit mode (long fkey press).

IP-DECT: Handset display

Status Closed
Id 80402

Names with special characters are not correctly shown in the handsets. This is fixed in the IP1200 now.


IP210 IP230 IP240: Handset receiver volume increased, especially at lower frequencies

Status Closed
Id 80403


For better performance with jinlida receiver.

Gateway: Pass through ctSetup facility

Status Closed
Id 80437

Lync sends diverting party information inside Referred-By header.
Referred-By is interworked to ctSetup facility.
ctSetup facility needs to be passed through by Gateway application.

Linux: IP address with external DHCP server

Status Closed
Id 80471

Assigning a IP address to Linux by a external DHCP server is not working if the network interface which is used is configured with a fix IP address (DHCP disabled). This is fixed now.

IP232: Invalid text on second line of fkey label

Status Closed
Id 80484

Invalid text on second line of fkey label when changing registrations.

PBX: Routing problem from sub-slave to master to object in sub-slave node, but registered to master

Status Closed
Id 80591

Collateral damage from #77874: PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX

SIP: Calls may remain in clearing state

Status Closed
Id 80623

SIP calls may remains undeleted.

SIP: Code optimization

Status Closed
Id 80635

Reduce number object constructions/destructions during message encoding.

PBX: Update of conference id did not always work after multiple transfer accross PBXs

Status Closed
Id 80656

A scenario which did not work was A calls B, B does consultation to C, B Transfers, C does consultation to D, C transfers with B on different PBX then A.
After this the conference id on the call on A should be identical to the conference id on D. This was not the case.

SOAP/TAPI applications which are keeping track of transfered calls could have a problem with this.

AD Replication: LDAP filter encoding failed, when Poll Timer was configured

Status Closed
Id 80658

AD Replication: LDAP filter encoding failed, when Poll Timer was configured

PBX: More consistent use of conferenceId after transfer

Status Closed
Id 80660

The conference ID is used (SOAP/TAPU, CDRs) to associate different call legs to the same call. After a transfer two calls, which have been seperate are connected, so one of the call legs has to change its conference ID, so that the resulting call has a single conference ID again.

There was a complicated logic implemented in the PBX to decide which conference ID should be used, this is now changed to a simple logic: The conference ID of the call on which the transfer is performed, is used.

Example:

A calls B, B does a consuktation to C, and B transfers A to C - This means the transfer is performed on call leg A, so the conference ID of the original call A-B is used for A-C

Gateway: Routing of incoming SIP calls may not work

Status Closed
Id 80709

... because the To-URI is not reduced to a called number (CDPN).
Gateway routing is based on CDPN and does not work for URIs.

Before this fix:
.LOG CALL 2 Alloc
.LOG CALL 2 A:Call -> / GW8::->*::
.LOG CALL 2 B:Call :081604998@212.13.249.90->:018108680@192.168.5.230 / GW8::->GW1::


After this fix:
.LOG CALL 1 Alloc
.LOG CALL 1 A:Call -> / GW8::->*::
.LOG CALL 1 B:Call 081604998->018108680 / GW8:081604998:->GW1:018108680:


SIP: Memory leak when closing signaling interface

Status Closed
Id 80752

Memory leak when closing signaling interface while DNS request is pending on a call.

Phone: Message function key cannot be configured

Status Closed
Id 80766

Configuration is not saved

SIP: Diversion information was wrong on incoming SIP calls

Status Closed
Id 80812

Parsing of History-Info header was wrong, but only if multiple History-Info headers were present.

PBX: Routing problem with nodes/escapes/slaves with calls to extern numbers in other PBX defined nodes

Status Closed
Id 80853

Could happen that the number dialed when sending the call to the extern interface was not adjusted correctly

Build Number format changed

Status Closed
Id 80925

The build number of the hotfixes changes from the 90600.xx format to the 9.061xxx format. This is due to organizational changes without any other significance.

IP-DECT: Memory leak for SIP calls

Status Closed
Id 80937

buffer for received name-id was not freed. Could happen with other facilities as well.

SIP: Transcation handling was wrong

Status Closed
Id 80989

Double delete on a INVITE client transaction object.

0:0027:988:5 - SIP_TAC_INVITE:serial::delete(805b0c80) caller=800440ec

PBX: Adjusting called number, when sending call to extern

Status Closed
Id 81005

depending on the node of the extern object and the called node, the called party number has to be adjusted (escapes added, prefixes added/removed). This did not work unders some conditions.

V9 Hotfix 13 (9061024)

Changes included in Version 9 hotfix13 Definition

New Features

myPBX: Interface for tracing

Status Closed
Id 80973

The web application can now write messages into the trace file of the myPBX launcher.

Oem Code: Accelerating Boot Snmp Traps

Status Closed
Id 81132

Oem Code: Accelerating Boot Snmp Traps

PBX: Handling of call limits at PBX objects improved

Status Closed
Id 81232

A call from a PBX, which is sent back to the same PBX is not counted anymore. This can happen because of node-extern.
Incoming calls at master, which are above the limit are rejected now. They can be rerouted on the slave with "Route Master calls if no Master to"

SIP: New config option for endpoints not refreshing their registration during call

Status Closed
Id 81243

Interoperability:
New config option /keep-active-endpoints for endpoints not refreshing their registration during call.

AVM FRITZ!Box Fon WLAN 7270 v2 (UI) 54.05.21 (Apr 2 2012)

UI hint where to update Linux AP

Status Closed
Id 81295

Changed a hint on the upload/update tab and added a link to the update/upload page on the firmware side.

SIP: New config option /no-certificate-check

Status Closed
Id 81601

New config option to disable validation of remote certificate name
when opening outbound TLS connection.

IP-DECT: Own priority for idle display set by messages

Status Closed
Id 81616

The idle display set by messages has got an own priority now.

SoftwarePhone: Support for Jabra PRO 930

Status Closed
Id 81618

Support for Jabra PRO 930 with product id 0x1016 added.

SIP: New config option "No Inband Disconnect" on GW interfaces

Status Closed
Id 81803

Introduced VOIP_OPTION_NO_INBAND_DISC.
Option was missing on Gateway interfaces.

SIP: New config option /product-id-format

Status Closed
Id 81880

Change the User-Agent string from
User-Agent: (innovaphone IP232/10.00 dvl [90910/90879/501])
into
User-Agent: innovaphoneIP232x90910x501
with /product-id-format 1

phone: ip222,ip232: more USB headsets supported

Status Closed
Id 81972

Sennheiser and some more Jabra Headsets added. Look for "(since V9hotfix13)" in
http://wiki.innovaphone.com/index.php?title=Reference9:Concept_USB_Headset

Debug information on assertion

Status Closed
Id 81973

More debug information on default event handler.

phone: ip222,ip232: support for advanced USB headset functions (redial, reject call, accept waiting call)

Status Closed
Id 81975

A lot of USB headsets generate special indications to request redialing of last number dialled, to reject a ringing call, to accept a waiting call and to put the active call on hold or to switch between an active and an held call.



Bug Fixes

IP222 equalizer update

Status Closed
Id 80004

- handsfree speaker equalizer enabled
- handset mic and receiver equalizer smoothed
- ADC gain reduced, input gain increased ( after ec ) to avoid clipping
- halfduplex mode disabled

IP222 IP232 IP241: repeated ethernet link status 1000M wrong

Status Closed
Id 80029

.


IP222 IP232 : Default LCD backlight standby brightness reduced

Status Closed
Id 80031


IP241 handsfree equalizer update, IP222 handsfree mic equalizer update

Status Closed
Id 80157

- IP241 handsfree speaker equalizer enabled
- IP241 handsfree micro equalizer enabled
- IP222 handsfree micro equalizer enabled
- All three filter calculated with measured frequency response
- IP241 ADC gain reduced, input gain increased ( after ec ) to avoid clipping


phone: ip222, ip232: Pressing Talk button on USB Headset when there is an active and a held call does not transfer

Status Closed
Id 80413

Instead of transfer the active call is disconnected and the held call is retrieved Status: checked in to 10.00

IP241,IP222,IP232: Show name of diverting party in incoming calls

Status Closed
Id 80839

Show name of diverting party in incoming calls (not only number).

PBX-SOAP: UserRedirect - original called number got lost

Status Closed
Id 80854

For example if an application used an Waiting Queue object to monitor for incoming calls and redirected these calls to agents. The agent receiving the call could not see if the call was diverted to the waiting queue already.

SIP: Send 200/OK for MESSAGE(text/plain) when accepted by application

Status Closed
Id 81017

Application gives "NormalCallClearing" to SIP stack.
SIP stack better sends "200 OK" instead of "603 Decline".

SIP: Un-escape content of XML elements

Status Closed
Id 81019

XML element content requires some resevered characters to be escaped (<>).
These escape sequences (&gt; or &lt;) must be un-escaped onthe receiving end.

IP241,IP222,IP232: Two status symbols may overlay each other

Status Closed
Id 81027

Symbol for "Call diversion" and symbol for "PIN-locked" overlay each other.
Hide "Call diversion" as long as phone is PIN-locked.

SIP: Handling of multiple 401/407 responses

Status Closed
Id 81045

Implemented handling of up to 3 401/407 responses per transaction.

SIP: Trap while releasing call

Status Closed
Id 81089

Timer fires during call release and causes trap.

H.323: Memory leak when sending special OEM H.235 key elements

Status Closed
Id 81160

Only happens if Avaya SRTP is enabled in OEM products

H.323: Potential Trap in case of high load and media renegotiation/call clearing collision

Status Closed
Id 81173

There is a very unlikely situation when media-renegotiation is started and then the call is cleared, which could cause a message related to the media-renegotiation to be sent to a already deleted call object. High load could make this situation more likely.

IP241 handset equalizer

Status Closed
Id 81174

Enable IP241 handset micro and speaker equalizer.
Same values as on IP222.
Reduces noise in G711 mode.

IP1060/3010/6010/22/24/28/302/305: sequence number to DSP changed from byte to word, caused jiter buffer warnings

Status Closed
Id 81189

IP222/232/241/1060/3010/6010/22/24/28/302/305: min jitter buffer changed to 10ms

Status Closed
Id 81197

was 35ms, now 10ms as on ip6000/800

IP-DECT: Duplicate call setup

Status Closed
Id 81213

The IP-DECT Master sends in some circumstances a call twice to the same radio in the same time. This affects only the IP1202 and OEM devices, not the IP1200, and is fixed now.

IP-DECT: OEM configuration option

Status Closed
Id 81214

The configuration format of an OEM configuration option was changed for the backward compatibility.

phone_orchid: micro mute when a waiting call was connected after an outbound call setup had been cancelled

Status Closed
Id 81229

- A goes offhook
- B calls A, a waiting call from B is indicated on A
- A goes onhook, phone rings
- A goes offhook again and is connected to B
- A hears B, B doesn't hear A

SIP: Pass display information to application

Status Closed
Id 81230

Pass display information to application when registration comes up.

IP0010 IP1060 IP3010 IP6010: During long reset the ready LED was blinking orange and the Ethernet LEDs were swaped

Status Closed
Id 81236

Now the ready LED shows the green blinking during long reset.
The Ethernet LEDs are initialized directly after reset to overwrite the default setting that swaps link and speed.


X.509: Creating certificate containing IPv4 address did not work

Status Closed
Id 81251

When creating a certificate or a request with an IPv4 address, the IPv4 address was mapped to an IPv6 address. This was not the expected behaviour.

172.16.10.32 -> ::ffff:172.16.10.32

Now both IPv4 and IPv6 addresses can be used.

ip222,ip232: waiting calls mute after having been accepted/connected on USB headsets, mainly seen with Jabra LINK 14201-30

Status Closed
Id 81312

In some cases only the accepted call was mute and the next call was OK again but the Jabra LINK 14201-30 lost the USB connection in most cases.
Delaying the HID-commands sent to the headset solves this problem.


PBX: Objects list filter for numbers did not work correctly anymore

Status Closed
Id 81376

only top level nodes were displayed
Collateral damage of fix

79982: PBX: When editing a Node object it was changed to a PBX object

SIP: Pass display information to application

Status Closed
Id 81379

Pass display information received in REGISTER response to app.

IP241,IP222,IP232: Show H323-ID instead of Display Name if Display Name is not available

Status Closed
Id 81419

Like on old b/w phones.

IP-DECT: Memory leak with special Cisco features

Status Closed
Id 81463

Only with special OEM features.

SIP: Problems with DNS resolving of proxy adresses

Status Closed
Id 81522

Not always re-tried when failed at startup.

IP241,IP222,IP232: Rendering errors on Fkey configuration screen

Status Closed
Id 81556

When scrolling or leaving Fkey configuration screen.
List (Toggle) controls were not rendered correctly.

IP-DECT: Mobility Master registration

Status Closed
Id 81571

The Mobility Master does not accept clients with the OEM name IP1202. This is fixed now.

HTTP: Possible trap on many simultaneous sessions

Status Closed
Id 81597

Sorting the TCP sessions did not work correctly, which caused an assertion because it could happen that it was not possible to remove a session

IP-DECT: System GUI, disabled local coder options

Status Closed
Id 81603

If the PARI function (only IP1202) of the IP-DECT Master is disable, configuration changes on the System GUI do not effect anything. The settings for the local coder are disabled on this GUI page now.

IP-DECT: System settings not to dynamically connected radios

Status Closed
Id 81612

System settings should not be sent if the radio-master connection is a dynamic one (IP1202 only). This is fixed now.

Support for old versions of Jabra BIZ 2400 headset with 48 kHz output sampling frequency

Status Closed
Id 81613

the newer versions come with the usual 16 kHz sampling frequency but have the same signature (vendor/product) as the older ones.

SoftwarePhone: Trap with trace and mute ringer event

Status Closed
Id 81619

A trap occurs if the trace is enabled and the ringer mute option is toggled.

IP222 IP232 IP241: Force same speed of the switch ports for 1000M/100M scenarios

Status Closed
Id 81693

In scenarios that operate one Ethernet port with 1000M and the other with 100M the switch througput was low.
Now the 1000M port is reconfigured to 100M, and the throughput is high.

SIP: Support for multiple audio media descriptions

Status Closed
Id 81712

One without encryption and one with encryption:

v=0
o=OpenStage-Line_0 968610650 446118927 IN IP4 10.30.1.144
s=SIP Call
c=IN IP4 10.30.1.144
t=0 0
m=audio 5012 RTP/AVP 9 8 0 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv
m=audio 5010 RTP/SAVP 9 8 0 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ujVU8G6kgknZnPflRwx8tadNskkp9glas/DFCbC3
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:+uoFBNVVhDY5OOGvMOdAlvpxvc98hX/VeProhlwH
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv


IP232: Show more of collapsed fkeys

Status Closed
Id 81807

140px when expanded.
100px when collapsed (80px before).

SIP: Double Replaces header in INVITE after REFER

Status Closed
Id 81847

Reject by "Swyx IpPbxSrv/8.1.0.246" with "400 Bad Request(Multiple values in single-value header Replaces)"

PBX Mobility: No response for call to busy mobile phone, if no fixed phone

Status Closed
Id 81850

If a user is called with mobility configured and no fixed phone and the mobile phone was busy, then the call did not complete and was hanging as if the number was not complete. The call should be answer with busy instead.

Chrome compatibility issue on PRI statistics page

Status Closed
Id 81859

If clear was clicked an error message was displayed

PBX Admin UI: Not possible to change DECT parameters for users with cf/grp admin rights

Status Closed
Id 81871

The user interface prohibited the editing of DECT parameters based on the rights of the object to be edited not based on the login

phone: ip222,ip232: when a call is released from remote while another call is waiting the waiting call cannot be accepted.

Status Closed
Id 81907

When the call is released from remote a buys tone is generated for two seconds. Therafter phone rings to indicate that the waiting call can be accepted now.
When trying to accept this call by pressing the headset talk button the call was disconnected instead.

SIP: Memory leak in SIP stack

Status Closed
Id 82058

Memory leak in SIP stack.

V9 Hotfix 14 (9061046)

Changes included in Version 9 hotfix14 Definition

New Features

phone: support PBX-directory access via TLS

Status Closed
Id 78275

A "Use TLS" checkmark has been added to the PBX directory config of the phone. If set, port 636 is used instead of port 389.

PBX-Mobility: Data Call Thru

Status Closed
Id 81513

To speed up dialing with the mobility client, the called number can be posted to the PBX via HTTP before the call

IP-DECT: DECT security (2)

Status Closed
Id 82191

Some changes in preparation for the new feature "DECT security". This feature will be available later (IP1202).

IP-DECT: No fall-back after unattended call transfer

Status Closed
Id 82198

Now there is no fall-back after an unattended call transfer and the behavior is consistent with the other call transfer types (attended, semi-attended). To switch back to the hold call the R-key must be pressed.

IP-DECT: Idle display update

Status Closed
Id 82199

Handling of idle display update message in the call release message is added. Used with an OEM PBX.


phone: ip222,ip232: the variable KEYS0/HID-MAP permits to map new USB headsets to builtin descriptors

Status Closed
Id 82635

Sometimes USB headsets come with a signature different from the signature of similar headsets which are already supported.
A "vars create KEYS0/HID-MAP p <map>" maps the new signature to an existing one.
<map> format is
manufacturer:product=manufacturer:product
the second manufacturer:product tuple is the signature of an already supported headset,
'manufacturer' and 'product' are plain 4 digit hex numbers without a "0x" prefix.

IP2x2, IP241: Coder Preferences for prefered coder G.722 suboptimal

Status Closed
Id 82815

In case G.7222 was selected as prefered coder and the called endpoint did not support G.722, as next best coder G.729 was selected. This is typically not what is desired in such a case, G.711 is the better alternative in this case

myPBX: Show version of launcher in the list of sessions

Status Closed
Id 82821

The version of the used launcher is now shown on the page PBX/myPBX.
Status: checked-in: 10.00
checked-in: 9.00

H.323: Support for Avaya SRTP with AES128/80

Status Closed
Id 82829

Needed for OEM

SoftwarePhone: HID Support for new headsets

Status Closed
Id 83184

With this new version the following headsets are supported for call control:
- Jabra GO 6430 (Jabra LINK 350 USB with firmware 5.4.17 or later) with product id 0xa342. Please select the first device.
- Jabra SUPREME UC (Jabra LINK 360 USB) with product id 0xa346. Please select the first device.
- Jabra PRO 9470 with product id 0x1042.
- Sennheiser VoIP USB headset (SH 350 IP) with product id 0x0008.
- Sennheiser DW Office with product id 0x740a. Please select the first device.
- Sennheiser CEHS-CI 02 (USB adapter cable) with product id 0x0030. Please select the second device.

PBX: Description was missing for DECT System object

Status Closed
Id 83198

A configurable description is useful for the DECT System object as for all other objects

Permit logging to a second (shadow) log server

Status Closed
Id 83206

"Services/Logging/Log Server/Log Server Shadow/Address" defines the adress of a second server.
"Services/Logging/Log Server/Log Server Shadow/Enable" starts/stops logging to the second server.
Except the address the configuration for the second server is copied from the first server.


Bug Fixes

Incorrect disk usage calculation for more than 4GB

Status Closed
Id 81209

The disk usage calculation was wrong, if more than ~4 GB of the card were used.

PBX Waiting: Call state shows "Disconnecting" after switch from announcement 1 to announcement 2

Status Closed
Id 81370

Announcement disconnect changed state

PBX: Switch from Music on Hold to inband ringback on Alert after unpark oder transfer

Status Closed
Id 81407

To give the caller feedback that soon somebody may answer the call

PBX: Don't forward in-band info indicator if no media channel

Status Closed
Id 81879

Not good to indicate in-band info available if no media channel can be negotiated

SIP: Trap when configuring user presence

Status Closed
Id 81996

Trap may occur when configuring user presence.

HTTP-Server: Configuration of "Public compact flash access" did not work for all cases

Status Closed
Id 82064

E.g. /DRIVE/CF0/Neuer Ordner/ does not work, because HTTP request contains escaped sequences.

phone: ip222,232.241: no notification tone on a successfull redial attempt

Status Closed
Id 82166

On a failing or unanswered call the menu key opens the "Recall" menu. If "Redial" is selected the call is automatically redialed for 20 minutes in intervals depending on the result of the previous attempt. On success the user should be notified about the connection.


IP-DECT: Trap during subscribing handsets

Status Closed
Id 82190

A trap occurs during subscribing handsets on the IP1200. This is a fix for the previous fix #80424 and fixed again.

IP-DECT: Wrong release code

Status Closed
Id 82195

The IP-DECT sends the wrong release code "User not reachable" if the handset was not connected and the remote parts disconnects. This is fixed now.

IP-DECT: No delay with semi-attended call transfer and SIP

Status Closed
Id 82197

If the SIP protocol is used and the user do a semi-attended call transfer, the call transfer is directly confirmed again. The semi-attended call transfer is stored in the base station and executed as an attended call transfer if the target party connects.

IP-DECT: Display info with failed user log-in

Status Closed
Id 82200

If the user log-in fails, the cause is shown in the display.

IP-DECT: Short tone info by PBX only one time

Status Closed
Id 82204

Tone information with defined length and requested by the gatekeeper is played only one time to the handset. This is used by an OEM PBX and fixed now.

IP-DECT: Hanging aborted semi-attended call transfers with SIP

Status Closed
Id 82205

If the SIP protocol is used, a semi-attended call transfer is done by the user and the call transfer can not be executed, the remaining call party is not disconnected. This is added now.

IP-DECT: LDAP server GUI description

Status Closed
Id 82258

The LDAP server GUI description is wrong and corrected now. (The LDAP server can not be a HTTP server.)

DHCP: Name registration at WINS was not refreshed as requested by TTL in registration response

Status Closed
Id 82289

If the DHCP-client gets a lease containing a WINS-server address and a NETBIOS node type P or M (1 or 2) the client tries to register it's NETBIOS-name (ipxxx-xx-xx-xx) with the WINS-server. The TTL returned by the server in the registration response determines when a name refresh has to be sent.

Phones: Presence info during ringing state may show garbage data

Status Closed
Id 82306

In case a presence update arrives at the phone while phone is in ringback state.
Have been observed in conjunction with call forking with mobility only.


myPBX: Support contact names containing a single quote

Status Closed
Id 82323

Configuring contacts with a H.323 id containing a single quote (') caused script errors in the web application.

PBX: RTP-DTMF was disabled by Voice Mail object during re-negotiation

Status Closed
Id 82332

RTP-DTMF acc. to RFC-2833 was disabled by Voice Mail object during re-negotiation.

Gateway CDR with '0. 0' charge amount

Status Closed
Id 82359

Should be '0.00' instead

phone: call completion did not recover when DND(busy) was set at the phone requesting the call completion

Status Closed
Id 82390

when DND(busy) was set on the phone reqesting the call completion and was cleared some time later a "Recall possible" was not indicated anymore although a pending call completion was indicated on the called phone.


ipv6: memory leak when sending fragmented packets

Status Closed
Id 82394

H.323:No Media for calls with reverse media to a H.323/SIP exclusive Code Media Relay interface

Status Closed
Id 82408

The execlusive coder/media relay config is used to avoid media negotiation problems with carrier which do not support media renegotiations. In case of a call with reverse media to such an interface, this did not work. This happens for example if a CFNR is configured at a Waiting Queue which redirects a call, which received an announcement from the Queue to such interface.

Kerberos: Allow editing multiple fields in admin UI

Status Closed
Id 82425

This is a fix for the page General/Kerberos in the admin UI.

Editing more that one item at a time could result in strange effects. Now it is possible to edit multiple fields.

phone_inca: when a call completion was set up with CLIR active the called party was not displayed on a possible recall

Status Closed
Id 82435

A sets CLIR, A calls B, B is busy
A sets a CCBS request via Menu/Recall
B goes on Hook
A rings and sees 'anonymous' instad of the number of 'B', status line is empty (should show "Recall possible")


IP-DECT: Display info with remote control call

Status Closed
Id 82444

The display info shown after accepting a remote control call is fixed now.

IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: enable modembypass

Status Closed
Id 82458

Modembypass is enabled on all calls with disabled T.38 and coders G711A or G711U.
Switch to modem bypass is indicated in the trace by "switch to modembypass".
The feature can be disabled with http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl

Modembypass works best if T38 is disabled on both sides.
If T38 is enabled on the called side the CED may trigger a T38 session, this changed back to voice and modem bypass is enabled (if G711 is active). The first modem tone is interupted, but we still have modembypass on both sides.
If T38 is enabled on the calling side the calling side stays on regular G711.




IP800 IP6000: reduce probability of false DTMF detection

Status Closed
Id 82471

Change DTMF detector signal to noise ratio from 12db to 18db.


Gateway: 'Enable PCM' option added at the CONF interface of the IP800

Status Closed
Id 82481

The option 'Enable PCM' is available at the CONF interface of the IP800 now.

Gateway: PCM mode of the CONF interface fixed

Status Closed
Id 82482

The PCM mode of the CONF interface is not activated for a call even though it is configured and possible. This is fixed now.

IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: T38 trace flag added

Status Closed
Id 82484

Allows to trace T38 connection on the PCM port and on the DSP host interface.
Use this if fax modem problems are suspected.
Enable at http://addr/debug.xml at trace->T38 trace.

Gateway: Routing problem with blockdial route and following matching non-blockdial incomplete routes

Status Closed
Id 82486

Example:

Blockdial Route 00->
After this a non-blockdial route with 0...

If now a number of 001 was dialed, the first route should match and after the enbock dialout the call should be sent to the destination of the route. Instead the call was rejected with "no destination found"

failure of analog ports of ip28

Status Closed
Id 82488

ip28 analogue ports do not react to incoming calls and hook-off. Problem could only be solved by reset.

Debug "HTTP_GET LOG_HTTP.1: retry, authentication failed" removed

Status Closed
Id 82499

Phone: Display text received with BYE

Status Closed
Id 82525

Pass display text to phoneapp.

SIP: Trap during call handling

Status Closed
Id 82544

Trap during call handling

IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: update to DSP code 660.27

Status Closed
Id 82551

660.27 is the released version on the audiocodes web.
Fax relay and modem bypass improved.

IP222 IP232 IP241: update to DSP code 660.27

Status Closed
Id 82554

660.27 is the released version on the audiocodes web.
Improved acoustic echocanceller.

phone: call forwarding options offered even when call forwarding was not possible

Status Closed
Id 82567

Call forwarding is not supported when running SIP. But when the menu key was pressed after entering a number call forwarding options were offered (happened with the primary registration only).

PBX Mobility: Trap in case of Transfer of a call from a mobile endpoint to another mobile endpoint

Status Closed
Id 82584

The Trap happens in the following call scenarion
- Mobile endpoint calls in, using mobility two-stage dialing
- call is accepted at local phone
- on local phone a consultation call is initiated to another user with mobility
- when mobile phone rings, a transfer is initiated on local phone
- the called mobile phone accepts the call
- the trap happens when the called mobile phone hangs up

There could be other call scenarions where the trap happens as well

SIP: SRTP key exchange failed

Status Closed
Id 82616

Bug in base64 decoding of SRTP key.

PBX Trunk: Number to Name Feature did not work for calls to busy endpoints

Status Closed
Id 82619

Such a call was not rejected with cause 'User Busy', but was just hanging.

PBX Mobility: Call to mobile phone was sent with invalid diverting information

Status Closed
Id 82622

A call to a mobile phone is sent with a diverting leg2 info, which means, the call contains the information, that it was diverted by the called user to the mobile phone. So in theory this could be displayed on a mobile phone.

The coding of this information was wrong and created interop problems with some networks.

SIP: Trap on subscription handling

Status Closed
Id 82623

Trap on RAS_REGISTRATION_VERIFY between sending SUBSCRIBE and receiving 200/OK.

SIP: Removed cisco-special retrieve signaling

Status Closed
Id 82637

Removed cisco-special retrieve signaling.

phone: ip222,ip232: Plantronics APU70 - Savi 7xx - Radio Link not cleared on release from remote

Status Closed
Id 82654

On a release from remote for a call set up by pressing the Talk button (headset or base) the Radio Link between base and headset was not cleared until the Talk button was pressed again.

SIP: Mobility did not work due to RTP-DTMF

Status Closed
Id 82674

DTMF must be passed through signaling channel to get mobility working.
Suppress RTP-DTMF capability in SDP answer also.

IP3010 IP810: number of DSP channels and number of conference channels was wrong.

Status Closed
Id 82675

Now the IP810 shows 20 DSP channels and 30 conference channels.
Now the IP3010 shows 42 DSP channels and 60 conference channels.

SIP: Restart NAT discovery if failed

Status Closed
Id 82676

Restart NAT discovery if failed

IP222 IP232 IP241: Reduce DTMF level

Status Closed
Id 82685

During DTMF receive and transmit levels similar as on IP240 are used

-10db level
0xc0=208--> 22db attenuation
also insgesamt ein level von -32db ( bei Vollauschlag ) oder -29dbm.
Der alte Wert beim ac_phone3.cpp war -9db

Weitere Diskussion:
Es gibt den Fall das inband DTMF zum IP Netz geschickt wird, da gabs in Fall 59846 die Änderung zum IP mit LEV=0x28 -->-10db und Attenuation 0xff-->18db, also mit -28db zu senden.
Da das gut funktioniert und die beiden Pegel nicht so unterschiedlich sind unde der ac_dsp3 nicht unterschiedliche Pegel zum IP und zum Codec kann nehmen wir die -28db=-25dbm.


Laut www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-Q.24-198811-I!!PDF-E&type=item Tabelle A-1 sind die -25dbm am unteren Ende, da ist vom Wert A
-22..-30 je nach Administration die Rede. Wahrscheinlich ist das kein Problem,
bei Audiocodes (ac_dsp2) kann man die sensitivity von -28 bis -38db einstellen (DTMF_DETECTION_ENERGY_THRESHOLD__28dBm)


-->

IP222,IP232: Going offhook in call list always dials first list entry (not touched list entry)

Status Closed
Id 82732

Going offhook in call list always dials first list entry (not touched list entry)

Memory leak after Firmware or Bootcode download

Status Closed
Id 82740

After a Firmware or bootcode download via Maintenance/Download the associated command processor instance was not released.

IP-DECT: Call transfer compatibility with SIP

Status Closed
Id 82742

For compatibility reasons with the SIP protocol the call transfer initiate result message should not be sent until the connect message is received. This is changed now again.

SIP: Trap if multiple calls arrive at same time

Status Closed
Id 82743

Trap if multiple calls arrive at same time.

SIP: Bug in digest authorization

Status Closed
Id 82761

Sometimes wrong method is used in digest calculation.

phone: ip222,ip232: USB headset mute after activation of changes in codec parameters, headset icon cleared on status line

Status Closed
Id 82823

After for example
config add AC-DSP0 HEADSET /InputGain 32
config activate
the headset icon was cleared on status line and the headset was mute although the headset Talk key was handled.



PBX Waiting: Evaluate Busy on ... Calls for calls to an operator

Status Closed
Id 82858

If a operator has configured Busy on 1 call, there should be no call from the Waiting Queue if the operator is already busy.

PBX: Avoid signaling loop after call transfer

Status Closed
Id 82865

A signaling loop could be created by calling from a phone registered at one PBX to a phone at another PBX, then putting the call on hold and do the same call again, accept on the other side the waiting call. If both parties do then a transfer there is the signaling loop.

Such loop ist now detected and the call is cleared.

PBX Broadcast: No diverting name sent with broadcasted call

Status Closed
Id 82880

The diverting and original-called name info was missing from the diverting leg2 info generated by the broadcast object

Web-UI: Misplaced reset-required indication

Status Closed
Id 82896

Misplaced reset-required indication on ISDN interface config (TEL1,TEL2,...).

IP0010 IP1060 IP3010 IP6010 IP810 IP302 IP305: ISDN: Enable fax detection only after connect - fixed

Status Closed
Id 82900

Fax detection was not enabled in all cases.


Happens in test/10.00/box/dsp/ip6010 with
fix: #78316: SIP/H.323: Don't complete media negotiation if no media can be seen

IP1060: Memory size is not correctly shown

Status Closed
Id 82906

The memory size is not correctly shown in the IP1060 in diagnostics counter page.

phone: a calling party name found by inverse directory lookup for an external call was sometimes not stored in call list

Status Closed
Id 82919

When the external call setup came in with a name identification provided by the external source and there was another name found by inverse directory lookup the name from directory was displayed on the call screen but the name identificication was stored in the call list. Now the name found by inverse directory lookup will be stored.

PBX Routing: Node extern did not work for calls from a trunk marked as local object

Status Closed
Id 82948

The call was not routed back to the originating slave

IP222 IP232 IP241: New equalizer and volume setting

Status Closed
Id 82952

- use equalizer up to approx 6db
- use digital volume to adjust volume


SIP: Trap on out of memory

Status Closed
Id 83004

SIP-Client allocations not deleted.
Outbound control calls without facility interworking.

PBX Routing: A CFNR at a PBX object for WAN re-routing did not work if it contained escapes

Status Closed
Id 83017

The number configured at the PBX object is interpreted in the context of the node of this PBX object. If escapes were needed to dial the WAN trunk, it did not work.

External-UC: Presence info assigned to wrong PBX object

Status Closed
Id 83020

When presence eventlist is received from External-UC
all presence info was assigned to same PBX object.

PBX: Editing Config Templates impossible, if by some old firmware a strange config ended up in a User object

Status Closed
Id 83033

The problem was a empty hardware id (hw=""). This is now ignored.

PBX mobility: The forking destination put in for mobility at a user should be dialed from the node of the user

Status Closed
Id 83040

This number was dialed from the node of the Mobility object. This was confusing, because this number was configured at the user and it was also different behaviour as with forking without mobility

PBX: Standyby PBX generated alarms for missing slave registrations, even if active PBX up

Status Closed
Id 83089

There should be only an alarm, if the standby PBX is active

SIP: One-way audio after mutual hold on dect systems

Status Closed
Id 83126

Dect ep gets FTY_HOLD_NOTIFY but no FTY_RETRIEVE_NOTIFY.

X.509: Avoid alarms on missing system time after reboot

Status Closed
Id 83137

Allow 60 seconds for setting the system time before an alarm is set. Also certificates are now rejected silently, in that time.

SIP: Problems with presence signaling on External-UC link

Status Closed
Id 83177

Missing parameters on Contact-URI in 200/OK for SUBSCRIBE(presence):
maddr
transport

Wrong Contact-URI in presence XML in PUBLISH.


IP-DECT: Support for more than 32 LDAP attributes

Status Closed
Id 83191

The IP-DECT devices support up to 256 LDAP attributes now.

PBX: Support for more than 64 LDAP attributes

Status Closed
Id 83192

The PBX supports up to 256 LDAP attributes now.

IP-DECT: LDAP replication alarm loop

Status Closed
Id 83249

If the LDAP replication is configured, but it can not be synchronized, there is an alarm loop. This is fixed now.

Gateway: On IP24, IP302, IP305 it could happen that multiple routes disappeared if a route was deleted

Status Closed
Id 83278

Hard to predict when this happened.

Gateway: No Media received event was generated for T.38 calls, which started without actual RTP Traffic

Status Closed
Id 83433

This happend for calls to/from Fax servers, which switch to T.38 without sending any RTP packets first.

SIP: Multiple subscriptions for 'message-summary'

Status Closed
Id 83469

After every RAS_START another subscription for 'message-summary' is established.

V9 Hotfix 15 (9061078)

Changes included in Version 9 hotfix15 Definition

New Features

New remote control codes to be used for phone tests by soap applications

Status Closed
Id 83468

The new UserRc codes are executed only when the addressed phone is either in handset, headset or handsfree mode, i.e when calling, connected or disconnected but not when alerting:
16 - change to handset mode
17 - change to headset mode
18 - change to handsfree mode
19 - monitor mode on (add speaker to handset or headset mode)
20 - monitor mode off (back to plain handset or headset mode)

Alarm/Event handling: Authentication for received remote Alarms/Events

Status Closed
Id 83603

Allow by configuration to only accept authenticated alarms or events

PBX: Preparations for objects visible only if appropriate license installed

Status Closed
Id 83615

for v10

PBX: New Feature to allow registration with password to devices regardless of address filter

Status Closed
Id 83794

It is now possible to configure a flag at a device to allow a registration for this device even if there is an IP Filter which does not match. This is useful if registrations from the public internet to the PBX shall be possible. Without this feature this could be opened only for the complete PBX. Now it can be restricted to a few devices.

SoftwarePhone: Signature added

Status Closed
Id 83915

The SoftwarePhone installer and the install package is signed now.

IP-DECT: Phone book 'Use TLS' option

Status Closed
Id 84001

'Use TLS' option added for the central phone book search. This changes the standard port from 389 to 636 if no port is configured. The central phone book search is only available with the IP1202.


simple static logging interface

Status Closed
Id 84204

to simplify sending of log messages, alarms and errors a simple static interface to the logging module was added. log_if::log(class serial src, const class event & event) passes the given event to the primary logging module (aka LOG0). This works also with 'src' = 0.

myPBX: Pass selected device to launcher

Status Closed
Id 84556

Needed for version 10 remote video

SIP: Announcing "a=T38MaxBitRate:14400" in T.38 offer

Status Closed
Id 84770

Announcing "a=T38MaxBitRate:14400" in SDP offer for T.38
since all current devices support that modem speed.

SIP: Set numbering plan to "ISDN/telephony" in case of "Bellcore-dr2" as Alert-Info in INVITE

Status Closed
Id 84939

Set numbering plan to "ISDN/telephony" in case of "Bellcore-dr2" as Alert-Info in INVITE.

IP-DECT: Phone book error events

Status Closed
Id 85001

The phone book module sends an error event now if the LDAP directory search fails.

Bug Fixes

IP-DECT: Trap during subscribing handsets

Status Closed
Id 83690

A trap occurs during subscribing handsets on the IP1200. This is a fix for the previous fix #80424 and fixed again.

Trap identification:
XCPT: no 2 (TLB load) pc 942e23d8 ra 942e23cc va 00000000

PBX Waiting: Trap on leak-check if dtmf maps are configured

Status Closed
Id 83691

Only happened if debug.xml leak check was used

IP222 IP232 IP241: Adjust gains for better echo canceller performance

Status Closed
Id 83703

..

myPBX: Remove unimportant notifications

Status Closed
Id 83707

The following events are not so important that the user has to be notified using a windows bubble notification.
- Visibility requests
- Missed calls

IP222 IP232 IP241: Enable noise reduction

Status Closed
Id 83715

Enable noise reduction on all handset/handsfree/DHSG headset micro.
Use DSP code 660.27.pa.03

Noise reduction parameter can be tuned at
http://addr/AC-DSP0/mod_cmd.xml?xsl=phone-dsp.xsl

Voicemail: Prevent <prompt>, <record> on incoming control calls

Status Closed
Id 83826

Turned out to cause idle-reset requests never being processed.

An administrative solution also exists: Watch out for the URL variable "$_noctl" in this article:
http://wiki.innovaphone.com/index.php?title=Howto:Configure_the_innovaphone_Voicemail#URL_Query_String_Variables

HTTPCLIENT: Allow configuration of a http authenticated URL even if the server needs no authentication

Status Closed
Id 83900

If a password was configured and the server didnt need it, a HTTP put created a file of zero length.



PBX Waiting: Trap on collision of operator connect and two-stage dialing

Status Closed
Id 83926

If two-stage dialing (Maps) is used to call a Trunk or Gateway object, the call is sent after a blockdial timeout. If an operator connected the call before this timeout happened, a trap occured.

For this to happen DTMF maps and operators have to be configured on the same Waiting Queue object, with is kind of unusual

PBX: Call to a Trunk/Gateway was not marked correctly as external, if no connected number was received

Status Closed
Id 83940

Calls from a Trunk must be explicitly marked as internal with respective connected number, otherwise they should be treated as external

HTTP: Chunked transfer fails if the last 2 bytes of the chunk header are in the next tcp packet

Status Closed
Id 83986

Seen with the application platform as broken pipe

PBX: Registration with Name/Number did not work correctly if default device not first

Status Closed
Id 83989

For a registration with name or number, the information if the PBX password shall be used was always taken from the first device regardless if this was the default device (hw-id identical to name) or not

PBX: Partial Rerouting was prohibited in Alerting State (CFNR)

Status Closed
Id 83993

This was done under the assumption the partial rerouting is not supported as CFNR by public networks.

PBX Waiting: Call to operator with Twin Phone Checkmark did not work anymore

Status Closed
Id 84084

Collateral damage of

fix: #82858: PBX Waiting: Evaluate Busy on ... Calls for calls to an operator

SoftwarePhone: Product string

Status Closed
Id 84119

The product string is changed, used e.g with the PBX registration.

phone: provide complete dialog info to a phoneapp

Status Closed
Id 84192

Both group indications and dialog infos are signaled via a group indication facility. For dialog infos the parked_to_alerting member was overloaded to provide the info as expected by the existing phoneapp.
Now the parked_to_alerting member is passed to a phoneapp as received.

IP222 IP232 IP241: Force same speed of the switch ports for 1000M/100M scenarios (configuration option added)

Status Closed
Id 84200

In scenarios with frequent transistions of the attached PC to sleep renegotiating the link speed may be undesired. For this case the force same speed mechanism can be disabled.

Other changes:
1000M is only changed to 100M if the other port runs at 100M. The previous version changed from 1000M to 100M if the other port runs at 100M or 10M.

The statistics can be collected from the PC port or from the LAN prot or from both.

Packet forwarding on the PC port is disabled if the port is down to avoid misleading collision counter behaviour.



CDR fixes for external call detection

Status Closed
Id 84211

The type="ext" attribute was not set reliably. Additionaly an attribute pseudo was added to the <user/> tag to indicate the type of object the CDRis created for.

http client : authentication was not retried after a failure when the offending request was repeated in the same session

Status Closed
Id 84217

When a httpclient user repeated a failing request in the same session the authentication was not tried again. Thus a change of the client side URL password or a change of the server side password had no effect until a new session was started.

IP6000: Prevent blinking error LED on old IP6000 with HW-Build <110

Status Closed
Id 84227

Conference DSP driver was started on old hardware that doesnt support the conference DSP

IP222 IP232 IP241: pressing speaker key when phone is in handset mode switches to handsfree mode instead to toggle monitor mode

Status Closed
Id 84297

handset/headset plus speaker is not supported

myPBX: Inconsistent display of group monitoring rights

Status Closed
Id 84350

In the visibility settings groups were not displayed if the membership is "dynamic out". But members of that group still have monitoring rights.

phone: when scrolling directory search results sometimes one of the numbers of a contact was not displayed

Status Closed
Id 84362

the tag characters assigned to the different numbers were not included in sort order.

phone: status messages for outbound external calls were sometimes garbled on display

Status Closed
Id 84365

happened when for a preceeding outbound internal call in alerting state presence info was displayed

IP0010: DSP didnt start with build 9061044

Status Closed
Id 84388

Page Ldap/Replicator/Status didn't display in WebKit

Status Closed
Id 84400

Page Ldap/Replicator/Status didn't display in WebKit

phone_orchid: displaying both dialed and connected number may be misleading, it's better to omit dialed number

Status Closed
Id 84422

when for example 022222222 was dialed and the network reported a connected number 03022222222 the display info "022222222 -> 03022222222" looked like a transfer.


Hide LDAP Server Password For Viewer Accounts

Status Closed
Id 84557

Was accessible for viewers

phone: enable directory search function key also in connected state

Status Closed
Id 84559

it's sometimes useful to browse the directories for a number while talking.

myPBX: Only send one command at a time

Status Closed
Id 84568

Queue commands instead of sending overlapping commands. This limits the number of open AJAX connections to two.

H.323: Accidential fallback to slowstart if faststart response received in PROGRESS

Status Closed
Id 84601

Only happened in some H.323/SIP interop scenarios

IP222 IP232 IP241: Updated gain and equalizer setting

Status Closed
Id 84605

Tuned for high MOS values according to ETSI ES202737 ES202738 ES202739 ES202740

IP0010 IP1060 IP3010 IP6010 IP22 IP24 IP28 IP302 IP305 IP800 IP6000: Minifirmware not shown on LED

Status Closed
Id 84616

Minifirmware should blink long green short red. This happened only after a firmware/bootcode update.

phone: ip222, ip232: status stage added to all USB control transfers

Status Closed
Id 84617

the status stage is mandatory as well for IN as for OUT control tranfers

H.323: More information on "Unexpected Message" event

Status Closed
Id 84699

The message type and the state for which this message was unexpected is needed to find out what the problem is.

SIP: Trap during channel handling

Status Closed
Id 84800

Rare trap when re-assigning channels.

IP232,IP222,IP241: Rendering errors when trying to use backround image with indexed colors

Status Closed
Id 84814

PNG mode "indexed colors" is not supported.
Trying to use an indexed color png leaves phone screen in bad shape.

NAT: Don't forward DNS requests from public network

Status Closed
Id 84842

As kind of denial of service attack, bursts of incoming DNS requests were seen. The nat process was forwarding these requests to the public DNS. This is a useful function for DNS requests from the private network, but not for requests from the public network.

These DNS requests are now discarded

H.323: Media Negotiation did not work for call with reverse media and media response in CALL-PROC

Status Closed
Id 84848

Caused SIP interop problems with CUCM

Maximum LDAP PDU Size Too Small

Status Closed
Id 84851

Maximum LDAP PDU Size Too Small. Now internal maximum allocation unit plus a bit for encoding overhead.

PBX Broadcast: Call to group members was not cleared when CFNR was executed

Status Closed
Id 84857

This was a collateral damage from
fix: #79549: PBX Waiting/Broadcast: Incomplete CDRs if CFNR configured on object

IPVA: V10 Code Merge Aiming To Stabilize FW Upload To DRAM

Status Closed
Id 84862

Addresses a problem in conjunction with innovaphone's automated test environment

PBX: Filters should only be applied to calls with media channels

Status Closed
Id 84892

A join group operation could be prohibited with filters. This was unexpected.

phone: ip222, ip232: raise alarm if an USB Headset does not respond on USB bus anymore

Status Closed
Id 84893


SIP: Fix for media negotiation on calls re-routed from TONE interface to outbound SIP

Status Closed
Id 84932

Helps on gateway interfaces configured for media-relay with exclusive codec.

IP-DECT: Call transfer timer

Status Closed
Id 84951

The call transfer timer is stopped with the call proceeding event now. This fixes a call transfer to e.g. a mobile user with a delayed alert.

IP-DECT: Reverse phone book

Status Closed
Id 84953

The reverse phone book does not work till hotfix 11. This is fixed again now.

SIP: Loosing remote IP addresses when DNS becomes temorarily unavailable

Status Closed
Id 84991

Loosing remote IP addresses when DNS becomes temorarily unavailable.
Results in interfaces without remote ip addresses.


PBX Routing: A CFNR on a slave gateway object redirecting the call to the master failed

Status Closed
Id 85017

This was a collateral damage of

fix: #77874: PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX

SIP: Switch to T.38 did not work when interworking with H.323 slowstart (XCAPI)

Status Closed
Id 85047

Switch to T.38 did not work when interworking with H.323 slowstart (XCAPI).

IP222 IP232 IP241: Headet volume adjust added, re-enable noise reduction in headset mode

Status Closed
Id 85058

Headset volume adjust can be done with the webinterface at
http://addr/dsp.xml

This feature can be used to adapt to different DHSG headsets.

The noise reduction in headset mode is re-enabled to avoid sporadic noise with

SIP: Memory leak

Status Closed
Id 85083

Memory leak.

SIP: Fix for STUN problems

Status Closed
Id 85118

Try alternative STUN server address if first fails and another was gathered by DNS.

IP232,IP222,IP241: Do not hide configured fkey if neither text nor icon is displayed

Status Closed
Id 85211

Do not hide configured fkey if neither text nor icon is displayed.
E.g. Call Forwarding fkey with CF destination but no label text.

PBX: Slave with non-ASCII PBX name did not register at master

Status Closed
Id 85235

Error in utf-8 to unicode convertion in this case

V9 Hotfix 16 (9.061101)

Changes included in Version 9 hotfix16 Definition

New Features

Media: Preparing for G.722.1

Status Closed
Id 85316

Decoding SDP containing G.722.1

\tv=0
\to=- 5140 5141 IN IP4 10.138.6.91
\ts=-
\tc=IN IP4 10.138.6.91
\tt=0 0
\tm=audio 50000 RTP/AVP 122 8 0 18 121 101
\ta=rtpmap:122 G7221/16000
\ta=fmtp:122 bitrate=32000
\ta=rtpmap:121 L16/16000
\ta=rtpmap:18 G729/8000
\ta=fmtp:18 annexb=no
\ta=rtpmap:101 telephone-event/8000
\ta=fmtp:101 0-15

SIP: New config file option /tls-unchecked

Status Closed
Id 85449

New config file option /tls-unchecked disables validation of remote server certificate.

IP-DECT: Configuration changes for OEM device

Status Closed
Id 85522

For an OEM device the configuration update behaviour is changed.


SIP: New config file option /fixed-contact-addr

Status Closed
Id 85529

New config file option /fixed-contact-addr to keep SIP client from changing it's Contact address into public address of NAT mapping after registration.
(RFC-3581 Symmetric Response Routing)

IP-DECT: DECT security (3)

Status Closed
Id 85554

Some changes in preparation for the new feature "DECT security". This feature will be available later (IP1202).


Phones: New config file option /recording-without-remote-party-info

Status Closed
Id 85833

For interop of recording feature with VOXTRON application.
VOXTRON application gets confused by Diversion header in INVITE.

config change PHONE SIG /recording-without-remote-party-info

phone_orchid: configuration prameters to adjust microphone and speaker volume

Status Closed
Id 85848

These parameters set a volume correction factor which is applied at any volume level. Parameter changes are applied immediately even in an active call.

config add AC-DSP0 HEADSET /mic-volume <mic-adjust> /spk-volume <spk-adjust>
config write
config activate

-20 <= <*-adjust> <= +20


IP-DECT: Radio reconnect handling with OEM PBX

Status Closed
Id 85952

The handling for the endpoint location update in combination of a radio reconnect is changed. This fixes the base station behaviour with an OEM PBX.

ip1202: support capture of raw ethernet packets exchanged between MSP (aka DSP) and the firmware

Status Closed
Id 86049

All exchange between the firmware running on the ACP (Application Command Processor) and the firmware running on the MSP (Media Strem Processor) is in ethernet packet format. The Mindspeed support prefers this trace format.
The capture is enabled via
config add MSP0 /mtrace
It includes as well command and RTP data packets and thus duplicates the RTP packets traced by the general "All TCP/UDP Traffic" option.

HTTP-Client: Requests with specified credentials

Status Closed
Id 86133

Additional function for OEM httpclient::auth_request
It uses the specified credentials and ignores the username and password from the configuration.

Phones: New config option "Allow User Settings at Phone"

Status Closed
Id 86243

Allow changing of User Settings even if 'Protect Configuration at Phone' is activated.



IP-DECT: New hidden Radio option 'Max RTP streams'

Status Closed
Id 86527

The new hidden option 'Max RTP streams' is added to the IP-DECT Radio module. The option is only visible for an OEM device, but can be used with config change command ("/max-rtp-streams <count>"). The feature is useful to limit the RTP streams for radios connected to the IP-DECT Master with a low data bandwidth. Conference calls are not limited with this feature.

Gateway: New config option "No blind transfer"

Status Closed
Id 86689

New config option "No blind transfer" to keep Gateway from handling blind transfer requests.
If set blind transfer requests are passed through.
Handling is performed at the next signaling hop.

Bug Fixes

phone: ip222, ip232: USB Headset could not be disabled via Menu or by Headset Function in Enable mode

Status Closed
Id 84444

sometimes the user want's to use the phone as if no headset is connected, i.e. when for example redial key is pressed after a number has been entered or a list entry has been selected the call should be started in handsfree mode and not in headset mode.
If now the headset is disabled via menu or the headset(enable) function key all headset functions are completely disabled and no calls are directed to the headset, the status bar displays an icon indicating the disabled state.


H.323: Interop Problem with CUCM SIP for Transfer

Status Closed
Id 85120

For the transfer the CUCM first sets the call on hold and then requests a new media proposal from this call, which we cannot deliver. The request is just ignored, there should be an answer.

IP152: Call replacement (blind transfer) did not work

Status Closed
Id 85313

Call replacement (blind transfer) ends up in no audio.

Trap when disabling Gateway interface

Status Closed
Id 85377

0:0204:814:3 - SIP_UDP.17 -> SIP_TRANSPORT.1 : SOCKET_RECVFROM_RESULT(87.173.157.2:5060,10.2.2.1:5060;95fd:75c0:9485:bc94:9485:bc30:9406:630c:5060,948a:2198:9476:5924:19:72:0:1:5060)
data(913),SIP_TRANSPORT

\tREGISTER sip:gwdl_ip800 SIP/2.0
\tVia: SIP/2.0/UDP 87.173.157.2:5060;branch=z9hG4bKA072411261DD0135
\tRoute: <sip:87.139.89.223;lr>
\tFrom: <sip:93@gwdl_ip800>;tag=4110839322
\tTo: <sip:93@gwdl_ip800>
\tCall-ID: DECE171E57A1898F@192.168.178.1
\tCSeq: 7998 REGISTER
\tContact: <sip:93@87.173.157.2;uniq=D526C620EFAB7D1ACBBC469D33242>
\tAuthorization: Digest username="_!(schuetz)", realm="gwdl_ip800", nonce="7977d820e909d311", uri="sip:gwdl_ip800", response="6999f52d0e7d4ddbd18b23b59936aaa6", algorithm=MD5, cnonce="F56053DD72D3E1B3", qop=auth, nc=00000401
\tExpires: 1800
\tMax-Forwards: 70
\tUser-Agent: AVM FRITZ!Box Fon WLAN 7112 (UI) 87.04.87 (Jun 7 2011)
\tSupported: 100rel,replaces,timer
\tAllow-Events: telephone-event,refer,reg
\tAllow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
\tAccept: application/sdp, multipart/mixed
\tAccept-Encoding: identity
\tContent-Length: 0


0:0204:815:5 - SIP-Client(SIP-CLIENT.0) <sip:93@gwdl_ip800;user=phone> ...
0:0204:816:5 - SIP-Client(SIP-CLIENT.0) Idle->Registering
0:0204:816:6 - SIP_RAS_APP.3 -> GK.0 : RAS_DISCOVERY, ip=87.173.157.2
data(6),GK
00 00 00 02 39 33 ....93
0:0204:816:7 - GK.0 -> SIP_RAS_APP.3 : RAS_DISCOVERY_CONFIRM
0:0204:817:0 - SIP_RAS_APP.3 -> GK.0 : RAS_REGISTRATION(87.173.157.2:5060)
data(6),GK
00 00 00 02 39 33 ....93
0:0204:817:1 - GK.0 -> SIP_RAS_APP.3 : RAS_REGISTRATION_CONFIRM
data(32),SIP_RAS_APP
00 00 00 02 39 33 00 01 00 16 00 5f 00 21 00 28 ....93....._.!.(
00 73 00 63 00 68 00 75 00 65 00 74 00 7a 00 29 .s.c.h.u.e.t.z.)
0:0204:817:1 - DEBUG this=948b325c event.sig=0
0:0204:817:2 - DEBUG this->reg_reference=0 event.reference=948b3ebc

Presence note got lost

Status Closed
Id 85469

When configuring a presence on the phone (Main menu/User setup/Presence)
the note is not save along with the selected activity

PBX: Potential trap on registration of a slave PBX, with Master GK-ID configuration changes

Status Closed
Id 85477

The trap happened if on the Slave a Master GK-ID was configured, then the slave registered, and afterwards the Master GK-ID was removed again and the slave registered again and then was restarted once more.

Presence: Note moves into 'tel' presence

Status Closed
Id 85482

Presence note moves into 'tel' presence when changing activity by use of 'presence' fkey.

IP232,IP222: Automatically enter input mode 'alpha' when entering directory search screen through fkey

Status Closed
Id 85493

Automatically enter input mode 'alpha' when entering directory search screen through fkey.

SIP: Interop with Genband SBC

Status Closed
Id 85534

Handling this kind of SDP offer:

\tv=0
\to=IOTMSX1-0 17 2 IN IP4 206.165.51.38
\ts=sip call
\tc=IN IP4 0.0.0.0
\tt=0 0
\tm=audio 42076 RTP/AVP 0 8 4 18 101 13
\ta=sendonly
\ta=rtpmap:101 telephone-event/8000
\ta=fmtp:18 annexa=yes
\ta=fmtp:18 annexb=no
\ta=fmtp:101 0-15
\ta=ptime:20
\ta=silenceSupp:off - - - -


PBX-Exec: Call at secretary for executive had wrong destination name/number

Status Closed
Id 85535

Call should be displayed as call to the scretary, diverted by the executive

Media: RTP did not work on switch from slowstart T.38 to EFC G.711

Status Closed
Id 85548

T.38 retransmission packets were sent, which caused problems with the RTP

IP222 IP232 IP241: DSP Update to Version 680.05

Status Closed
Id 85552

Sporadic problems with noise reduction fixed.
Noise reduction can be enabled for Handset/Headset/Handsfree mode

SIP: Providing private RTP address although public RTP address is available

Status Closed
Id 85751

During re-negotiation:
Providing private RTP address to external endpoint although public RTP address is available.

SIP: Do not send "504 Server Time-out" after "200 OK"

Status Closed
Id 85788

Do not send "504 Server Time-out" after "200 OK".

IP-DECT: Call transfer compatibility with SIP

Status Closed
Id 85795

For compatibility reasons with the SIP protocol no hold signal should be sent to the call transfer destination. This is fixed again.

SIP: Handling of "sendonly" offer after "inactive" offer

Status Closed
Id 85857

Switching from "inactive" into "recvonly".
Giving REMOTE_RETRIEVE and HOLD_NOTIFY to app.

SIP: Ignore From-URI in re-INVITE and UPDATE when "from-change" not supported

Status Closed
Id 85873

Ignore From-URI in re-INVITE and UPDATE when "from-change" not supported by remote side.

Phones: Touching should start dialing of selected directory entry

Status Closed
Id 85959

Touching should start dialing of selected directory entry
instead of opening directory entry for editing.
At least in Context of Wahlvorbereitung.

SNMP-Traps: agent-addr wasn't reflecting altered IP adress

Status Closed
Id 85970

SNMP-Traps: The SNMP trap's agent-addr member carried an outdated ip adress if the DHCP leased ip address changed.

SIP: Optimization when validating local media address

Status Closed
Id 86029

Optimization when validating local media address.
Keep number of SOCKET_GET_LOCAL_ADDR low.

DHCP: client IP connectivity lost when a renew/rebind request for the currently assigned address was refused by the server

Status Closed
Id 86030

When a client renew/rebind request is refused by the server providing the current address the client starts a new discovery. But in case of success the new address was not set and the client could not be reached anymore.

ENUM: Port in SIP-URI was not honored

Status Closed
Id 86037

Port in SIP-URI was not honored in regex of DNS result.

IP-DECT: Transferred/rerouted call display

Status Closed
Id 86041

The remote party number of transferred and rerouted calls are not correctly shown in the handset's display. This fixes the display of CTI initiated calls.

IP-DECT: MAC-alias change of OEM device

Status Closed
Id 86047

The MAC-alias of an OEM device was changed and this results in conflicts within several DECT modules. Different product short names of the same device are correctly accepted now.


ip1202: Send Inband DTMF did not work

Status Closed
Id 86052

the "DECT/Config/Master/Send Inband DTMF" checkmark had no effect.

SIP: Support for multiple audio media descriptions

Status Closed
Id 86083

One without encryption and one with encryption:

v=0
o=OpenStage-Line_0 968610650 446118927 IN IP4 10.30.1.144
s=SIP Call
c=IN IP4 10.30.1.144
t=0 0
m=audio 5012 RTP/AVP 9 8 0 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv
m=audio 5010 RTP/SAVP 9 8 0 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ujVU8G6kgknZnPflRwx8tadNskkp9glas/DFCbC3
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:+uoFBNVVhDY5OOGvMOdAlvpxvc98hX/VeProhlwH
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv


Change order of links on debug page

Status Closed
Id 86121

Make "Tracing" the first item so that no leak check is triggered when opening the page.

Phones: Using config option "CGPN" as CGPN on outbound SIP calls

Status Closed
Id 86122

Using config option "CGPN" as CGPN on outbound SIP calls.

phone: ip222, ip232: recover from USB port failure probably caused by electric sparks

Status Closed
Id 86125

An USB port failure is indicated when a Plantronics DA45 headset adapter is plugged and a certain kind of table lamp (halogen) is switched on or off.
It happens independent of current state of the headset (idle or in call)
but only with the abovementioned adapter.
The exact reason is not known yet, may be it's an electric spark from the switch of the lamp or some pulse.
The fix is to reset the port and to restart the plugin process, a possibly active call is terminated.


phone: coder settings of a "Create Registration" function key were not applied to the created registration

Status Closed
Id 86164

Disable leak check if debug flag is not set

Status Closed
Id 86165

The leak check is only allowed if the config flag CPU /debug is set.

AD Replication: Configuration Buffer Increased

Status Closed
Id 86211

Was too small for many maps

whistling tone in all ip28 a/b ports on incoming call

Status Closed
Id 86212

This phenomena occured after few day uptime after sending CLIP

ip72 firmware did not boot anymore since V9hotfix11

Status Closed
Id 86246

SIP: Avoid re-configuration of DSP channel when processing re-INVITE

Status Closed
Id 86316

Fix is required for interop with SIP devices sending re-INVITE for session-refresh,
but incrementing version field in SDP body, altough there is no change in SDP.

phone: headset function key mode 'control' could be configured via WEB interface only

Status Closed
Id 86327

must be possible locally at the phone too

phone: Partner state provided via dialog info was not reset when the subscription call was released because of network errors

Status Closed
Id 86330

Gateway: Routing on blind transfer call starts from wrong IF

Status Closed
Id 86386

Transfered endpoint was used as source interface on routing.
Better use transfering endpoint as source on routing of (blind) transfer call.
Also transfer-to endpoint missed ctSetup.
Also transfered endpoint missed ctComplete.

Linux: Start-up failures

Status Closed
Id 86399

Linux start-up is improved. This fixes hanging Linux start-ups caused by compact-flash failures.

IP-DECT: GUI Master Configuration

Status Closed
Id 86442

Disabling LDAP Directory Search fixed for the IP1202.

Linux: GUI in viewer mode

Status Closed
Id 86455

The Linux General page is fixed for the viewer only mode.

SIP: PRACK after CANCEL contains bad RAck header value

Status Closed
Id 86469

CSeq or original INVITE transaction was damaged.
But only if CANCEL has been sent right before PRACK.
CANCEL is sent before PRACK only if SDP answer of provisional response is invalid.

PBX: Conference trap

Status Closed
Id 86473

Potential trap in the PBX BC conference object is fixed.

SNMP: Obsoleted Enterprise-Specific Trap "innoIsdnFailure"

Status Closed
Id 86513

This SNMP trap is no longer necessary. Meanwhile it is covered more consistently by "innoDiagAlarm" and "innoDiagAlarmClear".

Linux: GUI removed from IP1060

Status Closed
Id 86547

The IP1060 does not support the Linux Application Platform. The GUI is removed now. Also a trap is fixed, if somebody tries to enable Linux.

Voicemail: <pbx-disc> failed sometimes

Status Closed
Id 86569

In case of unconnected calls that were subject of a prior <pbx-fwd>

phone: ip222, ip232: handset/handsfree speaker was not switched off when changing over to headset mode

Status Closed
Id 86706

When the headset talk-key or the phone headset-control-key is pressed while the phone is in handset or handsfree mode the phone changes to headset mode, i.e. headset micro and speaker are activated. The handset or handsfree speaker should be switched off then.

phone_orchid: pressing the speaker key while in headset mode did not switch over to handsfree mode

Status Closed
Id 86707

When the speaker key is pressed while the phone is in headset mode the headset should be deactivated and the phone should enter handsfree mode.

V9 Hotfix 17 (9061152)

Changes included in Version 9 hotfix17 Definition

New Features

Voicemail: <pbx-query-obj type="filter"/> couldn't read template distributed filter

Status Closed
Id 86459

Switched to a different internal api method

AD Replication: Refactoring Poll-Timer-based Replication

Status Closed
Id 86476

-Ridded usnChanged filter term
-Added two status messages

PBX Mobility: Take precaution against hanging calls because of misbehaving mobile phone or client

Status Closed
Id 86813

Use a 20s timer to terminate any call to the mobility object if no internal call is initiated

IPVA: Query Physical Link Status.

Status Closed
Id 86937

-The physical link status wasn't evaluated so far. Now it is.
-Added statistics page 'Interfaces/EthX/Statistics'

IP TOS value is now traced with Wireshark

Status Closed
Id 87025

This value is now correctly traced within IP4 UDP/TCP traces.

PBX Trunk: New destination for rejected calls

Status Closed
Id 87151

Calls which are rejected (busy after alert), can be redirected to a configurable destination.

SIP: Offer media encryption as separate media description

Status Closed
Id 87152

Interop issue.
New config file option "config change SIP /separate-encryption".

Offer:
\tv=0
\to=- 10 1 IN IP4 172.16.16.156
\ts=-
\tc=IN IP4 172.16.16.156
\tt=0 0
\tm=audio 16390 RTP/AVP 9 8 0 18 101 13
\tc=IN IP4 172.16.16.156
\ta=rtpmap:101 telephone-event/8000
\ta=fmtp:18 annexa=yes
\ta=fmtp:18 annexb=no
\ta=fmtp:101 0-15
\ta=ptime:30
\ta=silenceSupp:off - - - -
\ta=sendrecv
\tm=audio 16390 RTP/SAVP 9 8 0 18 101 13
\tc=IN IP4 172.16.16.156
\ta=rtpmap:101 telephone-event/8000
\ta=fmtp:18 annexa=yes
\ta=fmtp:18 annexb=no
\ta=fmtp:101 0-15
\ta=ptime:30
\ta=silenceSupp:off - - - -
\ta=sendrecv
\ta=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:VoGZgKwImVwTkJg4jBeYFBafl/CyJpfMX66WqDMZ

Instead of:
\tv=0
\to=- 9 1 IN IP4 172.16.16.156
\ts=-
\tc=IN IP4 172.16.16.156
\tt=0 0
\tm=audio 16386 RTP/SAVP 9 8 0 18 101 13
\tc=IN IP4 172.16.16.156
\ta=rtpmap:101 telephone-event/8000
\ta=fmtp:18 annexa=yes
\ta=fmtp:18 annexb=no
\ta=fmtp:101 0-15
\ta=ptime:30
\ta=silenceSupp:off - - - -
\ta=sendrecv
\ta=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:mDPFKhNdhm6fhkeyPbAI9uzZyTmtA4t9fy1MwXiD


IP-DECT: DECT security (4)

Status Closed
Id 87157

Some changes in preparation for the new feature "DECT security". This feature will be available later (IP1202).

SIP: Do registration refresh more ahead of expiration

Status Closed
Id 87381

Instead of refreshing 2 secs before registration expiration,
do the refresh after 98,33% of the registration ttl.

TTL=120secs -> Refresh 2secs before expiration
TTL=3600secs -> Refresh 60secs before expiration

phone: Modification of function key on phone can be disabled for any single key

Status Closed
Id 87405

A "Disable Modification on Phone" checkmark will be provided in the edit menu for each key. If checked the key cannot be edited on the phone anymore.
This mechanism works in addition to the phone local key type mask set via
"Phone/Protect/Function keys not modifiable on the phone"
A key of a type NOT marked as ``not editable´´ in this mask can be made ``not editable´´ by setting the above mentioned checkmark
A key of a type marked as ``not editable´´ remmains not editable, independent of the checkmark setting.



phone: "Spare" function key to reserve key positions for administrative purposes

Status Closed
Id 87406

This new key permits to reserve key positions for later definition by administrator.

IP-DECT: Remote hold in conference mode

Status Closed
Id 87487

If a hold notify message is received from a remote party and the conference mode is active, now the message is forwarded to the conference unit. This prevents the music on hold in conference calls. The state is also shown in the radio call list.


Webdav: Write information into trace if DELETE fails because file is in open state

Status Closed
Id 87488

Write information about Webdav session holding open file handle.

H.323: Better error handling in case of incompatible SRTP parameters

Status Closed
Id 87624

Was needed for DECT OEM.

PBX: No CDR was generated for enblock call, which was rejected because of filter config

Status Closed
Id 87869

A CDR with cause 52 (outgoing call barred) is generated

Voicemail: Add <pbx-getcallinfo out-leg2-orig="...">

Status Closed
Id 87880

Allows to query the divertingLegInformation2.orignalCalled number

IP-DECT: Fault logging for Master module

Status Closed
Id 88005

Call and channel fault logging for the IP-DECT Master module are added.

Voicemail: <pbx-prepcallinfo leg2=".." leg2-name=".."/>

Status Closed
Id 88351

New attributes allow to prepare <pbx-fwd> in such, that the supplementary service divertingLegInformation2 is going to be sent.

PBX: Allow configuration of default presence/dialog-info visibility for group members

Status Closed
Id 88352

Active group members got full presence/dialog-info because this matched the visibility be group-indications. However this is not desired always, so it can now be configured to restrict this.

IP-DECT: Unused OEM modules removed

Status Closed
Id 88413

Unused OEM modules has been removed now.

Gateway: Make interop flag "Ack incoming call" configurable on UI

Status Closed
Id 88499

This was a hidden interop flag, but was now needed multiple times, so it is more efficient to have it configurable on the UI

IP-DECT: Logging for handover calls

Status Closed
Id 88536

Logging events for IP-DECT handover calls added.

Bug Fixes

make update script parsing more tolerant to suspicious line ends

Status Closed
Id 84349

Some WEBDAV tools garble line end when a text file is stored after editing. Last seen \\r\\r
instead of \\r
. Any sequence consisting only of \\r
chars should be read as one line end because empty lines have no meaning in an update script.

SIP: Do not interwork holdNotific and retrieveNotific while on hold

Status Closed
Id 86736

Interworking problem with VOXTRON/XCAPI
Do not interwork holdNotific and retrieveNotific after remoteHold.

Disable LDAP Server When Erasing Flash Directory

Status Closed
Id 86763

During an upload of a complete configuration the command "mod cmd FLASHDIR0 erase-all" will erase all flash directory content. Replication clients are going to receive nil-responses making them assume a certain entry does no longer exist.

Replication clients are now barred from accessing the LDAP server as long as the box didn't process the post-upload reset.

IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: fax bad signal quality events are sent on good fax connections

Status Closed
Id 86788

Sometimes during fax transfer a bad signal quality (e.g.50) is reported, even if the connection is good. This happens during the TCF phase, in the image phase the signal quality is fine ( e.g. 3)


Mis-typing on Maintenance/Diagnostics/Counters

Status Closed
Id 86812

Show "kbit/s" instead of "kBbit/s".

Gateway: #11 could not be dialed on analog interfaces with feature codes enabled

Status Closed
Id 86819

This is a featiure code used on DECT systems and it was not disabled on analog interfaces

Gateway: Potential trap when recording is configured and a transfer happens

Status Closed
Id 86837

The problem is with a transfer, which is executed in the Gateway, not the PBX

PBX: XML Export/Import did not work, if DTMF Feature objects present

Status Closed
Id 86870

The automatically generated user objects caused a problem. This could result in a config that caused the PBX to restart in a loop. The export/import was fixed and the PBX does not restart because of the corrupt config any more.

PBX: Trap if a Hold was attempted for a call without media

Status Closed
Id 86874

Could be caused by a misbehaving application or voip device

PBX: Name beginning with '*' caused problems with SOAP applications

Status Closed
Id 86882

A search for such a name was treated as wildcard search for all users. The name '*' is now a reserved name, which cannot be used for an object.

Linux: IP0010 available again

Status Closed
Id 86903

With V9 hotfix 16 Linux can not be started on the IP0010. This is fixed now.


phone: call diversion override via indirect dialing could not be disabled

Status Closed
Id 86944

In the indirect dialing screen the right arrow key opens a menu with different options how to place the call.
"Dial - No Diversion" ssets up a call which will ignore the diversions active on the target phone. This menu item can be supressed via the "Fine grained function locking" bit PHONE_LOCK_DIVERSION_OVERRIDE 0x04000000

PBX Waiting: CFNR with number filter did not work

Status Closed
Id 86959

The CFNR was not executed

PBX Conference: Calling Party was missing in the CDRs for calls to conference members

Status Closed
Id 86977

Only the conference object itself was present as forwarding party

SIP: Rare trap when cancelling call

Status Closed
Id 86994

When call abort interferes with re-routing in gateway application.

PBX: Replication from a dyn PBX was not possible

Status Closed
Id 87065

Configuration of dyn PBX id now possible for replication

TLS: Ignore incoming HelloRequest messages

Status Closed
Id 87091

When receiving a HelloRequest the TLS client dopped the connection and sent an "Unexpected message" alert.

Now the HelloRequest is ignored and an "No renegotiation" warning is sent.

PBX: Call, which was rejected with busy, because of CF loop, could not be forwarded by Trunk Busy destination

Status Closed
Id 87150

Such a call was always disconnected

IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: DSP Update to Version 680.05

Status Closed
Id 87179

The jitter buffer performance has been improved.
The fax/modem bypass performance has been improved.
IP28 firmware size was reduced ( unused code is not linked ).
IP28 G279 didnt work with more than 4 channels.

IP222 IP232 IP241: Disable PC port didnt work

Status Closed
Id 87250

IP-DECT: Login feature with disabled local R-key handling

Status Closed
Id 87269

The login feature is fixed now if the local R-key handling is disabled.

Webdav: DELETE may fail if file has been opened for reading before

Status Closed
Id 87288

If reading (GET) stops but HTTP session remains open, the file remains in state 'open'
and subsequent DELETE request fails with "500 Internal Server Error".
Close file and re-try to delete.

HTTP: Chunked transfer fails if the chunk header is not in a single packet

Status Closed
Id 87292

PBX: Conference and hanging VM script

Status Closed
Id 87312

Wrong media initializing in the BC Conference object causes a hanging VM script if a file should be played. This is fixed now.

IP-DECT: No voice with early handover

Status Closed
Id 87504

This fixes outgoing calls without voice if early handover is done before the media channel is initialized.


phone: ip222, ip232: multiple reports indicating offhook were misinterpreted and could drop a call

Status Closed
Id 87506

Wireless USB headsets may send more than one report indicating headset offhook state in conjunction with different wireless link states. If the interval between the first and the second indication was very short (8 ms) the second indication was misinterpreted and the just setup call was dropped (observed with a Jabra PRO 930 after plugin).

SIP: Fix for Contact-URI in 200/OK for SUBSCRIBE

Status Closed
Id 87507

Fix for Contact-URI in 200/OK for SUBSCRIBE and NOTIFY requests.
May contain double port attribue:

SUBSCRIBE sip:8011@172.20.11.53:2053 SIP/2.0

Contact: <sip:8011@172.20.11.53:2053:2053;maddr=172.20.11.53;transport=UDP>

Status: Fixed in 10.00 and 9.00

IP-DECT: Disturbances GUI info change

Status Closed
Id 87522

The text of the GUI page disturbances is changed.

IP-DECT: Web UI info page, version and release state

Status Closed
Id 87538

Now the version and the release state are shown on the web UI info page of the IP1202.

PBX: Blind transfer to WQ by IP-DECT caused hanging calls

Status Closed
Id 87542

The call-leg to the transfering phone was not cleared by the PBX, so if the phone did not clear this call, it was hanging for ever. Other phones clear such a call after a timeout, but this is only a workaround, the call must be cleared by the PBX

H.323: Problem sending real big signaling messages

Status Closed
Id 87543

Happened with a configuration with more the 60 Join Group function keys on the phone. The message was not sent and the operation failed.

PBX-SOAP: UserCall on WQ did not take "Send Number" into account and call was not marked as internal

Status Closed
Id 87559

Calls should be send with "Send Number" as source if configured

PBX: CFB configured at PBX object did not patch number correctly

Status Closed
Id 87581

In case the called endpoint was in different node then PBX object, the number did not contain all the needed prefixes

SIP: Trap when logging out dect user

Status Closed
Id 87606

Trap when logging out dect user.

0:0287:977:1 - MASTER_EP-SIG.0 default(948f3e18): serial_event(100) src=DECTMASTER-CALL.0 mod=SIP
0:0287:977:1 - Assertion failed line 790 in common/os/os.cpp, object deleted
0:0287:977:2 - assert-ep 94004474 called from 94055498

PBX: Called number was missing in CDRs for calls to busy endpoint

Status Closed
Id 87620

In case there was no other event then rel-to/from and the call was dialed with overlap sending

SIP: Handset type information also in Subscribe User-Agent

Status Closed
Id 87621

Handset type information also in Subscribe User-Agent as sent in REGISTER request.


H.323: Media Negotiation did not work for Hold/Retrieve after a DECT conference

Status Closed
Id 87635

No media after retrieve

PBX: Conference trap

Status Closed
Id 87662

A trap in the BC conference PBX object is fixed.

PBX Waiting: Missing ringback on call forward after announcement

Status Closed
Id 87674

This was a collateral damage of

fix: #81370: PBX Waiting: Call state shows "Disconnecting" after switch from announcement 1 to announcement 2

Gateway: Routing on blind transfer call starts from wrong IF (undo of #86386)

Status Closed
Id 87678

Rollback of #86386: Gateway: Routing on blind transfer call starts from wrong IF

There are other installations where the old behaviour is the expected behaviour.

PBX Waiting: DTMF overlap dialing or blind transfer to same Waiting object was rejected with busy

Status Closed
Id 87681

Even if this was caused by a CFB or CFU on the dialed destination

IP-DECT: No idle display update with own message

Status Closed
Id 87715

A message with the same sender and receiver name overwrites the handset's idle display. This feature is removed now.

Voicemail: <pbx-fwd>, Forward Timer Didn't Fire For Connected Calls

Status Closed
Id 87717

Voicemail: <pbx-fwd>, Forward Timer Didn't Fire For Connected Calls
<!-
pbx_vm.cpp
-->

IP-DECT: Trap with rejected handovers

Status Closed
Id 87730

A trap occurs in the rare situation if the handover is rejected by the radio.

H.323: Q.931 User Info could not be sent in incoming call proceeding state

Status Closed
Id 87740

This is used by the Fax interface

SIP: Parsing of multiple Allow lines was not implemented

Status Closed
Id 87753

Now all Allow lines are parsed (not only first one).

syslog packets were sent with the initially assigned source address even after a new address had been assigned

Status Closed
Id 87754

When a device runs as DHCP client the IP-adress assigned to an interface may change either because the DHCP-Server rejects a renew request and provides a new lease or because a WLAN device enters another network.
An IP-address change may also happen when the DHCP mode of a device is changed from 'disabled' to 'client' without reboot.
The source address of syslog packets does reflect such changes now.


Linux: Trap of IP810 if Linux is started

Status Closed
Id 87763

The IP810 with the V9 hotfix 16 firmware traps if Linux is started. Please do not use this firmware. It is fixed again now.

PBX: CFNR to same node but different PBX failed on Slave

Status Closed
Id 87772

The PBX routing did not work correctly in this case

PBX administration: Viewer could change phone configuration of users

Status Closed
Id 87774

SIP: Don't try to dns-resolve IP adresses

Status Closed
Id 87781

Don't try to dns-resolve IP adresses.

phone: Function key options which need to be enabled in Phone/Userx/Preferences could be edited on phone although disabled

Status Closed
Id 87792

Partner-Intrude: hide when "Phone/Userx/Preferences/Enable Call Intrusion" is not checked
Dial-Announce: hide when "Phone/Userx/Preferences/Announcement Calls/Outgoing/Allow" is not checked


SIP: Error handling huge INVITE requests

Status Closed
Id 87811

Failed to compose provisional and final response.
Lots of error messages in trace and truncated responses.

ERROR: SIP message buffer (2049) exceeded! (40,35,941b0c90)

Gateway: MOH from call on hold in CONF interface calls

Status Closed
Id 87823

If a new call joins the conference or a call on hold retrieves the conference, and there are calls on hold in the conference, the music on hold can be heard in the new or retrieving calls. Now this is fixed.
This affects all devices with a CONF interface, but not the IP800 and the IP305.

IP241,IP222,IP232: Presence note may is not enough truncated on 'presence' fkey

Status Closed
Id 87825

Presence note may is not enough truncated on 'presence' fkey.
Esp. when there's also an icon to be displayed.

PBX Trunk: If a call was forked to a trunk with no-name option, name was removed from the original call as well

Status Closed
Id 87843

A global flag was set on the incoming call, which was wrong.

IP230 IP240: Handset receiver volume increased

Status Closed
Id 87878

receiver volume adapted to changed equalizer settings from fix #80403


phone: ip222, ip232: better handling of spurios headset disconnects

Status Closed
Id 87908

Sometimes USB headsets get disconnected from USB port because of certain electric pulses. To overcome this problem the headset port is reset and the media stream routed to the handset. If the headset comes up (logical plugged) again in a reasonable time the media stream is routed to the headset again. Otherwise the media stream remains on the handset and the call can be continued by taking off the handset.


PBX Trunk: Diverting as Calling Feature should replace the name as well

Status Closed
Id 87913

With this feature the Trunk object uses a Diverting Number as calling party number. But not only the number, but also the Name and Name Id should be replaced

Phones: Allow lcd_dump.bmp to be retrieved with viewer credentials

Status Closed
Id 87945

Allow lcd_dump.bmp to be retrieved with viewer credentials.


IP-DECT: Wrong radio list after MAC-alias change

Status Closed
Id 87950

The radio list can be wrong after the MAC-alias change. This fixes a bug of the feature "MAC-alias change of OEM device" (#86047). This is only relevant for OEM devices.

SIP: Keep Contact-URI when registering via TCP

Status Closed
Id 87969

Keep Contact-URI when registering via TCP, even if rport is present in 200/OK for REGISTER


PBX-SOAP: UserCall with 'cn' as destination to a user in different node did not work

Status Closed
Id 88008

The call is done using the number, but the number needs to be adjusted according to the nodes

OS: Quota mechanism did not work for CPU time consumed by timer handling

Status Closed
Id 88034

This could result in a MAX_BUSY_TICK restart if many timer expired at the same time

HTTP: Chunked transfer sporadic fails with webdav

Status Closed
Id 88078

IP6000 IP2000: Webinterface for ETH1 Link setting didnt work

Status Closed
Id 88137

..

SIP: Decoding problem on application/simple-message-summary

Status Closed
Id 88188

application/simple-message-summary may get decoded wrongly.

Voicemail: Possibility To Switch Off An Internal Automatism

Status Closed
Id 88246

An internal automatism could lead to VM-calls getting disconnected after 15s.
The new URI variable "$_divconn=false" turns off auto-connection for diverted/transferred calls:
http://wiki.innovaphone.com/index.php?title=Howto:Configure_the_innovaphone_Voicemail#URL_Query_String_Variables

SIP: DNS priority value not honored always

Status Closed
Id 88248

DNS priority value not honored, if SRV query returns names without address in additional records.

SIP: Do not take "9564+4631559300" as E.164 number

Status Closed
Id 88254

Do not take "9564+4631559300" as E.164 number.

PBX Boolean: Unpredictable behaviour if more the 16 times are entered

Status Closed
Id 88263

There was a limit of 16 times, which was not checked everywhere. The limit is now 32 and checked, so that no more of 32 times can be entered.

IP222 IP232 IP241: Restart on Jitter buffer overrun

Status Closed
Id 88312

..

SIP: Fix for auto-answer handling

Status Closed
Id 88334

Fix for auto-answer handling.

IP-DECT: SIP/Intop - rejected call transfer by target

Status Closed
Id 88356

If the call transfer target rejects the call in ringing state, no fall-back to the initiator call is done and it is not released. This is fixed now. It is only important for a third party PBX.

H.323: Problem with Media Re-Negotiation on a DECT handover call

Status Closed
Id 88379

The DECT handover call works a little special concerning media renegotiation in a way that local preferences are never honored (the real media negotiation takes place between the original radio and the remote endpoint, the handover radio is just told the result. This special mode did not work correctly

PBX Waiting: User Information Message from announcement interface accidentally forwarded to caller

Status Closed
Id 88426

The announcement interface uses User Information signaling messages to send status information for example at the end of the announcement. This was forwarded to the caller by accident. Usually this does not do any harm, but on some ISDN networks it could result in clearing of the call because of unexpected message.

H.323: Potential Max-Busy-Ticks restart

Status Closed
Id 88441

The H.323 state machine could enter a endless loop

IP-DECT: No media after conference and toggled to held call

Status Closed
Id 88442

If the DECT user leaves the conference mode and toggles to an held call, there is no voice. This is fixed now.

Admin IP: static routes configured at an ETH interface disappear when NAT or VLAN config of this interface is updated

Status Closed
Id 88451

Routes configured under "IP4/ETHn/IP/Static IP Routes" were cleared when the "IP4/ETHn/NAT" page or the "ETHn/VLAN" page was left by pressing "OK".

IP232,IP222,IP241: Reduce memory requirements of display rendering

Status Closed
Id 88485

Reduce memory requirements of display rendering.

PBX: Call Completion monitoring call should be terminated only after the CC ringout call is alerting

Status Closed
Id 88489

This is needed for QSIG interoperability

Viewer was able to download config with standard password

Status Closed
Id 88503

This was a security hole

Gateway: Hanging calls in state "Clearing"

Status Closed
Id 88527

Hanging calls in state "Clearing".
Caused by failed call replacement.

IP-DECT: Potential buffer overrun

Status Closed
Id 88535

Buffer overrun check added in dectusers module for command 'show'.

SIP: Missing UPDATE on call pickup

Status Closed
Id 88550

The caller does not received UPDATE containing the connected party information.

SIP: Handle one way of DTMF only

Status Closed
Id 88572

Handle DTMF received via signaling message (INFO)
or DTMF received via RTP (tlephone-event).
Not both at the same call.

PBX Conference: No Media in case media offer only received with alert/connect from called member

Status Closed
Id 88590

This could happen if a broadcast object was called by a conference

IP222 IP232 IP241: Adjust equalizer above 6.3Khz for lower noise level

Status Closed
Id 88658

to reduce noise with G722 codec


SIP: Memory leak

Status Closed
Id 88688

Memory leak on postponed RETRIEVE_NOTIFY.

SIP: Invalid SDP answer if SDP offer contains RED

Status Closed
Id 88700

Invalid SDP answer if SDP offer contains RED.
Only on local media.

V9 Hotfix 18 (9061158)

Changes included in Version 9 hotfix18 Definition

New Features

IPVA: Enhancement for innovaphone testbed: Indicate whether Upload to DRAM took place

Status Closed
Id 88223

Required for innovaphones' automated testbed

Announcement Calls with DTMF feature code object

Status Closed
Id 88643

A new DTMF feature code allows to make announcement calls to a dialed number.

Alarm and Event forwarding to a SYSLOG server

Status Closed
Id 88659

If "Services/Logging/Alarm and Event Forward Server/Type" is set to SYSLOG the xml-formatted alarm and event info is sent to the Server(s) specified under "Services/Logging/Alarm and Event Forward Server/Address".


Bug Fixes

TLS: Do not ignore early SOCKET_RECV

Status Closed
Id 88668

Allow SOCKET_RECV from application between SOCKET_CONNECT and SOCKET_CONNECT_COMPLETE. This event flow is used by the HTTP client.

Kerberos: Admin UI trap when having too many Kerberos hosts

Status Closed
Id 88698

The problem occured if many Kerberos hosts (~1000) were registered on the server. In this case the box trapped due to an XML encoding problem when opening the page General/Kerberos or PBX/Config/Security.

SIP: Provide display name in 200/OK for SUBSCRIBE

Status Closed
Id 88764

Add display name to To header in 200/OK for SUBSCRIBE.

IP22 IP24 IP28 IP305: Sometimes the DSP stops after sending CLIP

Status Closed
Id 89130

..

PBX: Configuration UI broken for checkmarks on PBX/Config/General and PBX/Config/myPBX

Status Closed
Id 89140

The checkmarks on these two pages where all cleared when the other page was edited.
Collateral damage of

fix: #88352: PBX: Allow configuration of default presence/dialog-info visibility for group members

phone: ip222,ip232: the destination of an acticve diversion was not saved when the diversion was deactivated at phone

Status Closed
Id 89177

The destination (name or number) disappered when the diversion was deactivated. It was also not visible via the WEB interface.

Gateway: Missed FAX pages received with the FAX interface

Status Closed
Id 89587

If the FAX interface is used to receive a FAX document with ECM mode and the transmitting terminal appends additional EOLs, the page counter is wrong and document pages are not written. This is fixed now.

V9 Hotfix 19 (9061180)

Changes included in Version 9 hotfix19 Definition

New Features

PBX-SOAP: Present normalized number of peer also

Status Closed
Id 88521

The SOAP API presents the adjusted number of the peer (called/calling), which is the shortest possible number which can be dialed to call this. It is the same number as displayed on the phone. Sometimes an application needs to know the normalized number of the peer, which is the number in the context of the root node. This number is sent as additional number with the identifier "norm"

IP232,IP222,IP241: Config option to adjust LCD brightness in idle state

Status Closed
Id 89261

-> Main Menu -> Phone Setup -> LCD light (idle state)
Can be tuned down to zero.

H.323: Automatically connect signaling TCP if NAT router is detected

Status Closed
Id 89497

When regestering an endpoint from a private network to a PBX within the public network, the signaling TCP connection must be established and maintained by the endpoint. Otherwise calls to the endpoint are not possible.

PRI-QSIG: Interop config for channel numbering

Status Closed
Id 89578

The QSIG standard defines to use Channel numbers (1-30) instead of timeslot (1-15, 16-31) as it is defined for EDSS1. There are many 'old' QSIG implementations around, which do it wrong. The QSIG-ECMA1 protocol setting is used for these 'old' implementations and the QSIG-ECMA2 setting for standard conform inplementations.

With the QSIG-ECMA1 also 'old' facility coding is used. There is also the combination of standard facility coding and timeslots for channels around so an independent mechanism to configure the channel numbering is needed.

Other new Features

89367 jfr case independence for the characters of the Basic Russian Alphabet added


Bug Fixes

SIP/TLS: Rejecting server certificate

Status Closed
Id 88444

Validating server certificate against configured domain name.
Must be validated against configured proxy domsina name.

SIP: Wrong branch value in Via header in ACK request

Status Closed
Id 89317

Branch value in Via header in ACK request must be new after 200 response.
Branch value in Via header in ACK request must be same after non-200 response.

H.323: RTP-DTMF did not work on exclusive coder/media relay configurations

Status Closed
Id 89328

Problem for DTMF on SIP trunks

AD Replication: A Buffer for Processing The Paged Result Cookie Was Too Small

Status Closed
Id 89385

1KB wasn't enough, now 2KB.

LDAP Expert: "Next"-Browsing through DB failed

Status Closed
Id 89391

URI encoding error

phone: ip222, ip232,ip241: the notification tone indicating a new message when a call is active was sent to remote

Status Closed
Id 89392

instead to notify the receiver of the message the remot party did hear the tone

SIP: "Supported: timer" missing in UPDATE message

Status Closed
Id 89429

"Supported: timer" missing in UPDATE message.

phone: ip222,ip232: humming noise in USB headset speaker in outbound call setup phase, disappears once connected

Status Closed
Id 89432

sometimes a humming noise was heard in the USB headset speaker in the setup phase of an outbound call. it disappeared as soon as the call was connected.

phone: an intrusion call set up via Partner function key could not be cleared at the intruding phone via TAPI

Status Closed
Id 89443

The TAPI interpreted the recording state as a conference

log message forwarding to another innovaphone device did not work since V9hotfix15 (on the receiving device)

Status Closed
Id 89492

this problem is located on the reciving device, it does not depend on the the version of the forwarding device.

IP22 IP24 IP28 IP305: DSP debug code added

Status Closed
Id 89493

..

PBX Mobility: Conference ID not set for outgoing calls initiated by myPBX

Status Closed
Id 89502

This could cause several problems:
- When the call was sent to a local user with multiple registrations, the call to each registration had a different conferenceID, so myPBX could not match these calls to actually being only a single call, so multiple calls were dissplayed
- The CDRs created for this call could not be matched

SIP: Don't tell application that registration is down when handling redirect response

Status Closed
Id 89582

Don't tell application that registration is down when handling redirect response for REGISTER.

SIP: No T.38 parameter when indicating capabilitity only

Status Closed
Id 89617

No T.38 parameter when indicating capabilitity only.

Offer
\tv=0
\to=- 2 1 IN IP4 172.16.16.124
\ts=-
\tt=0 0
\tm=audio 16386 RTP/SAVP 8 101 13
\tc=IN IP4 172.16.16.124
\ta=rtpmap:101 telephone-event/8000
\ta=fmtp:101 0-15
\ta=ptime:20
\ta=silenceSupp:off - - - -
\ta=sendrecv
\ta=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:WJl714I+mpSr47ld1YjYMf8t9xQo0xYHUng1CnDi
\tm=image 0 udptl t38
\tc=IN IP4 172.16.16.124

Instead of
\tv=0
\to=- 2 1 IN IP4 172.16.16.124
\ts=-
\tt=0 0
\tm=audio 16394 RTP/SAVP 8 101 13
\tc=IN IP4 172.16.16.124
\ta=rtpmap:101 telephone-event/8000
\ta=fmtp:101 0-15
\ta=ptime:20
\ta=silenceSupp:off - - - -
\ta=sendrecv
\ta=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:sPngj1zoO9wI1pc1tMTymWCzTgwMoamPuJAFIhga
\tm=image 0 udptl t38
\tc=IN IP4 172.16.16.124
\ta=T38FaxVersion:0
\ta=T38MaxBitRate:14400
\ta=T38FaxFillBitRemoval:0
\ta=T38FaxTranscodingMMR:0
\ta=T38FaxTranscodingJBIG:0
\ta=T38FaxRateManagement:transferredTCF
\ta=T38FaxUdpEC:t38UDPRedundancy


OS: Potential trap when starting a timer, which is already deleted

Status Closed
Id 89628

Could result in a strange trap with PBX transfer/recall

phone: while recording was running for a call this call ould not be cleared via SOAP/TAPI

Status Closed
Id 89629

The TAPI assumed the intruding phone to be in a conference

PBX Admin UI: Reconfiguring user visibility (Access) did not work correctly

Status Closed
Id 89636

If a setting for a user/group was deleted, it could happen that some of the settings (Group, Online, Presence, Dialog, Ids) where copied to the next entry.

Voicemail: Trap

Status Closed
Id 89648

If Trace-Checkmark activated, a diverted call carrying an mwi interrogate was processed

Possible Trap While Getting Post-Mortem Log

Status Closed
Id 89652

Possible Trap While Getting Post-Mortem Log

Media: Redirecting SRTP streams for NAT clients only after successful SRTP authentication

Status Closed
Id 89661

Media endpoints support NAT.
If receiving RTP/SRTP from an address other than negotiated one
media endpoints redirecting their media stream towards source of incoming media stream.
In case of SRTP, this NAT workaround is only executed if incoming media stream has passed authentication.
For securitiy reasons.

AD Replication: Merge v10 code into v9 code

Status Closed
Id 89671

Some recent refactoring works weren't in sync. A state machine didn't reach "Completed".

phone: if a call ringing while the handset was offhook was accepted via SOAP/TAPI the call was not cleared when going onhook

Status Closed
Id 89680

This could happen when the handset was lifted and kept lifted after the disconnect key was prressed. an inbound call arriving in this state could be accepted via SOAP/TAPI and was connected to the handset but the call could not be cleared by going onhook. Only the disconnect key did clear the call.
Now the call is cleared as expected when going onhook.

phone_orchid: pressing speaker key in handset/headset mode switches to handsfree mode, pressing again returns to previous mode

Status Closed
Id 89730

handset/headset plus speaker is not supported on phone_orchid, the previous solution where the connection was dropped when the speaker key was pressed again (see #84297) was perceived as irritating.



IP22 IP24 IP28 IP305: Sometimes the DSP stops after sending CLIP (2)

Status Closed
Id 89760

..

SIP: Bug in handling of INVITE with Replaces

Status Closed
Id 89777

Bug in handling of INVITE with Replaces.
Results in hanging call.

IP-DECT: Trap with call transfer

Status Closed
Id 89786

A trap occurs if a call transfer is received in the IP-DECT radio module. This is fixed now.

Fix for MIPS counter

Status Closed
Id 89804

MIPS counter was incorrect on IP1201 and IP4001

SIP: Media negotiation fails on calls into Waiting Queue

Status Closed
Id 89838

Media negotiation fails on calls into Waiting Queue if
caller put call on hold before WQ agent accepts the call.

'power-off loop' relay switching function failure

Status Closed
Id 89942

Especially POE-switches with higher supply voltages than 48V lead to a decreased timespan of powering the build-in relays of a ip6010/ip810 gateway. The detection of a power-fail condition is therefore derived from the POE ICs which react earlier and thus increases powering time of the relays.

myPBX: Default group visibility was not displayed correctly

Status Closed
Id 89954

The default group visibility can be configured from v9hotfix17. Regardless of that configuration myPBX showed full visibility in the visibilty settings.

IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: DSP Update to Version 680.07

Status Closed
Id 89956

Fixed modem bypass with slow modems

V9 Hotfix 20 (9061198)

Changes included in Version 9 hotfix20 Definition

New Features

Linux: Memory allocation changed for IP810

Status Closed
Id 86420

The memory allocation for the IP810 is changed to 128MB/384MB for innovaphone/Linux.

Important:
Linux must be used with the kernel version 3.4.10 or later. This kernel is included in the Linux Application Platform V9.00 hotfix12 and later. The kernel is automatically updated with the Linux Application Platform V9.00 hotfix12.

phone: new "Do Not Disturb" action "ring once"

Status Closed
Id 89960

If "Phone/User-x/Preferences/Do Not Disturb/Action: ring once" is selected a new inbound call is indicated with a short tone only.
Both the the tone and the duration of the tone can be configured under ""Phone/User-x/Preferences/Ring Tones/Do Not Disturb".
If not configured the default ring tone is played for one and a half second.

PBX: Support of long user-user-informations by SOAP

Status Closed
Id 90029

Support of long user-user-informations (UUI) for SOAP sessions added. A long UUI is split into multiple short UUIs supported by Q.931.
It is required by the FAX interface.

Relay: Support of long user-user-informations by FAX

Status Closed
Id 90030

Support of long user-user-informations (UUI) for the FAX interface added. A long UUI is split into multiple short UUIs supported by Q.931.

Gateway: Support of a header line for FAX documents

Status Closed
Id 90374

Support of a header line for FAX documents is added in the FAX interface.


SIP: Support for MESSAGE inside voice call

Status Closed
Id 90408

Support for text messages inside voice call.

PBX-SOAP: Support for 'rc' and 'srce164' on UserCall on Waiting Queue

Status Closed
Id 90537

If a Waiting Queue is used for outgoing calls, these features can be usefull for some applications

PBX: New configuration option 'Hide connected Number' at object

Status Closed
Id 90693

In some cases it is desireable not to reveal the final destination of a call to a caller. For example a call center agent should not be called directly by the customer.

IP-DECT: Configuration option 'Registration with system password'

Status Closed
Id 91460

The configuration option 'Registration with system password' is added. If ticked, all users are registered with the system password. This is useful, if the PBX users are only allowed to register with the PBX password.

Voicemail: URL-En-/Decoding

Status Closed
Id 91609

new statement allows to URL-encode or URL-decode a string
<lib-enc string=".." string_out="$var" type="url"/>
<lib-dec string=".." string_out="$var" type="url"/>

Bug Fixes

SIP: Auto answer with SDP in ACK

Status Closed
Id 89539

Auto answer results into no-audio when INVITE comes w/o SDP offer.

SIP: Fix for overlap dialing with KPML

Status Closed
Id 89581

Dialing digits entered before KPML subscription is established
need to be queued until KPML subscription is established.

SIP: Re-try INVITE after 407 even if no password configured

Status Closed
Id 90024

Calculate Digest with zero-length password and re-try INVITE.

IP232,IP222,IP241: Truncate directory entry information

Status Closed
Id 90027

Truncate directory entry information to keep from overlapping with number type indication.

IP-DECT: Don't show DTMF in radio call list

Status Closed
Id 90034

User dialled digits during calls which are sent as DTMF should not be shown in the radio call list. This is fixed now.

H.323: Allow media offers with 0.0.0.0 as address

Status Closed
Id 90053

Needed for SIP interoperability. Some third party SIP PBXs use addresses of 0.0.0.0 to indicate that they don't receive media. This may happens if an endpoint is put on hold. We did not forward such an offer and thus no Music on Hold was heard.

PBX: Busy On ... Calls at PBX objects did not take into acccount that a call may be routed back to Slave

Status Closed
Id 90054

The busy on ... calls on PBX objects can be used to limit bandwidth usage between a master and a slave to a certain number of calls. Some calls are sent from a slave to the master and back to the slave if the routing decision cannot be done on the slave alone. This happens if escapes are used which overlap other obects (e.g. the local trunk). It is a common configuration the the E.164 routing scheme.
With this fix, these calls are not counted for this purpose.

Voicemail: Send silence RTP during recording

Status Closed
Id 90095

Send silence RTP during recording for some SIP carriers that do not send RTP without receiving RTP.

IP2000: Prevent blinking error LED IP2000

Status Closed
Id 90102

The firmware tried to load the conference DSP, which is not available on the IP2000


SIP: Memory leak when receiving NOTIFY(message-summary)

Status Closed
Id 90106

Memory leak when receiving NOTIFY(message-summary)

H.323: Support for registration from a private network thru NAT

Status Closed
Id 90306

In case an endpoint registers to a PBX from within a private network thru a NAT router, the signaling TCP connection must be maintained in order to be able to receive calls. When the registration is up a dummy call is sent to the PBX to establish the signaling TCP. This TCP connection is maintained after the dummy call is cleared. If this TCP connection is lost (e.g. NAT Router reset), the Registration is cleared and restarted, so that after the re-registration another dummy call is sent.

This is a fix for the previous fix

fix: #89497: H.323: Automatically connect signaling TCP if NAT router is detected

which did not work well

SIP: SUBSCRIBE using old IP address in Contact field

Status Closed
Id 90320

If the IP address is changed at DHCP renew (or network change) the endpoint will immediately do a re-register to update the SIP Proxy with the new IP address.
All SIP messages but SUBSCRIBE uses the new IP address in the Contact field.

Voicemail: Memory Load High With Repeated Calls To <store-getnext>

Status Closed
Id 90362

The use-case does not happen within the standard voicemail, but within e.g. innovaphone conferencing.

PBX: CC Requests were sent with wrong number if a SendNumber was configured at the user

Status Closed
Id 90432

Usually this is no problem only when interworking with some QSIG PBX's this causes the call-completion to fail.

phone_orchid: dialtone missing when recording is active and the active call is held to open a consultation call

Status Closed
Id 90433

the consultation call could be established but there was no dialtone after pressing the R-key and no ringback tone after the number had been entered.

Possible trap when doing a leak check

Status Closed
Id 90451

When many leaks exist or leak check is done when much tracing is turned on. The leak check itself could cause a watchdog trap, because the collecting of the leaks is done on highest priority so not even the timer interrupt could trigger the watchdog.

IP-DECT: Busy state on maximum call count

Status Closed
Id 90461

The base station does not go to the busy state if the maximum call count is reached and the last call is an incoming call. This is fixed now.

PBX: Support for Opticaller Data Callthru did not work

Status Closed
Id 90480

Support for Opticaller data callthru was added, but did not work

802.1x: Version 3 Frames Weren't Processed

Status Closed
Id 90485

Now they're read initially. Handshakes remains to be in version 1.

PBX: Forking a call to a Trunk with "Outgoing call restricted", causes the original call to be restricted as well

Status Closed
Id 90487

The "Outgoing call restricted" flag on the trunk object to which the call was forked caused the call as a whole to be marked as Calling Line Presentation Restricted.

SIP: Locally configured DNS entries were not used if no DNS server configured

Status Closed
Id 90508

If no DNS server was configured, but DNS names are to be resolved,
local DNS entries can be added (Services/DNS/Hosts).
SIP stack fails with SRV query and does not try A query which would deliver IP address.

IP-DECT: RTP stream

Status Closed
Id 90539

If a remote hold event is received, no RTP data should be sent by the IP-DECT device.
A CTI initiated call is established with a call transfer and a "No Media data received" error event can occur.
This is fixed now.

SIP: Trap on IP-DECT

Status Closed
Id 90569

Trap in GK-CHANNEL when Dectmaster application sends DTMF before call if created.

SIP: Disabled IP-DECT interface tries to register

Status Closed
Id 90597

Disabled IP-DECT interface tries to register.

H.323: Timer to monitor response to setup too short for some traffic cases

Status Closed
Id 90696

This created unnecessary event in IP-DECT systems when calling powered off or out of range handsets

PBX Waiting: Potential Trap if editing while a call is initiated with SOAP

Status Closed
Id 90766

The Waiting object can be used as outgoing dialing object with SOAP. If this is done and the configuration is changed while an outgoing call was pending, a trap could happen

H.323: Potential Trap in special case which could only happen in version 10

Status Closed
Id 90768

This fixed is merged to version 9 only of consistency reasons

SIP: Respect changes in PAI/PPI header when receiving UPDATE with SDP offer

Status Closed
Id 90778

PAI/PPI was processed when receiving UPDATE without SDP offer.
PAI/PPI was ignored when receiving UPDATE with SDP offer.
Now PAI/PPI is processed when receiving UPDATE with SDP offer.


SIP: Possible buffer overrun

Status Closed
Id 90780

Fix for possible buffer overrun.

SIP: Wrong error log "Timeout during media negotiation for call"

Status Closed
Id 90821

Error log "Timeout during media negotiation for call" may occur after re-negotioation.
Re-negotioation occurs during hold/retrieve/transfer.

PBX Trunk: Name to Number Feature did not work with calls to extern

Status Closed
Id 90858

If an endpoint cannot be found by name, the call should be forwarded to extern

PBX CSV Import: Corrupted objects at buffer boundaries

Status Closed
Id 90942

The upload is processed in chunks of 2K. At boundaries of these chunks data could be corrupted. This was fixed and the chunk size increased to 10K

Phone: Trap when selecting registration for a directory entry

Status Closed
Id 90999

Trap when selecting registration for a directory entry,
but only if the registration has either no name or no number.

H.323: No event should be generated in State 11 and 25

Status Closed
Id 91020

State 25 is incoming overlap sending. This means a call was received with incomplete dialing information and the caller failed to dial more digits within the timeout of 2min. This is no indication of any malfunction but only a usage problem, so no event should be generated.

State 11 is disconnecting with inband announcement. A timeout happens if a user listens to the announcement for more then 30s. This could be normal.

SIP: Trap when cancelling call

Status Closed
Id 91023

Double delete of a call entity.

SIP: Heavy TLS retry load when server certificate was rejected

Status Closed
Id 91033

Collateral damage from earlier fixed for DNS refreshing.

H.323: Incoming faststart call was sometimes not accepted as faststart

Status Closed
Id 91136

Only happens if non EFC is used on the incoming call, so this only happens in interop cases with other H.323 equipment

SNMP Get-Next Requests Carrying an Octet-String Value Caused Memory Leak

Status Closed
Id 91215

SIP: Media negotiation for video fails if called through waiting queue or multi reg

Status Closed
Id 91235

Media negotiation for video fails if called through waiting queue or multi reg.
In this case the PBX has to handle offer/offer-collision.
In this case the PBX must select audio and video codec.
In this case the PBX must send SDP answers to both endpoints.

SIP: Secondary target (hostname) is not resolved

Status Closed
Id 91287

Usually a response to a SRV query delivers additional records containing the ip address of any target (hostname).
Some DNS servers do not.
Additional A querys are required.
An A query was issued for the primnary target (most preferred hostname).
No A query was issued for the secondary target (less preferred hostname).
Fixed now.

IP-DECT: Trap in Radio module (IP1202)

Status Closed
Id 91315

A trap in the IP-DECT Radio module occurs if the Mobility Master is used and a duplicate IPEI command is sent to the Master. The Master handles it with a location cancel and an endpoint delete command sent to the radio. If the two commands arrives with no delay, the Radio module traps. This is fixed now.

Ldap Replication from NDS

Status Closed
Id 91347

Skip isDeleted attribute with content others than 'true'. Occurred when replicating from an NDS running in AD compatibility mode.

SIP: Wrong call was disconnected after successful transfer

Status Closed
Id 91349

Wrong call was disconnected after successful transfer.

SIP: SDP answer for T.38 switch-over must contain multiple media descriptions

Status Closed
Id 91377

... if SDP offer contains multiple media descriptions.

PBX: Not possible to login as user with non-full admin rights if 'Password protect all Pages'

Status Closed
Id 91414

Some pages needed for the UI, which are normally not password protected, could not accessed with the reduced rights

SIP: Follow offers ptime proposal

Status Closed
Id 91421

Better follow offers ptime proposal.
Otherwise SAMwin operator does not stop sending re-INVITE.

IP0010 IP1060 IP3010 IP6010 IP22 IP24 IP28 IP302 IP305: switch from modem bypass to voice did not work

Status Closed
Id 91429

In some cases announcments with music trigger modembypass, in this case at least the fallback to voice should work.

H.323: Allow update of Registration password

Status Closed
Id 91440

needed in case a password is changed, which is replicated to a DECT system

H.323: PROGRESS in connected state was treated as 'unexpected'

Status Closed
Id 91483

This happened with H.323 connections without registration when disconnecting a call with inband information (e.g. a call to an ISDN interface). Unnecessary events were generated.

V9 Hotfix 21 (9061222)

Changes included in Version 9 hotfix21 Definition

New Features

PBX: Allow Name (instead of Long Name) to identify user for mobility data call thru

Status Closed
Id 91660

This should simplify Opticaller configuration

IP-DECT: Static ports between Master and Radio

Status Closed
Id 91815

Now the VOIP connections between the Master and the Radio use static ports instead of dynamic ones. This is useful if only a few ports should be opened through a firewall. For calls from the Radio to the Master the ports 1716 and 1717 (TLS) are used. For the default Master connection for calls from the Master to the Radio the ports 1718 and 1719 (TLS) are used. For dynamic Radio-Master connections the ports from 1722 are used. Every connection needs two ports.


IP-DECT: Cipher key index request for security test devices

Status Closed
Id 92223

Cipher key index request procedure is changed to pass the test with security test devices. The cipher key index is used for DECT "Early Encryption"(EE).

Voicemail: <pbx-getcallinfo out-calling-name="..."/>

Status Closed
Id 92286

Pass H.450 callingName into the script

PHONE_SIG_MODE_KEEP_NUMBER_TYPE can be set at phonesig startup to keep type of number in all q931 numbers

Status Closed
Id 92385

PBX Trunk: Flag to block presence/dialog-info subscriptions

Status Closed
Id 92824

Some networks e.g. sip carriers behave badly when receiving subscribes for presence/dialog-info, which cannot be handled, so there is an option added to block these.

SIP: Debug information for problems with STUN

Status Closed
Id 93233

Added debug output to trace problems with STUN.

Gateway: FAX interface User-User-Info error response in disconnect event

Status Closed
Id 93455

User-User-Info response of the FAX interface is not forwarded in the alerting state. The problem exists when call to Fax interface was routed through multiple PBX. Now the UUI response is sent in the disconnect event if the response is a error notification.


IP222 IP232 IP241: DSP code update

Status Closed
Id 94916

DSP code update to version 680


SIP: New interop tweak "No Remote Hold Signaling"

Status Closed
Id 94954

New config option on gateway interfaces "No Remote Hold Signaling".
Disables interworking of "inactive" into remoteHold.

IP6010: DSP trace options improved

Status Closed
Id 95078

T38 trace flag worked only in one direction.
Changing trace options needed a reboot.


IPv6: Disable checkmark on ethernet interfaces added

Status Closed
Id 95452

Allows to disable sending/receiving of IPv6 packets on this interface. This may be desireable for security reasons

IP222 IP232 IP241: DSP code update

Status Closed
Id 95764

Channel was muted during silent periods ( no CNG during NLP operation ).

Bug Fixes

PBX Mobility: Connected number from mobile phone was forwarded to caller

Status Closed
Id 90722

If a call is answered on the mobile phone, it should look identical to the caller to the case that the call was answered locally. This means a connected number from the mobile phone must not be forwarded.

Potential Trap when rapidly switching local Media connections (Conferencing)

Status Closed
Id 90933

There was a race condition when switching local media channels (e.g. ISDN channels to conference interfaces), which could cause media not functioning or even a trap

SIP: Better handling of incoming calls

Status Closed
Id 91153

Msg sequence INVITE,CANCEL,INVITE may result into second call rejected.

IP0010,3010,6010,1060,810: ethernet link down not detected when a cable was unplugged after boot

Status Closed
Id 91600

The link state interrupt was triggered only once after boot, further link state changes were not indicated.

myPBX: Rejecting an incoming broadcast call disconnected the call for all alerting endpoints

Status Closed
Id 91719

The call sould continue alerting on the other endpoints.

phone: ip222, ip232: reset/restart USB headset when a hang condition or a port disconnect condition is detected

Status Closed
Id 91732

For unknown reasons some types of wireless headsets stop working after some hours or days. Either the port state changes to disabled or the device rejects control commands with a stall response. In both cases the device is reset and restarted now. If even this fails the complete USB host controller is reset and in most cases the device returns to operational state thereafter


phone: ip222, ip232: Some USB headsets were not detected after a soft reset

Status Closed
Id 91734

This was observed with Jabra BIZ 2400 USB at IP222/232 with hardware build 800 and newer.

IP-DECT: Trap with data calls

Status Closed
Id 91807

The IP-DECT Radio traps if a data call is released and the release includes a facility.

Kerberos administration: Increase maximum number of Kerberos users from 20 to 50

Status Closed
Id 91847

Increase limit on page General/Kerberos.
Use POST for submitting form, instead of GET.

Voicemail: <pbx-getcallinfo out-leg2-name=".." out-leg2-orig-name="..">

Status Closed
Id 91874

Passing name info of divertingLegInformation2 facility into the script.

SRTP: Avoid one-way media with high start sequence numbers

Status Closed
Id 91892

In some cases SRTP calls had one-way media because the RTP sequence number wrapped from 65535 to 0 at be beginning of the call before the receiver started receiving and processing packets.

The scope of start sequence numbers for RTP streams is changed from [0;65535] to [0;32767] to make sure that the receiver can always receive packets before the overflow happens.

The calculation of the roll-over counter (ROC) is also improved to be more reliable.

H.323: Unnecessary re-initializing of rtp-channel on incoming calls to phone

Status Closed
Id 91898

This did not create any problems except CPU load and together with another problem in RTP it caused no media on incoming SRTP calls approximately every 1000th call.

IP22,IP24,IP28,IP302,IP305: RTP-DTMF not offered when using a/b interface

Status Closed
Id 91905

For example:

\tv=0
\to=- 14 1 IN IP4 10.17.1.91
\ts=-
\tt=0 0
\tm=audio 16414 RTP/AVP 8 0 18 4 97
\tc=IN IP4 10.17.1.91
\ta=rtpmap:97 CLEARMODE/8000
\ta=fmtp:18 annexa=yes
\ta=fmtp:18 annexb=yes
\ta=ptime:20
\ta=silenceSupp:off - - - -
\ta=sendrecv

PBX: MOH URL parameters for parked calls should refer to the parked-to and not parking user

Status Closed
Id 91931

The MOH URL Paramter (%l, %h, %n, ...) can be used to use different MOH Files based on the User who is holding the call. In case of a parked call this should refer to the object where the call is parked, not to the user who has initiated the parking.

IP-DECT: Wrong XML data in Radio endpoint

Status Closed
Id 91933

The XML data in the Radio endpoint is wrong and fixed now. The data is only used for the command show-endpoints, not for the GUI.

PBX Mobility: Call from mobile endpoint to a user with mobility, but without PBX phone was disconnected

Status Closed
Id 91976

This was a collateral damage of

fix: #86813: PBX Mobility: Take precaution against hanging calls because of misbehaving mobile phone or client

SIP/DNS: Wrong port used on secondary SIP server

Status Closed
Id 91995

If SRV query returns 2 hosts with different port, but no IP address in additional records,
SIP starts two A queries for the two host names.
Both resolved IP addresses are combined with the port of the most preferred host of the SRV answer.

IP-DECT: Hold/Retrieve could result in no media for incoming SIP calls with SRTP

Status Closed
Id 92014

This was in fact a H.323 media negotiation problem between master and slave

SIP/WLAN: Keep local Contact-URI up-to-date on subscriptions

Status Closed
Id 92134

Keep local Contact-URI up-to-date on subscriptions (e.g. message summary) when local ip address changes at runtime.

IP-DECT: Hold and Retrieve with SIP and SRTP could result in no media

Status Closed
Id 92189

Problem with forwarding changed SRTP Keys from master to radio

phone: ip241: do headset volume control in digital domain

Status Closed
Id 92203

IP232,IP222,IP241: Reduce flicker when opening details of call list entry

Status Closed
Id 92242

Reduce flicker when opening details of call list entry.

IP-DECT: Potential trap when switching to an from 3pty conference

Status Closed
Id 92262

In case of a very unlikely collision of events, a message was sent to an already deleted object, which cause a restart

IP232,IP222,IP241: Fix for display of international numbers on phone UI

Status Closed
Id 92335

Show international numbers as +4930123456 instead of I4930123456.

PBX: Call Completion to a user without registration, but forking, caused immediate Callback

Status Closed
Id 92341

Instead to call completion request should be rejected in this case

SIP: Huge SIP messages causes out-of-sync on TCP stream

Status Closed
Id 92373

Reading SIP messages from TCP stream gets confused by huge SIP messages.
Presence exchange with external UC was disordered.
Increased size limit from 100KByte to 200KByte.

Gateway: Transmitting FAX documents to receiver with polling mode

Status Closed
Id 92388

It is not possible to send fax documents to a receiver with polling capability. This is fixed now.

IP-DECT: Cipher key index table update function

Status Closed
Id 92467

The cipher key index table is wrongly updated in the Crypto Master if a entry line yet exists. This is fixed now.
The Crypto Master is needed for DECT Security Early Encryption.

IP222 IP232: Handset gains changed to avoid low microphone volume

Status Closed
Id 92577

Low microphones levels were squelched.

Voicemail: Emailing file names with '#'-characters failed

Status Closed
Id 92757

An Escape mechanism wasn't applied

Trap On Ldap DOS Attack

Status Closed
Id 92978

Malign asn.1 content wasn't rejected

IP800 IP6000 IP2000: duplicated DTMF digits in transcoding scenarios

Status Closed
Id 92981

When tandeming VOIP links for trancoding or other purposes DTFM digits were sometimes duplicated.
The RTP carried up to 25ms DTMF remaining DTMF, now its only 16ms.

PBX: Trap if user object is deleted, which is used by other applications (e.g. myPBX)

Status Closed
Id 92985

The applications need to cleanup in this case

IP-DECT: CSV user export

Status Closed
Id 92991

There are some missed users in the CSV user export file. It occurs if there are users with login rights. This is fixed now.

Voicemail: Url Parameter To Control Number Processing

Status Closed
Id 93043

$_leg2tweak
--
Controls <pbx-getcallinfo out-leg2=".."/>
true(default): set leg2 to <ext-nr> from <vm-nr>+<ext-nr>
false: set leg2 according to received divertingInfoLeg2 facility

$_trailhash
--
Controls <pbx-getcallinfo out-cdpn=".."/>
true: pass trailing (en-bloc) '#' into cdpn
false(default): don't pass trailing (en-bloc) '#' into cdpn

http://wiki.innovaphone.com/index.php?title=Howto:Configure_the_innovaphone_Voicemail#URL_Query_String_Variables

IP0010,3010,6010,1060,810: limit number of ethernet packets processed per receive interrupt

Status Closed
Id 93130

this prevents that too much time is spent in ethernet driver in case of broadcast storms or DOS attacks.

IP-DECT: Call transfer with enbloc dailing

Status Closed
Id 93185

Call transfer with enbloc dailing fails. This is fixed now.
This changes also the R-key handling: after dialling a digit for a consultation call the call must disconnect with R-1 like in ring-back state.

SRTP: Remove traces when packet authentication failed

Status Closed
Id 93203

SRTP and SRTCP software encryption produced traces when packet authentication failed. This is not needed, because an event is created anyway, when this happens frequently.

SNMP Walk udpTable, tcpConnTable Could fail

Status Closed
Id 93225

The table index returned wasn't always lexically ascending, causing a walk to stop.

phone: ip222,ip232: sporadic boot time trap when a Jabra LINK 280 adapter is plugged

Status Closed
Id 93261

the initialisation fails with CC=5 on first device descriptor read. after restart of host controller serial_irq() traps in reading the done list.
Happens mostly with upload DRAM.

IP-DECT/Relay: Blocked calls by hidden feature code *5/*7

Status Closed
Id 93272

Outgoing calls with beginning number *5 or *7 are blocked by the feature codes module because of hidden new service codes for an OEM device (#79028). This is fixed now.

PBX: Master Slave license update period 10s instead of 10min

Status Closed
Id 93330

For test purposes the period was reduced to 10s, but by accident this change ended up in version 9 as well

Gateway: CGPN-Maps executed even if the Route did not match in case of enbloc calls

Status Closed
Id 93400

For enbloc calls the CGPN of routes were executed even if the dialed number was incomplete.

PBX: Port License counting wrong, when moving users

Status Closed
Id 93477

When moving a user from a master to a slave, the license count on the master was reduced only after reboot.

PBX: XML error on User Interface if slave name with non-ascii characters used and registration was redirected from such a slave

Status Closed
Id 93483

Conversion of UCS2 as received from the registration to UTF-8 on Web UI was wrong

PBX: Hide connected Endpoint did nor work for forward to other PBX or Gateway object

Status Closed
Id 93513

Diverting Leg Information facilities or Name Identification facilities were still forwarded in this case.

LDAP Client: SearchRequest.derefAliases Changed To neverDerefAliases(0)

Status Closed
Id 94812

Was derefInSearching(1)

Gateway: Header line of FAX documents with big endian devices

Status Closed
Id 94824

The header line of FAX documents with big endian devices was not correctly printed. This is fixed now.


SIP: Port mapping with STUN failed since DNS resolvin of STUN server failed

Status Closed
Id 94876

Port mapping with STUN failed since DNS resolvin of STUN server failed.
Wrong STUN server port was used (5060 instead of 3478).


H.323: No media after blind transfer to Waiting Queue on other PBX, when operator connects

Status Closed
Id 94961

Happened when different coders where used on caller and called side.

SIP: Display names need escaping of " and \\ according to RFC-3261

Status Closed
Id 94966

Acc. to RFC-3261 characters " and \\ (%x22 and %x5C) are to be escaped as "quoted-pair".

PBX Waiting: Presence set for operator was not cleared, on delete or editing of Waiting Queue object

Status Closed
Id 95016

When configuration of the Waitinng object is now changed, any presence set by the Waiting object is cleared.

Gateway: FAX interface interop with non-conforming Fax devices improved

Status Closed
Id 95029

There are Fax devices sending wrong (too long) message initially after being called. In this case it is best handled by ignoring these message and wait for the retry instead of disconnecting the call.

SIP/UDP: Sending response to wrong address and port

Status Closed
Id 95065

Sending response to wrong address and port.
But only if Via header of incoming request contains domain name.

IP241: Headset receiver muffled sound

Status Closed
Id 95164

..

PBX Exec: Secretary availability monitoring did not work with multiple scretaries, with names starting identically

Status Closed
Id 95197

The availability state (secretary booked into the exec primary group) was not associated with the correct secretary. A compare of the names only covered the first half of the name.

Click sounds at caller side when calling another port of same gateway

Status Closed
Id 95436

occured since V9hotfix5

PBX Broadcast: Memory leak when calling busy broadcast object with round-robin config

Status Closed
Id 95439

If all destinations of a broadcast object are busy a name-id facility generates a memory leak.

IP222 IP232 IP241: Use fifo for DSP control channel

Status Closed
Id 95763

try to fix a trap with USB headset

IP232,IP222,IP241: Fix for call status display

Status Closed
Id 95769

If a call was remotely disconneced during HOLD,
the held phone was constantly displaying "held" instead of "disconnected".

SIP: Interworking issue with "LifeSize Passport/LS_PP1_4.11.9 (8)"

Status Closed
Id 95785

Problems decoding large and complex SDP offer from LifeSize.

802.1x (EAPOL) did not work on interfaces configured for VLAN

Status Closed
Id 95931

802.1x (EAPOL) frames received without a VLAN tag must always be passed to the protocol module, even if the interface is confiured to use VLAN

V9 Hotfix 22 (9061240)

Changes included in Version 9 hotfix22 Definition

New Features

PBX: New option for RTP Proxy - proxy only if different registration address

Status Closed
Id 88439

To avaoid RTP Proxy for two endpoints located within same private network behind NAT

PBX Executive: Allow monitoring of availability of secondary secretary, don't treat Exec as secretary

Status Closed
Id 95497

With these two additions a configuration with two executives and two secretaries, each secretary being primary to one executive an secondary to other can be configured with a single group for each secretary and both executives can monitor the availability of both secretaries.

SIP: Workaround for buggy registrar

Status Closed
Id 96313

Workaround for buggy registrar.
Different expirtes values in Contact header and Expires header.

\tSIP/2.0 200 OK
\tVia: SIP/2.0/UDP x.x.x.x:2069;rport=2069;branch=z9hG4bK-CDDD130C
\tTo: ;tag=5b8729d5-6f6353c4-cbsxz
\tFrom: <sip:38795988@193.90.37.3>;tag=2109370043;epid=0090331e0bef
\tCall-ID: 7ef63c56e909d311b3890090331e0bef@95.130.221.205
\tCSeq: 1004 REGISTER
\tExpires: 60
\tContact: <sip:38795988@x.x.x.x:2069;transport=UDP>;expires=300
\tUser-Agent: ZTE-SBC
\tX-ZTE-Cause: "SBC-4721-2002"
\tContent-Length: 0

Better apply the smaller expires value.

PBX: Send forking calls as diverted calls

Status Closed
Id 96370

So that on the called side, it will be displayed who forked the call the same way as a diverted call is displayed.

PBX: pbx_makecall.txt not only for mobility, but for Waiting Queue also

Status Closed
Id 96384

Allows to initiate call with simple web request from a Waiting Queue

IP-DECT: Physical location was wrong after logout/login on handset

Status Closed
Id 96392

The physical location information is based on the redirection of the registration from the PBX at the physical location to the registration PBX. Some information was not cleared with the logout, so re-registration startet with the registration PBX right away.

phone: if a number to be dialled contains a comma, the digits following the comma are sent as DTMF tones after connect

Status Closed
Id 96402

This applies to all numbers dialed en bloc, i.e. numbers dialed via indirect dialing, a phone directory or a function key. The comma must not be the first character of the number.

IP-DECT: OEM PBX type info in GUI data

Status Closed
Id 96723

Now the GUI data includes the type info of an OEM PBX.

IP22 IP24 Ip28 IP302: Don't complete media negotiation for ab-interfaces if no media can be sent

Status Closed
Id 96773

For incoming calls to a phone media negotiation was already completed during ringing, so that when going off hook the media channel was already established. This causes interop problems, because there are endpoints which asssume there is inband info (e.g. ringback) if media negotiation is complete so local tones (e.g. ringback) were turned off.

In the past with slowstart this premature media negotiation was usefull to avoid delayed media after off-hook. With SIP or H.323 faststart there is no use anymore.

Needed to avoid that the DSP send CLIP and tones at the same time, which can cause sporadic DSP failures.

SIP: New interop tweak /register-interval

Status Closed
Id 97834

New config file option /register-interval 60
Problem is too weired to explain.
This option can be used to set the REGISTER interval to a fixed value regardless of the negotiation.

Bug Fixes

PBX: URI dialing, should not be case sensitive and numbers should be possible

Status Closed
Id 89326

needed for federation

IP222 IP232 IP241: LCD display is after softreset sometimes out of sync

Status Closed
Id 95866

Displaycontroller needs to be stopped before restart.

Httpclient: Problems with HTTPS URLs

Status Closed
Id 96099

Shortcut to local file I/O did not work fot HTTPS URLs.

phone: do not report "No Media Data received" errors for connections to a recording device

Status Closed
Id 96102

some recoding devices, for example ASC never send data on a recording connection

IP-DECT: Possible no media in case of media renegotiation after handover

Status Closed
Id 96124

The problem happened in about 50% of the cases of a media renegotiation which results in a different coder after handover.

IP-DECT: Show release state in IP1202

Status Closed
Id 96157

The release state is not shown in the IP1202. This is fixed now.

SIP: Must not answer "refresher=uac" if request contains "refresher=uas"

Status Closed
Id 96167

Interop issue with Genband C20 PBX.

Must not answer
Session-Expires: 400;refresher=uac
in 200/OK, if INVITE contains
Session-Expires: 400;refresher=uas

PBX Trunk: "Outgoing Calls restricted" did not work correctly, Presentation restricted was set, but number could be wrong

Status Closed
Id 96200

For example if an analog Gateway was registered to a PBX user, and this Gateway did not send a Calling Party Number with the call, the call was sent with Presentation restricted, but without digits. This could affect Billing Applications which are based on CDRs from the Gateway.

phone: ip222,ip232: inbound calls automatically connected to Plantronics Savi W440/740/745 headsets with new firmware Versions

Status Closed
Id 96276

reported for:
- Savi W440 with firmware 0118 on USB/DECT Dongle D100
- Savi W740/745 with firmware 0115

reason:
the newer firmware versions reject truncated output reports (no trailing 0 bytes) with STALL. The error handling for this case was wrong and caused an autoconnect.

IP-DECT: Wrong GK id of standby Master to Mobility Master

Status Closed
Id 96302

The standby Master uses a wrong gatekeeper id to register to the Mobility Master. This is fixed now.

PBX-CDRs: Conference ID missing in CDRs created by Mobility data callback/callthru

Status Closed
Id 96399

This caused these calls not to show up in the reporting

PBX: Unexpected behaviour if too many filter were configured

Status Closed
Id 96416

No new filters were accepted without error message

IP22 IP24 IP28 IP302 IP305: Sporadic DSP host interface overruns

Status Closed
Id 96606

On the small gateways the DSP hangs if control packet on host interface arrive too fast. Now the rate is limited.

License download not working

Status Closed
Id 96622

License download stops.
Licenses are not downloaded.

PBX: Changing of PBX Object Name did not change the name to be used for registration

Status Closed
Id 96634

So if the name of the PBX registering as slave was changed as well, it did not register anymore. The PBX object had to be deleted and created with new name.

Gateway: FAX interface on IPVA

Status Closed
Id 96660

The FAX interface on IPVA can not connect to a remote device because of wrong protocol events. This is fixed now.

SRTP: One way audio after some minutes on IP6000 IP2000 IP6010 IP0010

Status Closed
Id 96673

Applies to v9hotfix21 on IP6000, IP2000, IP6010, and IP0010.

Collateral damage from fix #91892: SRTP: Avoid one-way media with high start sequence numbers.

Gateway: CGPN-Maps executed even if the Routing was already completed

Status Closed
Id 96685

A CGPN map, which was configured in a route following the route, which was actually executed was executed as well.

This was a collateral damage from fix: #93400: Gateway: CGPN-Maps executed even if the Route did not match in case of enbloc calls

IP22 IP24 IP28 IP302 IP305: Sporadic DSP host interface overruns - CLIP disabled, new trace option

Status Closed
Id 96716

On the small gateways the DSP hangs in some conditions.
Now the trace-stop is replaced with an Assert to recover from this situation.

To get a trace of this condition a new trace option is added at dsp.xsl, called txt-trace. This traces the DSP message as text, so that they can be read out after a trap.
Typical usage is to enable DSP-trace, DSP control messages DSP data messages and DSP txt trace.
DSP pcm trace and DSP T38 trace should be off to avoid excessive debug load.


Also, the CLIP messages are disabled since they caused problems in the past.


H.323: One-way-voice if SRTP call to a Waiting queue is forwarded via Waiting Queue Maps to a phone

Status Closed
Id 96721

This is a problem with SRTP key exchange, which could happen in other traffic scenarios as well.

IP-DECT: Web UI administrator user list removed on IP1202

Status Closed
Id 96726

The Web UI administrator user list is removed on the IP1202 now.

IP28: Click sounds at caller side when calling ip28 gateway

Status Closed
Id 96782

IP28: Sometime Ringing stopped working on an anlog port. Worked again only after reset.

Status Closed
Id 96787

This happened due to low ringing volatge, the default value of "low" on the interface configuration for the ringing voltage caused this to happen.

IP22 IP24 IP28 IP302 IP305: Sporadic DSP host interface overruns - Updated DSP code

Status Closed
Id 96852

Clip enabled on all channel.

phone: ip222,ip232,ip241: Local Network Coder default for User-2..6 was G711 instead of G722 as for User-1

Status Closed
Id 96890

WEB-Interface "Phone/User-2..6/General/Options/Local Network Coder" was preset to G711 instead to G722 as for User-1.

V9 Hotfix 23 (9061252)

Changes included in Version 9 hotfix23 Definition

New Features

PBX Waiting: timeout argument for pbx_makecall.txt URL

Status Closed
Id 97010

A timeout argument was added to the PBX0/ADMIN/pbx_makecall.txt URL to cancel an outgoing call initiated with this URL.

ISDN: Send legacy Redirecting Number for 'old' Fax Servers

Status Closed
Id 97615

The redirecting number is an old style information element, which contains part of the information as the diverting leg2 facility. Some Fax Servers do not understand the leg2 facility.

PBX-CDRs: Better CDR for pickup

Status Closed
Id 97681

A pickup was not indicated in the CDRs

PBX-SOAP: Status of Boolean object indicated as local number

Status Closed
Id 97738

When monitoring a Boolean object with SOAP a call is indicated. The local number of this call is set based on the status of the boolean object (00 automatic-off, 01, automatic-on, 10 - manual-off, 11 - manual-on)

PBX-SOAP: UserPark allows to park to another object

Status Closed
Id 97741

The argument 'cn' was ignored in v9 and earlier, now it can be used to identify a destination for the park.

Bug Fixes

SUBSCRIBE for MWI not correctly handled after change of IP address

Status Closed
Id 96898

SUBSCRIBE for MWI not correctly handled after change of IP address

phone: ip222,ip232: USB headset echo effects in call setup phase when a G722 call is started by a CTI application

Status Closed
Id 96974

The caller hears the calling tones and it's own speech from the headset microphone but the callers speech is not transmitted to the called party.

H.323: Offered packetization should be honored for SIP interoperability

Status Closed
Id 96983

Problem happend with calls from Samwin CBC

RTP-DTMF: Must increase duration field when sending RTP-Event with END marker

Status Closed
Id 97001

Must increase duration field when sending RTP-Event with END marker to comply with RFC.

ASN.1 BER: Decoding of Sequence Member with indefinite length failed

Status Closed
Id 97072

An Avaya supplementary service couldn't be decoded

H.323: Channel Close sometimes not sent on hold

Status Closed
Id 97080

Esspecially on the second hold within a call the Channel Close was not sent to the party, which put the other on hold. This caused the channel not beeing turned off on this side (the other side receives music on hold in this case)

IP4 did not work anymore when IP6 was disabled via WEB interface

Status Closed
Id 97111

When the "IP6/ETNx/IP6/Options/Disabled" was checked the IP4 operations did stop after a while.

phone: ip222,ip232: Plantronics Savi W440 dosn't report Talk-Key events in a call established at phone or by a CTI application

Status Closed
Id 97115

When a call via this headset was initiated/accepted by the Redial-Key, the Headset(Mode:Control) function key or a CTI application, the call could not be disconnected by pressing the Talk-Key at the headset because the Headset did not report this action.


phone: DTMF digits following a comma in a number to be dialed were not handled correctly in some cases

Status Closed
Id 97150

- in the "Destination Number" configured under "Phone/Direct Dialing" in conjunction with a nonzero "Autodial Timeout": the DTMF digits were sent as dial digits
- with a nonzereo "Enblock Dialing Timeout" configured under "Phone/User x/General/Options": sending of DTMF digits was delayed by the configured timeout\t

Gateway: FAX interface on IP800/IP305/IP302

Status Closed
Id 97571

Fix for the last fix #96660.
The FAX interface on the IP800/IP305/IP302 can not connect to a remote device because of wrong protocol events. This is fixed now.


H.323: Fast Unregister/Register operations could lead to failed registrations, in case of fixed signaling ports

Status Closed
Id 97637

A listening socket could still be in use. Only happened with IP-DECT and multi-master.

PBX: Name-Id of busy destination was not forwarded to other PBX

Status Closed
Id 97646

Display was different when calling a busy phone on local PBX or on another PBX

IP22 IP24 IP28 IP302 IP305: ASSERT on DSP queue overrun added

Status Closed
Id 97653

phone: "Prepare Override" function key did not work since V9hotfix21, the overriding source address was ignored

Status Closed
Id 97665

SIP: Trap when terminating a call while re-negotiation is ongoing

Status Closed
Id 97675

Trap when terminating a call while re-INVITE is pending.

phone: ip230,ip240,ip241 : unefined codes received from a DHSG Headset basestation were misinterpreted as Hookswitch indication

Status Closed
Id 97732

sometimes DHSG Headset basestations send codes not defined for DHSG which are silently discarded now.

H.323: Potential Trap when reconfiguring an H.323 registration

Status Closed
Id 97820

This is a collateral damage from

91815: IP-DECT: Static ports between Master and Radio

PBX Executive: Calls with calling id restriction and without calling id, were sent to executive, even if secretary available

Status Closed
Id 97897

This happend for calls coming in from public ISDN with calling id presentation restriction thru a trunk object without number

SIP: Trap when using STUN

Status Closed
Id 97898

Trap when using STUN.

SIP: Dialog-Info was encoded with wrong state attribute

Status Closed
Id 97926

Dialog-Info was encoded with wrong state attribute "full".
Must be "partial".

SIP: Logging was wrong

Status Closed
Id 98037

Logging was wrong.

PBX: Potential trap when receiving unknown presence activity

Status Closed
Id 98043

In the respective version unknown activities are mapped to "busy"

V9 Hotfix 24

Changes included in Version 9 hotfix24 Definition

New Features

phone: ip222, ip232: support Jabra UC Voice 750 and Jabra Speak 510

Status To-decide
Id 98092

IPVA: ETHx Transmit Queue Size Limited To 1MB

Status Closed
Id 98187

Was unlimited

Linux: Shutdown warning message

Status Closed
Id 98438

If Linux is running, a shutdown warning message is shown at the Linux General page now.

SIP/TLS: Using domain name as fall back to proxy name when comparing to the certificat "subject"

Status To-decide
Id 98643

Using domain name as fall back to proxy name when comparing to the certificat "subject".

Voicemail: Allow SMTP Email Port Different Than tcp/25

Status Closed
Id 98683

eg in email.xml: ..smtp.foo.bar:888..

phone: directory search highlights first matching entry if the search expression contains any non numeric digit

Status To-decide
Id 98806

To prevent unintended dialing of a directory entry starting with numeric digits the search expression was checked if it consists of dialable digits (0-9*#,) only.
In this case the first matching entry was not automatically highlighted (activated) so that the input (number) could be dialled by going off-hook.
Now the check includes the numeric digits (0-9) only.

SIP: New interop tweak /accept_any_reg_interval

Status To-decide
Id 98887

For endpoints that do not read the expires value from 200/OK response to REGISTER.
E.g. "User-Agent: TRBOnet.Enterprise"

phone: ip222,ip232: support Plantronics Blackwire C320, Blackwire C720, Voyager Legend UC, Calisto 620

Status To-decide
Id 99112

Bug Fixes

IPVA: Unused ETH1 Could Cause Out-Of-Memory Situation

Status Closed
Id 98179

Outgoing packets could queue up without ever getting purged.

SIP: Trap - not checking array boiundaries

Status To-decide
Id 98219

Trap - not checking array boiundaries when processing User-Agent header.

SIP: STUN not working

Status To-decide
Id 98223

STUN not working if STUN server IP address is configured.
No problem if if STUN domain name is configured.

phone: ip222,ip232: USB connection sometimes lost until reboot

Status To-decide
Id 98290

Observerved with with headsets from various manufacturers. Most probably caused by strong electrostatic discharges to the USB connection cable. In such case also unplugging/plugging of the headset was not noticed by the driver anymore.

Logging: "Alarm and Event Forward Server" address could not be changed anymore once configured

Status To-decide
Id 98457

LDAP Replication: ASN.1 Decoding Failed For More Than 100 Attributes Per Object

Status Closed
Id 98468

Happend with 100 attributes in a PBX stored phone configuration.

Corrupt trap buffer when tracing UART messages

Status Closed
Id 98533

IP232,IP222: Confirming blind transfer with redial key did not work

Status To-decide
Id 98591

Short user guide:

Press 'redial' before accepting the call, enter telephone
number for the call diversion and confirm by pressing 'redial' again.

phone: ip222,ip232: USB headset media connection lost after a release received from a remote conference peer

Status To-decide
Id 98600

Happened only on a release of the call which was the active call when the conference was established. The remaining VOIP connection was OK but the media stream was not passed from/to headset anymore.

IP241: DHSG Headset messages are sometimes wrong

Status Closed
Id 98642



Beim Neuladen des Sequencers bei Änderung der LCD-Helligkeit kommt der UART-Takt gelegentlich zu schnell. Besser noch wäre 2 sequencerprogramme zu definiere, aber wie das geht ist nicht im Orchid/Titan Usermanual nicht offensichtlich -->

SIP: SDP version not increased when answering an offer where only media-mode has changed

Status Closed
Id 98739

If remote side changes from 'sendrecv' to 'inactive'
the SDP answer follows this change of media-mode,
but SDP version was not increased.

Gatway: Configuration of Blockdial Timeout at Routes did not work

Status Closed
Id 98748

Wrong value was calculated, if multiple maps were used in a single route blockdial timeout configuration of a map was lost, when another map was configured.

enabled state of an external directory configured via a PBX config template was lost in some cases,

Status To-decide
Id 98816

This did happen for example when
- a second phone was registered to the same PBX user (twin phone)
- a "Phone/Reset/Reset User Specific Configuration" was done via the phones WEB GUI (but not when this was done via the PBX GUI)

SIP: Trap when sending <dialog-info>

Status To-decide
Id 98902

Trap when interworking group-indications into dialog-info.

Memory Leak when deleting voicemails

Status Closed
Id 98929

A list wasn't cleaned

IP222 IP232: Noise in the microphone at 6400Hz

Status To-decide
Id 98941

.

IP222 IP232 IP241: Codec register debugs added

Status Assigned
Id 98972

Codec register debugs are enabled with dsp trace.
Codec register are dumped after changes to analyse Manits 97903

PBX: No config updates were sent to the phones, when selection of templates was changed

Status Closed
Id 98983

When the config of the template itself was changed an update was sent, but if it was changed which templates were used on a user object, no update was sent.

Phone: Could not configure fkey labels containing single quotation mark

Status To-decide
Id 98986

Could not configure fkey labels containing single quotation mark.

myPBX: Missed calls for mobility calls accepted somewhere else

Status Closed
Id 98995

A cause code was missing in CDRs generated by mobility

SIP: Switch-over to t38 did not work in one configuration scenario

Status To-decide
Id 99130

Interworking of SIP and H.323.
Switch-over to t38 did not work if both Gateway interfaces were configured to media-relay with exclusive audio codec.

phone: ip222,ip232: ignore HID function of USB headset charging cables

Status To-decide
Id 99133

Some USB headset charging cables present a HID function as long as the headset is connected to the cable (probably used for headset firmware updates).
Dependent on the enumeration sequence the HID function of the cable could hide the HID function of the headset when the headset base station (or the bluetooth or DECT dongle) is plugged in paralll to the phone.

IP22 IP24 IP28 IP302 IP305: ASSERT on DSP queue overrun added (2)

Status Assigned
Id 99155
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