Da erscheint in V9 bei Auswahl von PPPOE das ISDN Menu, dafr ist bei <none> jetzt kein weiterfhrendes Men sichtbar. Das sollte auch bei V8 so sein.<br/><br/>Bei V8 erscheint derzeit auch bei <none> das ISDN Menu, dafr ist bei V8 das PPPOE Men korrekt vorhanden wenn man PPPOE0/1 als Schnittstelle auswhlt.<br/>
== Fixed comparison of FTY_IM_MESSAGE in unit tests ==
In fty_event_im_message::is_identical fty_event_im_message::data was handled as a null terminated string but it is a bufman buffer.<br/><!--<br/>test_fty.cpp<br/>test_engine.cpp<br/>test.h<br/>-->
== SRTP: Better diagnostics for SRTP on IP6000 IP2000 ==
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).
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
== Phone: long function key titles hide idle screen information ==
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/>
== PBX Waiting: Call forwarded with DTMF mapping was shown in myPBX for each registration ==
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.
== phone_orchid: microphone is not mute on a call intrusion in silent monitoring mode / microphone cannot be muted in a conference ==
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.
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/>
== PBX: License accounting did not work with Unknown Registrations under some special conditions ==
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 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 ==
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 ==
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/>
== H.323: A name_id of length 0 resulted in invalid H.450 coding ==
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/>
== SIP: Unwanted media-relay sessions when using forking/broadcast/multi-reg ==
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
== IP30x, IP1060, IP2010, IP6010: Fax did not work if rerouted from ISDN interface to a Voip destination ==
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 ==
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.
== SIP: Generate new SRTP key on every incoming re-negotiation ==
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 ==
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.)
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/>
== Phone: Switch presence fkey light on if presence activity is set ==
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 ==
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.
== myPBX: Hide passwords for application sharing and reporting in config ==
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.
== WEB GUI page cannot be scrolled completely when height of left hand logo is too big ==
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.
== X509: Fix for reading innovaphone info from flash ==
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.
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
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.
== SIP/TCP: Transport error when connection is closed by client ==
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.
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
== Denial of Service filter in ethernet library did not work ==
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 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 ==
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
== PBX: Translation of Cause "Call Rejected" to Cause "User Busy" for endpoint objects only ==
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 Trunk/Gateway: Round robin within registrations to same device, different devices sequentially ==
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
== IP-DECT: Configuration XML data for OEM device ==
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 ==
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/>
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.
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.
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/>
== HTTP-Client: Allow user names longer than 16 characters ==
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/>
== IP-DECT: Allow setting empty text for idle display ==
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.
== RTP-DTMF: Start handling of RTP-DTMF on reception of END event ==
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/>
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 ==
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 ==
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.
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"
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 ==
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.
== PBX: Reject calls without media, if no known facility ==
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)
== PBX Waiting: In case of "Announcemen w/o Connect" together with Alert-Timeout 0, DTMF dialing was not possible ==
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.
== SIP: Send BYE with Reason header with "Q.850 Recovery on timer expiry" ==
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/>
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.
== IPxx10-sata: trap after config /trace /track activation ==
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 ==
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.
== phone_orchid: Calls received with CLIR appear in call list with an empty entry ==
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: Take SDP id and version as 64bit integer ==
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) ==
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/>
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
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/>
== Improved protection against Denial of Service attacks ==
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 ==
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
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/>
== auto complete dtmf feature codes with '#' after 2 seconds ==
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 '#'.
== PBX: HTTP request to initiate call for mobile phone ==
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/>
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.
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 ==
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/>
== H.323: Renegotiation to Fax did not work under some conditions ==
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
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/>
== IP222: Cannot leave menu screen with ESC when IP address ctrl was active ==
-Clear IRQ in handle-interrupt after os_interrupt is too late, since IRQïs 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/>
On "Unrestricted Digital Information" only CLEARMODE is offered (no audio codecs).<br/>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 ==
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/>
== when editing a phone config template the dialing location inherited from a predecessor template was stored in the edited templat ==
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 ==
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.
== CX0 Wave-Encoding Not Working If Fact-Chunk Present In Header ==
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) ==
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/>
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.
== H.323 Signaling Timeout event was generated for 'normal' RAS re-transmissions ==
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 ==
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 ==
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 ==
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
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/>
== PBX: Tooltip on "PBX/Config/Log Calls" checkmark wrong ==
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'.
== PBX: Blind transfer with consultation to BC-Conference failed ==
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/>
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 ==
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.
== SIP: Trap on timer expiration during call release ==
A waiting means a call which was received while another call was active and the active call being disconnected and the waiting call not yet sent, or a call waiting for recall.
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)
== SIP: Diagnostics of transport error "SIP Overload" ==
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.
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 ==
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.
== Scheduling improved to avoid processes not being scheduled during long flashman operations ==
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.
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".
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.
== myPBX: Name and Number Display not correct on IM sessions across PBXs ==
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
== SoftwarePhone: Trap and media channel after conference ==
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.
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.
== PBX Config Templates: When editing config Templates wrong inherited values are displayed ==
A config Template inherits only from the 4 other Templates configured in it directly, whereas any other object inherits from the Config Template configured with it and the four other Templates this is refering to.<br/>
== ip241 - monitor mode (handset + speaker) did not work in V9hotfix5 ==
When ,, are encoded as a,o,u followed by diaeresis from unicode block 'Combining Diacritical Marks' only a,o,u where displayed.<br/>Now ,, are displayed.
== IP241,IP222,IP232: Support for hebrew and arabic presence notes ==
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 ==
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.
== IP241,IP222,IP232: Show number of missed calls in status bar ==
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.
* Neuer DSP code von Audiocodes ( patch Version 660 02 mit AMR_WB ).<br/>* Kein StrongEC mehr ntig<br/>* Codec PCM Bus luft mit 16KHz<br/>* Wideband Codecs zur Zeit disabled<br/>* Konferenz erfordert noch Anpassung im Treiber
Die IP222/IP232 wird gelegentlich als zu laut empfunden.<br/>Die Prozentanzeige mit den krummen Balken sieht "komisch" aus.<br/>0% (Slider ganz links) wre gut.<br/>Mehr Werte wren gut.
''Status:''
ac_codec3.cpp<br/>ac_codec3.h<br/><br/>
== IP232: Cursor positioning when touching ip address control ==
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 ==
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: Using wrong remote port when registering ==
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/>
== myPBX: Forward invitation links to incoming chat calls that are added to a chat conference ==
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.
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.
== PBX: Forward original received ISDN display element to picking up or forwarded call ==
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 ==
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.
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/>
== myPBX: Allow asterisk as a wildcard in LDAP search strings ==
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.
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.
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 (Lauthren): Level too low ==
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
== IP241,IP222,IP232: Could not activate first builtin background image from phone menu ==
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)
== myPBX launcher: Unhandled exception when accessing browser object ==
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 ==
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 Waiting: Diverting leg1 info not correct when diverting to a Waiting Queue ==
Statement <pbx-upd-obj type="cfu"..> failed to work properly after being used for diversion manipulation multiple times within a single script session.
== SIP: No interworking of "Q.931 CALL PROCEEDING" into "183 Session Progress" ==
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 ==
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.
== IP241,IP222,IP232: Call duration display wraps after 100 minutes ==
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.
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.
== IP232: Hiding touch keyboard by touching a control ==
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.
== IP241,IP222,IP232: Replace triangle by arrow to display diversion/transfer information ==
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
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
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
== reset could fail, if issued during firmware update ==
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.
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 ==
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.
== LDAP-Expert, Edit Object Dialog: Could show more than one object ==
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/>
== PBX: Tracing flag turns on tracing in all dyn PBX's as well ==
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"
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)
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 ==
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 ==
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
== H.323: Media Negotiation problem with transfer in Gateway (not PBX) ==
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 ==
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 ==
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 ==
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.
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.
== IP241,IP222,IP232: Wrong display name in call list ==
Name identification of remote party is to be displayed,<br/>but name identification of diverting party was displayed instead<br/>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 ==
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
== phone_orchid: remaining call mute after remote relase for the call which was the active call at start of a conference ==
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.
== IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax failure after transfer ==
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.
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
== phone: ip222, ip232: USB headset support - Plantronics C420, GN2000 USB - MS OC Version ==
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
== show linux shutdown warning on firmware reset page ==
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.
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.
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.
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
== H.323/SIP: Avoid delayed SDP within outgoing calls as far as possible ==
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'
== Media Relay: Don't terminate T.38 protocol in media relay, forward transparently ==
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.
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.
== SIP: REFER does not work as expected on Gateway interfaces without registration ==
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&type=bug<br/><br/><!--<br/>voip.h<br/>gk.cpp/h<br/>sip.cpp<br/>relay_edit_voip.xsl<br/>-->
== IP-DECT: Better handling when receiving a call, while a previous call is being released ==
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/>-->
== Edss1 Interworking: Sending divertingLegInformation2 as PublicPartyNumber(was UnknownPartyNumber) ==
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 ==
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/>-->
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/>-->
== PBX: Web User interface problem with IE and non-ascii PBX names ==
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/>-->
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/>-->
== SIP: Re-negotiation for T38 did not work in media-relay scenarios ==
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/>
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.
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/>
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/>
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/>-->
== IP-DECT: Three party conference with innovaphone PBX ==
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/>-->
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/>-->
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 Waiting: Filter did not work anymore for CFNR ==
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: Status displayed on boolean function key could be wrong ==
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/>-->
== DHCP Server Identifier was cleared after editing the DHCP-Server page ==
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/>
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/>-->
== Config: Could not dynamically set or reset /trace on the LICENSE module ==
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/>-->
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/>-->
== IP241,IP222,IP232: Show lengthy number information on Partner fkey ==
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/>-->
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/>-->
== SIP: Domain Name System (DNS) names compared case sensitive ==
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/>-->
== SIP: Problems with DNS resolving of proxy adresses ==
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/>-->
== PBX: When editing a Node object it was changed to a PBX object ==
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/>-->
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/>-->
== Name and Number configured for a registration must be completely replaced by registration result ==
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
== PBX: Possible trap on calls from misconfigured nodes/PBXs (node parent loop) ==
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/>-->
``Survivabilityïï 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
== AC-DSP3: Switch trace off if the DSP Host interface shows an error ==
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/>-->
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/>-->
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/>-->
== Phones: Show 'tel' presence on configuration screen ==
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/>-->
== PBX-Waiting: Calls to a Waiting queue object in altering state could not be cleared with SOAP ==
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/>
== IP210 IP230 IP240: Handset receiver volume increased, especially at lower frequencies ==
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/>-->
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/>-->
== IP232: Invalid text on second line of fkey label ==
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/>-->
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/>-->
== AD Replication: LDAP filter encoding failed, when Poll Timer was configured ==
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/>-->
== Gateway: Routing of incoming SIP calls may not work ==
... 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: Memory leak when closing signaling interface ==
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/>-->
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/>-->
== myPBX: Syslog for sessions and application sharing ==
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/>-->
== SIP: Pass display names of <dialog-info> to phoneapp ==
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/>-->
== IP-DECT: OEM Configuration of Cisco Features changed ==
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/>
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/>-->
== IP-DECT: New Master/Radio behavior for license incompatibility ==
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/>-->
== PBX: Better handling of presence information without activity ==
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!
== SIP: Support for Alert-info:<Bellcore-dr1> ==
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/>
== phone: ip222, ip232: Jabra USB Headset feature "Reject incoming call" supported now ==
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/>
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/>-->
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/>-->
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/>-->
== SIP: New config option /take-zero-addr-for-hold ==
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/>-->
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/>-->
== SIP: Pass display information to application ==
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/>-->
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/>
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/>-->
- 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/* -->
== IP222 IP232 IP241: repeated ethernet link status 1000M wrong ==
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/>-->
== PBX-SOAP: UserRedirect - original called number got lost ==
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/>-->
== SIP: Send 200/OK for MESSAGE(text/plain) when accepted by application ==
XML element content requires some resevered characters to be escaped (<>).<br/>These escape sequences (&gt; or &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/>-->
== IP241,IP222,IP232: Two status symbols may overlay each other ==
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/>-->
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/>-->
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/>-->
The configuration format of an OEM configuration option was changed for the backward compatibility.<br/><br/><!--<br/>Changed file:<br/>dectmaster.cpp<br/>-->
== phone_orchid: micro mute when a waiting call was connected after an outbound call setup had been cancelled ==
- 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 -->
== SIP: Pass display information to application ==
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/>
== X.509: Creating certificate containing IPv4 address did not work ==
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/>-->
== ip222,ip232: waiting calls mute after having been accepted/connected on USB headsets, mainly seen with Jabra LINK 14201-30 ==
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: Objects list filter for numbers did not work correctly anymore ==
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: Pass display information to application ==
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/>-->
== IP-DECT: System GUI, disabled local coder options ==
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/>-->
== IP-DECT: System settings not to dynamically connected radios ==
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/>-->
== Support for old versions of Jabra BIZ 2400 headset with 48 kHz output sampling frequency ==
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.* -->
== SIP: Support for multiple audio media descriptions ==
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 ==
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 -->
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/>-->
== SIP: New config option for endpoints not refreshing their registration during call ==
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/>-->
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/>-->
== SIP: New config option /no-certificate-check ==
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/>-->
== IP-DECT: Own priority for idle display set by messages ==
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/>-->
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 -->
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/>
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/>
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/>-->
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/>-->
== IP-DECT: No delay with semi-attended call transfer and SIP ==
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/>-->
== IP-DECT: Display info with failed user log-in ==
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/>-->
== IP-DECT: Hanging aborted semi-attended call transfers with SIP ==
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/>-->
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/>-->
== DHCP: Name registration at WINS was not refreshed as requested by TTL in registration response ==
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 -->
== Phones: Presence info during ringing state may show garbage data ==
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/>
== myPBX: Support contact names containing a single quote ==
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/>-->
== PBX: RTP-DTMF was disabled by Voice Mail object during re-negotiation ==
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/>
== ipv6: memory leak when sending fragmented packets ==
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/>-->
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/>-->
== phone_inca: when a call completion was set up with CLIR active the called party was not displayed on a possible recall ==
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/>
== IP-DECT: Display info with remote control call ==
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/>
== IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: reduce probability of false DTMF detection ==
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/>-->
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 -->
== Gateway: Routing problem with blockdial route and following matching non-blockdial incomplete routes ==
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"
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 nderung 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 -->
== IP222 IP232 IP241: update to DSP code 660.27 ==
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 -->
== phone: call forwarding options offered even when call forwarding was not possible ==
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 -->
== PBX Mobility: Trap in case of Transfer of a call from a mobile endpoint to another mobile endpoint ==
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
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/>-->
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 -->
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/>-->
== IP3010 IP810: number of DSP channels and number of conference channels was wrong. ==
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 nderung 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&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 <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/>-->
== IP222,IP232: Going offhook in call list always dials first list entry (not touched list entry) ==
After a Firmware or bootcode download via Maintenance/Download the associated command processor instance was not released.<br/><br/><!-- command.cpp -->
== IP-DECT: Call transfer compatibility with SIP ==
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: Trap if multiple calls arrive at same time ==
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/>
== PBX Waiting: Evaluate Busy on ... Calls for calls to an operator ==
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/>-->
== PBX Broadcast: No diverting name sent with broadcasted call ==
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/>-->
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 -->
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 -->
== PBX Routing: Node extern did not work for calls from a trunk marked as local object ==
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/>-->
== External-UC: Presence info assigned to wrong PBX object ==
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/>-->
== PBX: Standyby PBX generated alarms for missing slave registrations, even if active PBX up ==
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/>-->
== SIP: Problems with presence signaling on External-UC link ==
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/>
== IP-DECT: Support for more than 32 LDAP attributes ==
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/>-->
== Gateway: On IP24, IP302, IP305 it could happen that multiple routes disappeared if a route was deleted ==
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 -->
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 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/>-->
== IP-DECT: No fall-back after unattended call transfer ==
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/>-->
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/>-->
== phone: ip222,ip232: the variable KEYS0/HID-MAP permits to map new USB headsets to builtin descriptors ==
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 -->
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/>-->
== myPBX: Show version of launcher in the list of sessions ==
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
== H.323: Support for Avaya SRTP with AES128/80 ==
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/>-->
== PBX: Description was missing for DECT System object ==
"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/>
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/>-->
== PBX Waiting: Trap on leak-check if dtmf maps are configured ==
..<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 -->
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/>
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/>-->
== HTTPCLIENT: Allow configuration of a http authenticated URL even if the server needs no authentication ==
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/>
== PBX Waiting: Trap on collision of operator connect and two-stage dialing ==
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/>-->
== PBX: Call to a Trunk/Gateway was not marked correctly as external, if no connected number was received ==
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/>-->
== HTTP: Chunked transfer fails if the last 2 bytes of the chunk header are in the next tcp packet ==
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/>-->
== PBX: Partial Rerouting was prohibited in Alerting State (CFNR) ==
The vertical resolutions 96 lpi, 196 lpi and 400 lpi are correctly saved in the SFF file with a proprietary definition known by the tool sfftobmp.<br/><br/><!--<br/>Changed files:<br/>facsimile_sff_def.h<br/>facsimile_g3.h<br/>facsimile_g3.cpp<br/>facsimile_ECM.h<br/>t30.cpp<br/>--><br/>
== phone: provide complete dialog info to a phoneapp ==
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/>-->
== IP222 IP232 IP241: Force same speed of the switch ports for 1000M/100M scenarios (configuration option added) ==
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/>
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/>-->
== http client : authentication was not retried after a failure when the offending request was repeated in the same session ==
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 -->
== IP6000: Prevent blinking error LED on old IP6000 with HW-Build <110 ==
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 data is processed in chunks. If the chunk border was right behind the closing </user> tag, the decoding of the next chunk failed.<br/><!--<br/>pbx_admin.cpp<br/>-->
== phone: when scrolling directory search results sometimes one of the numbers of a contact was not displayed ==
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 -->
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/>
== Hide LDAP Server Password For Viewer Accounts ==
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-->
== phone: ip222, ip232: status stage added to all USB control transfers ==
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/>-->
== NAT: Don't forward DNS requests from public network ==
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/>-->
== H.323: Media Negotiation did not work for call with reverse media and media response in CALL-PROC ==
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/>-->
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/>-->
== SIP: Loosing remote IP addresses when DNS becomes temorarily unavailable ==
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/>
== PBX Routing: A CFNR on a slave gateway object redirecting the call to the master failed ==
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/>-->
== SIP: Switch to T.38 did not work when interworking with H.323 slowstart (XCAPI) ==
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 -->
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/>-->
== PBX: Slave with non-ASCII PBX name did not register at master ==
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 -->
== Alarm/Event handling: Authentication for received remote Alarms/Events ==
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/>-->
== PBX: Preparations for objects visible only if appropriate license installed ==
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/>-->
'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/>
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.<br/><br/><!-- log_if.cpp log_if.h logging.cpp http.cpp -->
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/>-->
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/>
== H.323: Interop Problem with CUCM SIP for Transfer ==
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/>-->
== IP152: Call replacement (blind transfer) did not work ==
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/>-->
== PBX: Potential trap on registration of a slave PBX, with Master GK-ID configuration changes ==
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/>-->
Automatically enter input mode 'alpha' when entering directory search screen through fkey.<br/><br/><!--<br/>phone_dir_ui.cpp<br/>forms_color.cpp<br/>-->
During re-negotiation:<br/>Providing private RTP address to external endpoint although public RTP address is available.<br/><br/><!--<br/>sip.cpp<br/>-->
== SIP: Do not send "504 Server Time-out" after "200 OK" ==
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/>-->
== SIP: Handling of "sendonly" offer after "inactive" offer ==
Touching should start dialing of selected directory entry<br/>instead of opening directory entry for editing.<br/>At least in Context of Wahlvorbereitung.
== SNMP-Traps: agent-addr wasn't reflecting altered IP adress ==
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/>-->
== SIP: Optimization when validating local media address ==
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.
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/>-->
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/>
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/>
== phone: coder settings of a "Create Registration" function key were not applied to the created registration ==
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/>-->
== phone: headset function key mode 'control' could be configured via WEB interface only ==
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/>-->
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/>-->
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/>-->
This SNMP trap is no longer necessary. Meanwhile it is covered more consistently by "innoDiagAlarm" and "innoDiagAlarmClear".<br/><!--<br/>gk.cpp/.h<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/>-->
== Voicemail: <pbx-disc> failed sometimes ==
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 -->
== phone_orchid: pressing the speaker key while in headset mode did not switch over to handsfree mode ==
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 -->
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 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/>
== Phones: New config file option /recording-without-remote-party-info ==
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/>-->
== phone_orchid: configuration prameters to adjust microphone and speaker volume ==
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/>
== IP-DECT: Radio reconnect handling with OEM PBX ==
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/>-->
== ip1202: support capture of raw ethernet packets exchanged between MSP (aka DSP) and the firmware ==
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.
== HTTP-Client: Requests with specified credentials ==
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/>-->
== Phones: New config option "Allow User Settings at Phone" ==
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/>-->
== IP-DECT: New hidden Radio option 'Max RTP streams' ==
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" ==
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/>-->
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 ==
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 ==
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 ==
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 ==
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/>-->
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 ==
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 ==
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 ==
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 ==
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/>-->
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 ==
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 ==
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/>-->
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 ==
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 ==
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 ==
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) ==
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 ==
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: <pbx-fwd>, Forward Timer Didn't Fire For Connected Calls ==
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/>
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 ==
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/>
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 ==
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 ==
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 ==
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 ==
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 ==
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 ==
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 ==
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/>-->
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 ==
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 ==
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 ==
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/>-->
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 ==
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 ==
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 ==
-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/>
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 Trunk: New destination for rejected calls ==
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/>-->
== SIP: Offer media encryption as separate media description ==
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/>-->
== SIP: Do registration refresh more ahead of expiration ==
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/>-->
== phone: Modification of function key on phone can be disabled for any single key ==
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 editableïï in this mask can be made ``not editableïï by setting the above mentioned checkmark<br/>A key of a type marked as ``not editableïï 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/>
== phone: "Spare" function key to reserve key positions for administrative purposes ==
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 -->
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/>
== Webdav: Write information into trace if DELETE fails because file is in open state ==
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/>-->
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/>-->
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/>-->
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 ==
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 ==
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 ==
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 ==
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/>-->
== IPVA: Enhancement for innovaphone testbed: Indicate whether Upload to DRAM took place ==
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 ==
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/>
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 ==
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 ==
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 ==
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 -->
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 ==
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/>-->
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/>-->
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 ==
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) ==
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 ==
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 ==
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/>-->
== PBX-SOAP: Present normalized number of peer also ==
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 ==
-> 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/>-->
== case independence for the characters of the Basic Russian Alphabet added ==
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 ==
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/>-->
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 ==
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 ==
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 ==
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 ==
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 ==
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/>-->
== PBX: CC Requests were sent with wrong number if a SendNumber was configured at the user ==
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 -->
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/>-->
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 ==
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 ==
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/>-->
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/>-->
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 ==
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/>
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 ==
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 ==
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 ==
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 ==
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/>-->
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 ==
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/>-->
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/>-->
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 ==
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/>-->
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" ==
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 ==
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 ==
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 ==
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/>
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 ==
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' ==
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/>-->
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/>
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) ==
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/>-->
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 ==
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 ==
A SOAP applicatoin (e.g. TAPI) uses the method UserClear to clear a call. This could cause a trap on some platforms when doing this for a mobile endpoint.<br/><!--<br/>pbx_xml.cpp<br/>-->
== SRTP: Avoid one-way media with high start sequence numbers ==
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 ==
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 ==
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/>-->
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 ==
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 ==
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 ==
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 ==
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 ==
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/>-->
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 ==
New URL parameter "fallback=true".<br/>If given and the specified media file cannot be retrieved,<br/>webmedia starts playing builtin MOH.<br/><br/><!--<br/>webmedia.cpp/h<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 -->
== PBX: Trap if user object is deleted, which is used by other applications (e.g. myPBX) ==
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 ==
$_leg2tweak<br/>--<br/>Controls <pbx-getcallinfo out-leg2=".."/><br/>true(default): set leg2 to <ext-nr> from <vm-nr>+<ext-nr><br/>false: set leg2 according to received divertingInfoLeg2 facility<br/><br/>$_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 ==
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 -->
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 ==
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/>-->
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 ==
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 ==
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 ==
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 ==
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 ==
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/>-->
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 ==
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 ==
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)" ==
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 ==
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/>-->
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 ==
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/>
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" ==
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/>-->
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/>-->
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/>-->
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 ==
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 ==
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 ==
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 hnlich wie bei AC494E -->
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/>-->
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 ==
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 ==
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/>-->
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 ==
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 ==
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 -->
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 -->
== PBX: New option for RTP Proxy - proxy only if different registration address ==
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/>--->
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 ==
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 -->
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 -->
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/>-->
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 ==
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 ==
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 ==
- 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
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 ==
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/>-->
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 ==
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/>-->
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 ==
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 ==
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 ==
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 ==
<!-- orchid_lcd.cpp orchid_lcd.h config.h --><br/><br/>Beim Neuladen des Sequencers bei nderung der LCD-Helligkeit kommt der UART-Takt gelegentlich zu schnell. Besser noch wre 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 ==
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 ==
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, ==
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 -->
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 ==
For a mobility call, which was established with data callback no DTMF R-Key Features (e.g. put the call on hold) were possible.<br/><!--<br/>pbx_mobility.cpp<br/>-->
== SIP: Switch-over to t38 did not work in one configuration scenario ==
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/>-->
== phone: ip222,ip232: ignore HID function of USB headset charging cables ==
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 -->
If the registration to the primary destination, which used discovery failed, the alternate registration to IP address was sent to the discovery port (1718) instead of the registration port (1719).<br/><!--<br/>h323sig.cpp<br/>-->
== phone: ip222,ip232: support USB headsets with two audio input channels from microphone ==
Simultaneous tone generation and caller id generation caused DSP problem on analog gateways.<br/><br/>Tone generation with undefined coder disabled to avoid this problem.<br/><br/><!-- ac_dsp3.cpp -->
== phone: ip222,ip232: "Phone/Preferences/Use Handset like a Headset" mode did not work ==
on an ip2x2 only the headset function key (Mode:Control) can be used to to start an outbound call or to accept an inbound call in this mode but the headset function key was ignored when no headset was plugged/enabled.
== SIP: Handling of 180 with SDP answer is required after 180 without SDP ==
Must re-create message-summary subscription after re-connecting to server after local address change.<br/>Must not re-use call-id and tags for re-subscription.<br/><br/><!--<br/>sip.cpp<br/>-->
== phone: ip222,ip232: prevent unintentional autoconnect of an inbound call arriving while the headset radio link runs down ==
This happens when another call was just released and the new call arrives while the phone is in idle state but the headset base is running down the radio link.<br/><br/><!-- keys_ip222.cpp -->
== Phone: NOTIFY(sipfrag) was missing after transfer complete ==
When call replacement (REFER) is completed a NOTIFY(sipfrag:200/OK) must be sent to the sender of REFER (transfering party).<br/><br/><!--<br/>phonesig.cpp<br/>-->
== Phones: Allow changes to language, ringtones and fkeys if "Allow User Settings at Phone" is activated ==
If a CFNR to a Slowstart endpoint (e.g. XCAPI) was performed after a call to an EFC endpoint, the fallback to slowstart did not work in a szenario with multiple PBXs.<br/><!--<br/>h323ch.cpp<br/>-->
== phone: ip222,ip232: delay ringing to USB headset when a previous call was released immediately before ==
Some USB headsets (even wired ones) need a surprisingly long time to disconnect (up to 500 millisecons). To play the ring tone the headset must be connected again and this may fail before the disconnect is completed.<br/>The default delay is one second from start of last disconnect, it can be set by<br/> config add PHONE APP /usb-calm <ticks><br/>where <ticks> means 20 ms timer ticks.<br/>
The new firmware rejects commands with STALL which were accepted by the older firmware (the commands were sent with trailing zeros which were silently ignored). This may also apply to Blackwire C420 / C435 / C620 which use the same firmware.
== SIP: Display name of "original called party" was missing ==
Display name of "original called party" (first diverting party) was missing in case there were multiple diversions.<br/>When processing INVITE with (multiple) History-Info header.<br/><br/><!--<br/>sip.cpp<br/>-->
== phone: ip222, ip232: support Jabra UC Voice 750 and Jabra Speak 510 ==
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 ==
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/>-->
New interop tweak /c-line-at-session-level for clients that do not read RTP address from media description.<br/>For clients not compliant to RFC-4566.<br/><br/><!--<br/>sip.cpp/h<br/>-->
== phone: a park function key with both 'Number' and 'Name' left empty implies to use 'Number' or 'Name' of the Registration ==
If the user prefers a different setting the automatically derived volume can be overridden with a fixed volume as before via<br/> config add PHONE SIG /notify-pickup-gain <gain><br/>( see http://wiki.innovaphone.com/index.php?title=Howto:Change_the_volume_of_the_pickup_key_audio_notification )<br/><br/><!-- phonesig.cpp app_fkey.cpp phone_edit.cpp inca_afe.cpp -->
== phone: flag to disable speaker key to prevent conversations in handsfree mode ==
When calling the Mobility object from the mobile phone, additional dialed digits are used to call the destination. This is an alternative to using DTMF for dialing. How many digits may be dialed depends on what the network of the mobile phone supports<br/><!--<br/>pbx_mobility.cpp<br/>-->
If an admin had restrictions for PBX oder Node, this retrcition was not shown after pressing Apply when editing an object.<br/><!--<br/>pbx_admin.cpp<br/>pbx:admin.h<br/>-->
NAT workaround was activated during transfer in some cases.<br/>Late packets from transferring endpoint may trigger the NAT workaround at the 2 transferred endpoints, making the transferred endpoints redirecting to the obsolete RTP addr/port of the transferring endpoint.<br/><br/><!--<br/>medialib.ccp/h<br/>-->
== Replication: Distribution Of Trace Flag To Existing Connections ==
Bug in the crypto crypto driver. When the SRTP key is changed while a packet is being encrypted the SRTP socket hung up.<br/><br/><!--<br/>srtp_cipher.cpp<br/>-->
The call transfer timeout is changed to 25s for transfer to mobile targets. Used by OEM PBXs.<br/><br/><!--<br/>Changed file:<br/>dectradio_call.h<br/>-->
== myPBX: Add node escape prefix to number from PBX LDAP directory ==
Internal numbers might differ depending on the caller, due to node membership. So the internal number has to be adapted for the individual users.<br/><!--<br/>pbx_api.h<br/>pbx.h<br/>pbx.cpp<br/>pbx_client.h<br/>pbx_client.cpp<br/><br/>test\\10.00\\unit-tests\\client\\iteration05\\160-int-search-other-node.xml<br/>test\\10.00\\unit-tests\\client\\iteration05\\161-int-search-other-node-escape.xml<br/>-->
== SIP: No INVITE is sent if client has registered with a local domain name ==
No INVITE is sent if client has registered with a local domain name instead of ip address.<br/><br/>E.g.<br/><br/>\tREGISTER sip:172.16.16.124 SIP/2.0<br/>\tVia: SIP/2.0/UDP kws6000.noexist.local:5060;branch=z9hG4bK40ce4a47e899<br/>\tFrom: ;tag=40b10bdf37c5<br/>\tTo: <sip:sga@172.16.16.124><br/>\tCall-ID: 4040c07c656162185bf6d53451f1ecae<br/>\tCSeq: 387119859 REGISTER<br/>\tContact: <sip:sga@kws6000.noexist.local><br/>\tAuthorization: Digest username="sga", realm="172.16.16.124",...<br/>\tAllow: OPTIONS, INVITE, ACK, CANCEL, BYE, SUBSCRIBE, NOTIFY,...<br/>\tMax-Forwards: 70<br/>\tUser-Agent: KIRK Wireless Server 6000 PCS13__ r40453<br/>\tSupported: replaces<br/>\tSupported: 100rel<br/>\tExpires: 60<br/>\tContent-Length: 0<br/><br/><br/><!--<br/>sip.cpp<br/>-->
== H.323: No media after Hold/Retrieve on call with multiple media-relay ==
This could be considered a mis-configuration, because a single media-relay should be good enough, but should work anyway.<br/><!--<br/>h323ch.cpp<br/>-->
a response to a client request received via ETH1 was sent with the ETH0 address as source address when routed through default gateway on ETH0.<br/>a response with a source address not matching the adressed server is discarded on client side.<br/><br/><!-- ntp.cpp -->
== Correctly distinct Quick Dial and Directory Search object ==
Sometimes die Quick Dial object became a Directory Search object, if the submit has been rejected.<br/><br/><!--<br/>pbx_edit_dirsearch.xsl<br/>pbx_edit_qdial.xsl<br/>-->
== Escape Mechanism for Flash Directory Objects ==
The mechanism's motivation is to avoid unnecessary binary encodings and to spare some length per configuration line.<br/>http://wiki.innovaphone.com/index.php?title=Concept_Flash_Directory#Escape_Rule_For_Object_Values<br/><!--<br/>test/platform/platform.h<br/>test/test/os/test_os.h<br/>test/test/os/test_flashman.h<br/>test/test/os/test_flashman.cpp<br/>test/test.vcproj<br/>test/test/os/test_os.h<br/>test/common/interface/flash.h<br/>test/common/service/ldap/ldaplib.h<br/>test/common/service/ldap/ldaplib.cpp<br/>test/common/service/ldap/flashdirui.cpp<br/>test/common/service/ldap/flashdir.h<br/>test/common/service/ldap/flashdir.cpp<br/>-->
== phone: support transparent recording of calls to/from other registrations than the active registration ==
If a registration is configured for 'transparent' recording a call via this registration is always recorded now even if the registration is not the 'active' registration and no recording is configured for the 'active' registration.<br/><br/><!-- phone_config.h phone_config.cpp phone_edit.h phone_edit.cpp fkey_edit_record.xsl app_ctl.h app_ctl.cpp app_call.cpp app_cc.cpp app_fkey.cpp phone_user_func.cpp -->
== SIP: Workaround for SIP client giving wrong ip address in Contact-URI ==
Workaround for SIP client giving wrong ip address in Contact-URI.<br/>3CXPhoneSystem gives ip address of PBX in Contact-URI.<br/>PBX sends upcoming SIP requests to its own ip address.<br/><br/><!--<br/>sip.cpp<br/>-->
Pressing a 'Dial' function key whith 'Prepare' checked permits further editing of the string configured under 'Number' or 'Name'. The keyboard mode was set to 'alpha' if the string contained any character not permitted in an e164 number. Now the mode is set to 'numeric' when the last character of the string is a decimal digit.<br/><br/><!-- forms_ctrl_impl.cpp -->
== myPBX: Possible trap on configuring visibility and call diversions ==
STUN server could not be removed from Gateway interface config without reboot.<br/>It disappeard from config, but was still used by SIP interface.<br/><br/><!--<br/>sip.cpp<br/>-->
== PBX: The checkmark "RTP Proxy - Except Addresses are identical or private" could not be cleared ==
The reset needed message is sometimes missed if the network configuration for Linux is changed. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>ethif.cpp<br/>-->
== PBX: Blind Transfer to a different Slave could fail under special circumstances ==
The blind transfer failed when<br/>- e.164 style nodes are used<br/>- transfer destination is thru a trunk on a different PBX<br/>- transfering endpoint is configured to a node, which is another PBX<br/>- transfered call is from trunk<br/>- ...<br/><!--<br/>pbx.cpp<br/>-->
The radio call list can't be shown with Mozilla if an OEM PBX is used and a 'R'-key has been pressed. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dectradio_mastercall.cpp<br/>-->
== PBX: Boot-Loop when replicating objects with wrong password ==
When a replication was configured, replicating all objects from a different PBX, with different PBX password then configured already on the box, the box entered a boot loop, which only could be stopped with a long reset.<br/><!--<br/>pbx.cpp<br/>-->
== phone: ip222,ip232: Unintended autoconnect to handset when headset USB port connection is lost while ringing ==
If a Jabra PRO 930 has "Auto sleep Mode" enabled it enters sleep (low power) state when it was not used for 8 hours.<br/>When a call arrives this is signaled to the headset. When the Jabra PRO 930 is in sleep state this works for about one second but then the USB port connection gets lost.<br/>The handling of such presumably temporary failures was not correct in case of a ringing call.<br/><br/><!-- keys_ip222 --> <br/>
== IP6000: Prevent blinking error LED on old IP6000 with HW-Build 200 ==
The destination of the reroute was wrong (digits missing) if the reroute was done on a trunk object not in the root node with escapes<br/><!--<br/>pbx.cpp<br/>pbx_trunk.cpp<br/>-->
Preconditions:<br/>- "Phone/User-x/Preferences/Enable Call Intrusion" checked<br/>- "config add PHONE APP /recall-ci-monitor" performed<br/>In the Recall-Menu opened when the Menu-key is pressed after dialling a busy user the "Monitor" option is offered in addition to the "Intrude" option.<br/><br/><!-- app_ctl.h app_cc.cpp app_cfg.h app_cfg.cpp app_gui.h app_gui.cpp -->
== SIP: New option "Filter incoming calls" on gateway interfaces ==
This only happend when during the submitting of the object the HTTP session was terminated extremly quickly.<br/><!--<br/>pbx_admin.cpp<br/>pbx_admin.h<br/>pbx.cpp<br/>pbx_api.h<br/>-->
== SIP: Config option /take-zero-addr-for-hold did not work anymore ==
Config option /take-zero-addr-for-hold did not work anymore.<br/>Since v9hf16 (#85534: SIP: Interop with Genband SBC).<br/><br/><!--<br/>sip.cpp<br/>-->
== SIP: Memory leak when handling SUBSCRIBE(dialog-info) ==
Memory leak when handling SUBSCRIBE(dialog-info),<br/>but only if a endpoint sends overlapping SUBSCRIBE requests<br/>to same destination with different Call-ID's.<br/><br/><!--<br/>sip.cpp<br/>-->
== Gateway: Trap on collision of call-complation termination collision ==
A restart could happen if a call-completion monitoring was terminated at the same time from the network and from the user/pbx.<br/><!--<br/>relay.cpp<br/>-->
some firewalls complain if short ehternet frames are padded to more than 60 bytes which was mistakenly done in some ethernet drivers.<br/><br/><!-- titan_drv.cpp, mv78x00_drv.cpp , orchid_drv.cpp -->
== PBX: Master/Slave Max Calls take subscriptions into account ==
The fact that the call was accepted was treated as a user action, but the accept was automatic, so no user has touched the phone<br/><!--<br/>pbx.cpp<br/>-->
== PBX-SOAP: Memory leak when terminating a SOAP session in an unusual way ==
There is a trap in DECT radio if the user login feature is used. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>dectradio_reg.h<br/>dectradio_call.cpp<br/>-->
== DHCP: A 'Coder' manufacturer option longer than 31 characters could not be configured at server and not evaluated by client ==
A coder config longer than 31 characters could not be entered in the field<br/>"IP4/ETXn/DHCP Server/Offer Parameters/Coder" and the DHCP Client silently discarded a longer coder config possibly provided by a non innovaphone DHCP server.<br/><br/><!-- dhcp_if.h -->
== phone: ip222,ip232: handle additional product id for 'Jabra BIZ 2400 Mono USB' ==
Previous DSP has a better echocanceller.<br/>Also the IP241 Handset micrphone parameters are updated.<br/>The IP241 handset receiver equalizer is unchanged.<br/><br/><!-- ac_494e.cpp ac_codec3.cpp --><br/>
The object initiating the CFNR was lost when the call was sent to the master.<br/>Collateral damage of<br/>fix: #99674: PBX: CFNR/CFB on PBX object did not work in some cases<br/><!--<br/>pbx.cpp<br/>test: pbx/dis, 392-cfnr-ntl-trunk-local.xml<br/>-->
== Media: Do not write error log if RTP is received before media negotiation is complete ==
Do not write error log if RTP packets are received before media negotiation is complete.<br/>Error 0x00050003 (Wrong Payload Type received) was generated before.<br/><br/><!--<br/>media.cpp<br/>sip.cpp<br/>-->
== Increasing memory usage when viewing PBX pages with Kerberos login ==
When the the PBX pages are displayed using a Kerberos login, some command_exec objects are never deleted. This causes increasing memory usage.<br/><br/><!--<br/>command.cpp<br/>missing mod_login_pening++ in commmand_exec::login_result<br/>-->
== PBX E.164 Configuration: Call forward to remote Trunk, should call internal loopback destination ==
If a call forward is configured to the switchboard of a remote location (typically -0, same as trunk prefix) the call should not be sent out to the trunk, but the internal loopback destination should be called.<br/><!--<br/>pbx_trunk.cpp<br/>-->
== EDSS1 Interworking: Interworking Of Incoming Partial Rerouting Failed ==
A number field wasn't initialised, leading to an interworking fault at the boundary between EDSS1 and H.450.<br/><!--<br/>q950.cpp<br/>relay.cpp<br/>-->
== SIP: Cannot change a password on DECT systems without restart ==
Memory leak when receiving BYE for a dialog in early state.<br/>On a call which is not connected yet.<br/>On a call where a INVITE server transaction is pending.<br/><br/><!--<br/>sip.cpp<br/>-->
== HTTP client: Update of nonce is ignored in digest authentication ==
Once digest authentication is chosen the HTTP client does not accept any more changes to the digest parameters in the same session.<br/><!--<br/>httpclient_i.cpp<br/>-->
== IP232,IP222, IP241: Fkeys may overlap call control ==
Memory leak when receiving 403 after 401 for REGISTER:<br/><br/> REGISTER<br/> 401 Unauthorized<br/> REGISTER with Authentication<br/> 403 Forbidden<br/><br/><!--<br/>sip.cpp/h<br/>-->
The volume of the pickup notification tone is derived from the volume configured for the internal ring tone.<br/>If this volume is not appropriate it can be set to a fixed value (see http://wiki.innovaphone.com/index.php?title=Howto:Change_the_volume_of_the_pickup_key_audio_notification ).<br/><br/><!-- phonesig.cpp phonesig.h app_fkey.cpp codec.cpp codec.h ac_codec3.cpp inca_afe.cpp -->
== SIP: New config file option /no-cng-tone-detection ==
Call was dropped after successful session refresh.<br/>Handling of 200/OK for UPDATE was wrong.<br/>Was wrong since bug fix #103996 (v9hotfix27)<br/><br/><!--<br/>sip.cpp<br/>-->
== SIP: Memory leak when receiving BYE while re-INVITE server transaction is pending ==
Do not send SDN answer in ACK if it already been sent in PRACK.<br/>Regards early media scenarios that starts with INVITE without offer.<br/><br/>INVITE(no sdp)<br/><br/>183(sdp offer)<br/>PRACK(sdp answer)<br/>200(PRACK)<br/><br/>180(no sdp)<br/>PRACK(no sdp)<br/>200(PRACK)<br/><br/>200(no sdp)<br/>ACK(no sdp)<br/><br/><!-- sip.cpp -->
When handling a call leg replacement the Gateway releases the replaced call before accepting the replacement call.<br/>May confuse the replacing endpoint.<br/>In case of SIP this regards handling of INVITE with Replaces header.<br/>In case of H.323 this regards handling of SETUP with ctSetup facility.<br/><br/><!--<br/>relay.cpp/h<br/>-->
== PBX Map: Overlap dial thru a Map Object on Slave with a call via the Master did not work ==
If a phone registers from a different location, any call from this phone has to be routed via the master to check for 'local' objects. In this case overlap dialing thru a Map object on the slave did not work.<br/><!--<br/>pbx_map.cpp<br/>-->
== phone: importing a phonebook may result in memory leaks ==
A rare trap can occur in the IP-DECT Radio with IP1202 and a multi-master solution.<br/><br/><!--<br/>Changed files:<br/>dectradio.cpp<br/>dectlocalusers.h<br/>dectlocalusers.cpp<br/>-->
== phone: fine grained function locking - PHONE_LOCK_USER_INFO bit supresses display of local user info ==
For phones installed in rooms open to the public it's sometimes required to prevent this phones from beeing called by non authorized persons. Adding this bit to the mask defined under "Phone/Protect/Fine grained Function Locking" supresses any info about the local user (number/name/display name).<br/>
happens for all non-ip28 (ip22/24/302/305) gateways if hardware build >= 402, causes the gateways to stall due lack of clocking<br/><!--<br/>ip24.cpp<br/>-->
== SIP: Different registrations for the same AOR from same ip address and same port were handled as one ==
SBC forwards different registrations for the same AOR to the PBX from same SBC ip address and SBC same port.<br/>PBX must take this as individual registrations as long as Contact-URI differs.<br/>Even is REGISTERs are sent from same ip address and port and for same AOR.<br/><br/><!--<br/>siptrans.cpp<br/>-->
In a rare case a trap can occur if the login feature is used and the master is changed. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>dectradio.h<br/>dectradio.cpp<br/>dectradio_call.h<br/>dectradio_call.cpp<br/>-->
Each call only a single group indication was removed from the queue, if the rate of group indications was higher then the rate of failed calls, the memory for group indications accumulated.<br/><!--<br/>pbx_gi.cpp<br/>pbx_gi.h<br/>-->
Some transferred remote initiated calls have no voice connection. This are calls which are initiated with myPBX to an external endpoint. This is fixed now.<br/><br/><!--<br/>Changed files:<br/>dectradio_call.cpp<br/>-->
The Linux DNS configuration changes the local device DNS configuration. This shouldn't be and is fixed now.<br/><br/><!--<br/>Changed file:<br/>ethif.cpp<br/>ipproc.cpp<br/>linux_ip_ethif.xsl<br/>-->
== Linux: Empty server identifier and no NTP server if only ETH1 is used ==
If only ETH1 is used and Linux gets a fixed IP address, the DHCP message doesn't include a valid server identifier and NTP server address. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dhcp.cpp<br/>-->
sometimes the display and USB hardware was not working after a reset (firmware update or configuration change) and a power cycle was required to bring them up again.<br/><br/><!-- boot241.y, boot222.y, boot232.y -->
== phone: ip222,ip232: support Jabra UC VOICE 550 / 750 Version A headset models ==
Headsets with Version A printed on the package have IDs different to the non-A versions even if the part numbers do not differ. The USB firmware of the Version A headsets differs from the predecessor firmware and requires a special timing.<br/><br/><!-- keys_ip222.cpp usb_audio.cpp usb_audio.h usb_hid.cpp usb_hid.h usb_msd.h ohci.cpp ohci.h usb_host_if.h physical.cpp physical.h -->
This happened if there was a call-forward, which resulted in a call to the master and the master could not send the call to the destination slave either because this Slave was not registered or there was a busy-out setting preventing it.<br/><br/>The problem only happend with a E.164 config.<br/><!--<br/>pbx.cpp<br/>-->
The Plantronics bluetooth headsets Voyager PRO UC, Voyager Legend and Calisto 620<br/>come with an USB bluetooth dongle with one of the product codes 0415, 0416, 0417. Dongles with the product code 0416 were not detected. <br/><br/><!-- keys_ip222.cpp -->
== phone: under soap control no audio data was sent when a call was retrieved after another call has been transferred ==
Since fix: "#97150: phone: DTMF digits following a comma in a number to be dialed were not handled correctly in some cases." from 21.3.2013 a series of comma didn't extend the wait time before DTMF dialing any more. The wait time was always 1 second because only the last comma was seen.
== phone ip222,ip232: Plantronics Savi W440 Headset sometimes mute when controlled by a SOAP-Application or myPBX ==
When an outbound call was started by a SOAP-Application after a call started using the headset Talk-button the headset was mute because the radio link was not established.<br/><br/><!-- keys_ip222.cpp --><br/>
== Gateway/H.323: Trap when canceling an call with Media Relay because out of Resources ==
In this case the cleaup of the outgoing call was incorrect and caused a trap. Only happened when the outgoing call was H.323.<br/><!--<br/>h323ch.cpp<br/>-->
== PBX: Connected Number missing on calls to some PBX objects ==
The connected number is needed to determin if the destination of the call is internal, which is needed for features like not automatic recording of external calls<br/><!--<br/>pbx.cpp<br/>pbx_map.cpp<br/>-->
== PBX: Handling of enblock (sending-complete) calls improved ==
Respond with CallProceeding, so that if the call is rejected, there is some ack before the reject. Otherwise this would look like an error.<br/><!--<br/>pbx.cpp<br/>relay.cpp<br/>pbx_trunk.cpp<br/>-->
== Trap in webdav client when processing XML directory listing ==
By default address and gatekeeper ID of the PBX where the user is registered are used and the 'Long Name' is searched. Now for example this can be changed to use the master PBX and to search the 'Display Name'.<br/><br/><!-- reg_edit_dir.xsl phone_dir.cpp phone_user_ui.cpp --><br/><br/>
== phone: set up call with "Sending complete" when the number has been provided before the call is initiated ==
This applies to calls initiated while browsing a directory or a call list, by pressing a dial function key or via indirect dialing, i.e. when a number is entered before going off-hook.<br/>To permit for incomplete numbers in a phone directory "Sending Complete" is not set when a number is terminated by a '+' character. Then the '+' is stripped off and the number can be completed by typing more digits.<br/>In this case and in case the user goes off-hook before typing any digit the number is assumed to be complete when a '#' character is entered or the "Enblock Dialing Timeout" is reached before the next digit was entered.<br/><br/>The old overlap sending behaviour can be restored by<br/> config add PHONE SIG /overlap-sending<br/><br/><!-- phonesig.cpp, phonesig.h, phonesig_if.h, phone_list_ui.cpp, phone_dir_ui.cpp, app_fkey.cpp --> <br/>
== phone: ip222,ip232: support for Jabra BIZ 2300, Sennheiser Presence UC ==
If a slowstart endpoint performed a transfer, connecting two efc endpoints on other PBXs, it could happen, that the media negotiation between the new endpoints was slowstart, because the PBX on which the transfer was performed did not receive the efc-featurse<br/><!--<br/>h323ch.cpp<br/>h323ch.h<br/>h323sig.cpp<br/>-->
== IP222 IP232: Option to disable Energy Efficient Ethernet (EEE) added ==
config add AC-DSP0 /headset-only<br/><br/>unconditionally disables ringing via speaker when a headset is plugged and enabled. this is done independent of the "Do not Disturb" setting. <br/><br/><!-- ac_phone3.h ac_phone3.cpp ac_codec3.cpp -->
== Voicemail: <exec> Without "url" Causes Trap ==
This fix just helps where an object's representation within the configuration file expands beyond the 8K barrier. <br/><br/>http://wiki.innovaphone.com/index.php?title=Concept_Flash_Directory#Config-Encoding_Of_Objects_Breaking_Through_8K_Line_Length<br/><!--<br/>flashdir.cpp/.h<br/>-->
== ip1202: config flag to force reboot when receive interrupts are missing for a certain time ==
By default the the MAC is is reset in case of missing receive interrupt.<br/><br/> config add ETH0 /rx-miss-reboot<br/>forces a reboot instead of a MAC reset. <br/><br/> config add ETH0 /rx-wait-max <seconds><br/>defines the maximum time to wait after the last receive interrupt before MAC reset or reboot (default is 30 seconds).<br/><br/> config add ETH0 /itrace<br/>activates an interupt backlog which is written to trace buffer before MAC reset or reboot.
== TLS: Problem with negotiation of protocol version on server side ==
When the client offered TLS 1.2 or higher, the connection was refused instead of downgrading to the highest supported protocol version.<br/><!--<br/>tls.cpp<br/>tls.h<br/>unit-tests/tls/basic/153-handshake-server-normal-tls15-downgrade.xml<br/>-->
== NT ISDN Point to Multipoint Interfaces: Rejecting of a call had delay of 4.5s ==
A call was not rejected right away, but SETUP was resent in case another endpoint would respond. This should be done only if the call was rejected because of incompatible destination.
== PBX: If an endpoint performs a pickup-req, the resulting call should be sent to the requesting endpoint only ==
If on a user two phones were registered and one phone performed a pickup, both phones were ringing for the call to be picked up.<br/><!--<br/>pbx.cpp<br/>pbx.h<br/>pbx_api.h<br/>-->
This could cause problems were it is important to know if the call is internal or external, for example if only external calls are to be recorded.<br/><!--<br/>pbx.cpp<br/>pbx_dtmf.cpp<br/>-->
== phone: In Recording Mode 'transparent' or'optional' internal calls were recorded although 'External Calls Only' was checked ==
happened only to outbound calls initiated by some dialing application.<br/>outbound calls initiated directly at the phone and inbound calls were recorded correctly.<br/><br/><!-- app_call.cpp -->
== Gateway trap with 'Out of Memory' when CF-card stucks ==
A CF-card that stucks leads to huge memory allocations of type cf_command containing non-processed CF-requests.<br/><!--<br/>cf_drv.cpp<br/>cf_drv.h<br/>-->
== phone: In Recording Mode 'transparent' or'optional' a 2nd call started by a dialing application could terminate the 1st call ==
This happened when a 2nd call was started by a dialing application and then terminated again while the call was in alerting state.<br/><br/><!-- phonesig.cpp --><br/>
== H.323: No media if a reverse Media call is sent to a slowstart endpoint and tranfered to a EFC endpoint ==
Media negotiation problem which could happen under special conditions when an XCAPI application is performing a call transfer<br/><!--<br/>channel.h<br/>-->
== DHCP: Increase maximum length of "Local Networks" and "IP Routing" option strings from 127 to 252 characters ==
If there is no progress indicator indicating inband tones, channels should not switched on for calls sent out to an NT Mode ISDN interface. Otherwise RTP containing silence could switch off any locally generated ringback.<br/><!--<br/>nt_tbl.tbl<br/>-->
Since v9hotfix22/v10beta6 recording connections are established in sendonly mode because usually recorders do not send any media data and thus bulks of ?No Media Data received? events may be reported.<br/><br/>Now two way media can be explicitely enabled by checking "Phone/User x/Recording/Two Way Media" if required (for example for the Innovaphone Operator "Greeting Function").<br/><br/><!-- phone_config.h phone_config.cpp phone_edit.cpp reg_edit_rec.xsl phonesig_if.h phonesig.h phonesig.cpp app_ctl.h app_ctl.cpp app_call.cpp -->
== Gateway trap with 'Out of Memory' when CF-card stucks ==
A CF-card that stucks leads to huge memory allocations of type cf_command containing non-processed CF-requests.<br/><!--<br/>cf_drv.cpp<br/>cf_drv.h<br/>-->
== SIP: The 2xx response to the REGISTER request MUST contain, in a Contact header field, a complete list of bindings ==
The 2xx response to the REGISTER request MUST contain, in a Contact header field, a complete list of bindings that have been registered for this address-of-record.<br/>Not only the one that has just been added.<br/><br/><!--<br/>sip.cpp/h<br/>siptrans.cpp<br/>-->
== SIP: Interop to "Thomson TB30 hw4 fw2.77.0.2 00-26-44-30-5D-A2" ==
Calls disconnected by the gatekeeper can be wrongly resent to the handsets, if the calls are disconnected with the release code Non-selected-user-clearing (26). This is fixed now.<br/><br/><!--<br/>Changed files:<br/>dectmaster.cpp<br/>dectfty.cpp<br/>-->
== PBX Gateway: Internal Destination flag did not work for outgoing calls ==
If external transfers are not allowed, this flag should allow a transfer to a gateway object for a call coming in from an external source.<br/><!--<br/>pbx.cpp<br/>pbx_api.h<br/>pbx_gw.h<br/>-->
== H.323: Internal/External information got lost on Endpoint after Transfer ==
The information if the endpoint to which a phone is connected after a transfer is internal or external was not available on the phone. The recording of internal or external calls only did not work in this case.<br/><!--<br/>h450.cpp<br/>-->
== phone: ip222,ip232: support Jabra Pro 935 USB-Bluetooth Headset ==
If two secretaries were configured with names, being identical in the first half and identical length (e.g. 'Hans' and 'Harz'), for some functions like presence status not the correct secretary was found.<br/><!--<br/>pbx_exec.cpp<br/>-->
== phone: ip110/150/200a/230/240: false "Excessive loss of Data" reports when playing Music on Hold (MOH) ==
- workaround for 10/100Mb/s gemac Rx lockup:<br/> the interface is run in promiscuous mode and the driver filters the packets<br/>- workaround for Rx Queue Overrun problem:<br/> on a Rx Queue Overrun interrupt gemac and phy are rest completely<br/><br/>- for test purposes promiscuous mode can be disabled|enabled by<br/> !config add ETH0 /rx-promiscuous 0|1<br/> or temporaryly by<br/> !mod cmd ETH0 rx-promiscuous 0|1 <br/><br/><!-- comcerto_eth_drv.cpp comcerto_eth_drv.h comcerto_eth_drv_s.xsl -->
== PBX: Pickup with partner key did not work if visibility was configured by name ==
The call was displayed on the partner key, but the pickup did not work. It did work if visibility was configured with a group.<br/><!--<br/>pbx.cpp<br/>-->
== PBX-SOAP: Clearing of call to waiting queue took 30s ==
There is a rare trap in DECT-Master if a new call is sent to the radios and there still exists an old call for the endpoint and this call is assigned to an unregistered radio. The trap only occurs on the IP1202, not the IP1200. This is fixed now.<br/><!--<br/>Changed files:<br/>dectmaster_if.h<br/>dectmaster.h<br/>dectmaster.cpp<br/>dectfty.h<br/>dectmsg.h<br/>mihotdesk_call.h<br/>-->
== DHCP-Server: strip leading and trailing spaces from values entered in "IP4/ETHx/DHCP-Server/Offer Parameters" ==
If a Park function key is used to park a call and pickup it again, a restart happend. This is a collateral damage from<br/><br/>fix 115569: PBX: If an endpoint performs a pickup-req, the resulting call should be sent to the requesting endpoint only<br/><br/>from v10sr8 and v9hf33<br/><!--<br/>pbx.cpp<br/>-->
== PBX: Pickup accross locations from different nodes did not work ==
Offered wrong local IP address as RTP address.<br/>Collateral damage of #119269: SIP: Offered wrong local IP address as RTP address<br/><br/><!-- sip.cpp/h -->
== phone: ip222,ip232: Jabra UC VOICE 550/750 Version A - Microphone occasionally mute ==
Happened with command lines containing options without a value, for example a<br/> config change AC-DSP0 HEADSET /spk-volume /mic-volume 5<br/>did not affect the microphone volume.<br/>Further input was not validated so big negative or positive values gave confusing results. <br/> <br/><!-- ac_codec3.cpp -->
== PBX: Transfer with consultation in ringback - no ringback after transfer if performed by analog phone on IP22/.../IP28 ==
It is not a problem of the analog interface, but the PBX, which does not play ringback if a retrieve is done before the transfer, which is done by the FXS.<br/><!--<br/>pbx.cpp<br/>-->
== PBX: When a dyn PBX was deleted, with an id identical to the start of the id of another dyn PBX, this other dyn PBX was broken ==
There is a watchdog trap on IP1202 with enabled feature codes with firmware V9 hotfix37, if more than one master is used and the handset makes a handover. This is fixed now.<br/><br/><!--<br/>Changed file:<br/>dectfty.h<br/>-->
== SIP: Wrong expires value in Contact header of 2xx response for REGISTER ==
Wrong since v10sr9, v9hotfix35.<br/>Wrong expires value in Contact header during registration refresh.<br/>Correct value in Expires header.<br/><br/><!--<br/>sip.cpp/h<br/>-->
With the PBX SOAP API a call can be initiated for a user, without specifying for which device the call should be initiated. In this case a default device is picked. If an application does this for a user with mobility, a restart happens because of a null pointer access.<br/><!--<br/>pbx.cpp<br/>-->
== PBX CDRs: CDR was missing for calls rejected because of busy_on ... ==
If a name with '+' or '%' was configured as device name, these charecters were nocz displayed correctly in the input field.<br/><!--<br/>command.cpp<br/>-->
== phone: do not mute microphone in alerting state ==
For some some analogue endpoints it is not possible to detect when the media connection is really established, it may hapen before connect is signaled to the phone. To prevent confusion when voice is received from remote but the answer is supressed the microphone is unmuted now already in alerting state by default.<br/>The former behaviour can be restored by<br/> config add PHONE APP /mute-while-dialing 1<br/><br/><!-- app_cfg.cpp app_cfg.h app_ctl.cpp -->
== phone recording - supress calling tones and call status display for calls to recording device ==
The customer complained that DTMF feedback tones to the user were too loud. There was one obvious reason in the sources: the table of VoiceOutputGain steps was not in sync with the table of SignalLevel steps, i.e. the relation between speech and DTMF level differed depending on the volume setting. The other point is that DTMF tones are perceived louder than the lower frequency call progress tones. Therefore adjusted the SignalLevel steps to correlate with the VoiceOutputGain steps and introduced 6 dB extra attenuation for DTMF tones compared to call progress tones.
== SIP: Bug in media negotiation when processing reINVITE without SDP offer ==
If a retrieve happens at the same time as a call release of the held call, a trap could happen. The propabilty of this to happen was very low.<br/><!--<br/>relay.cpp<br/>-->
Needed for some PCïs that loose the link with EEE.<br/><br/>This is caused by the "propietary smartEEE " feature of the ethernet phy.<br/>When disabled the link is stable<br/><br/>Regular EEE still works.<br/><br/><!-- orchid_drv.cpp -->
There are noise and peaks in a conference call on the IP800 and IP305 caused by the CONF interface. This is fixed now.<br/><!--<br/>Changed file:<br/>ipac_drv.cpp<br/>-->
== Phone sometimes returns to speaker-mode instead to release the call when going onhook ==
This could happen when handset was lifted while holding speaker key pressed or when the speaker was relased at the same time when the handset was lifted.<br/><br/><!-- forms_keyboard.cpp keys_gen.cpp keys_gen.h keys_ip11x.cpp keys_ip21x.cpp keys_ip230.cpp keys_ip240.cpp keys_ip241.cpp keys_ip222.cpp --><br/>
There is a change in the default configuration which prevents a wrong configuration if a software factory reset is done with the command config clear. IP-DECT handover fails with the wrong configuration. This is fixed now.<br/><!-- Changed files:<br/>dect.h<br/>dect.cpp<br/>-->
== PBX Map: Hide Connected Endpoint, should be evaluated for calls thru Map ==
Happen with a Multicast object: Call to multicast, then hold and another call to the same multicast object, which returns busy, then hangup, which initiates a blind transfer to busy endpoint.<br/><!--<br/>pbx.cpp<br/>-->
== Media Relay: Support for unknown audio codecs ==
Support for unknown audio codecs (e.g. iLBC) when forwarding RTP audio (RTP-Proxy, Media-Relay).<br/><br/><!--<br/>medialib.h<br/>media.cpp<br/>sip.cpp<br/>channel.cpp<br/>-->
== Trap in PBX during boot after downgrade from v11 ==
Call flow:<br/>- A calls B<br/>- B answers -> recording is started<br/>- A stops recording with redial-key<br/>- A opens a consultation call to C<br/>- C answers -> recording is restarted<br/>- A stops recording with redial-key<br/>When A presses the Menu-key a 3-party conference should be started.<br/>This did not work anymore since V9hotfix25 and not at all in V10. <br/><br/><!-- app_ctl.h app_ctl.cpp app_fkey.cpp --><br/>
This could create problems in different places for example in Reporting when two CDRs with same GUID were sent.<br/><!--<br/>box.cpp<br/>command.cpp<br/>-->
== phone: ip222/232 - support for new Jabra EVOLVE headset series and for additional Plantronics Blackwire headsets ==
Using the configuration given below an inbound call is automatically connected to the headset if a headset is plugged and enabled, otherwise the call is rejected with cause busy.<br/> "Phone/User-x/Preferences/Announcement Calls/Micro On"<br/> "Phone/User-x/Preferences/Announcement Calls/Treat any Call as Announcement"<br/> "Phone/Preferences/Route Automatically Connected Inbound Calls to Headset (if enabled)"<br/> "Phone/Preferences/Reject Automatically Connected Inbound Call routed to Headset if Headset is not plugged or disabled"<br/>The last checkmark affects only normal inbound calls. Announcement calls via the PBX MCAST-Announce object or via the "Dial/Announce" Function key will be routed to the speakerphone if no headset is plugged or if the headset is disabled.
== phone: ip222/232 - added support for Jabra BIZ 2300 USB Duo headset ==
If called non-SRTP endpoint rejects, the call is re-tried as RTP call without encryption.<br/>But no if the caller is a SIP endpoint.<br/><br/><!--sip.cpp-->
== PBX CDRs: Records for calls to objects without registrations were missing ==
The call was disconnected without cause, which typically resulted in a display "call aborted" on the calling endpoint instead of "user busy" which could be desired by the application.<br/><!--<br/>pbx_wait.cpp<br/>-->
== IP305, include interface licenses in firmware ==
After an update, Internet Explorer 11 showed a script error "Access Denied" when using popup windows in myPBX 9. This problem only occured when using the myPBX launcher.<br/><--<br/>client.js<br/>-->
== AD Replication: Oversized AD Objects Deleted Replicated Objects ==
An internal error code wasn't set by a handling for the resulting decoding failure. The internal error code is now set to error=86, "LDAP Decoding Error".<br/>The replication will stop consequentially.<br/>Actual cause were the AD objects being member in too many AD groups.<br/><!--<br/>ldapapi.cpp/.h<br/>adrep.cpp<br/>ldapdir.cpp/.h<br/>ldaprep.cpp/.h<br/>-->
== phone: ip222,232: audio connection to remote conference peer sometimes lost after a coder renegotiation on one connection ==
This problem occured in the folllowing situation:<br/>- a local call (audio+video) was established via an USB headset and then put on hold<br/>- a consultation call (audio) to an external peer was established<br/>- a 3-pty conference was established but the local connection remained mute<br/><br/><!-- ac_phone_3.h ac_phone_3.cpp ac_codec3.cpp --><br/>
DTMF dial from a Waiting Queue to a Call Broadcast object with many destinations. This caused special timing in H.323, which created the problem.<br/><!--<br/>h323sig.cpp<br/>pbx.cpp (v9)<br/>signal.cpp (v9)<br/>signal.h (v9)<br/>-->
Switch from Media-Relay to No-Media-Relay when handling INVITE with Replaces.<br/>May result into no media after INVITE with Replaces.<br/><br/><!--relay.cpp-->
== PBX: OEM Registration licenses did not work anymore ==
Partner fkeys with subscriptions or favourites may not work in some cases.<br/>But only if partner's name is used as destination<br/>and if namesmatch partly.<br/>E.g. "name" and "name.x"<br/><br/><!--phonesig.cpp-->
== IP6000: Prevent blinking error LED on old IP6000 with HW-Build 201 ==
If there is a similar number in the LDAP directory, the number can be resolved in a wrong name. This is fixed now.<br/><!--<br/>Changed file:<br/>dectradio_mastercall.cpp<br/>dectmaster.cpp<br/>quickdb.h<br/>quickdb.cpp<br/>-->
Trying to comply to RFC-7044 and RFC-7131.<br/><br/>Decoding: Skip top-most entry "History-Info" (highest index value) if this entry reflects the called party itself.<br/><br/>Encoding: Add top-most entry "History-Info" (highest index value) that reflects the called party itself.<br/><br/>SIP header "Diversion" is removed since it is declared as deprecated (RFC-5806 Category Historic now).<br/><br/><!--<br/>sip.cpp<br/>siptrans.cpp<br/>-->
== OEM Registration licenses did not work anymore ==
Port the hash algorithm to our platform.<br/><br/><!--<br/>sha256.c<br/>sha512.c<br/>sha.h<br/>md32_common.h<br/>opensrc.mak<br/><br/>test_tls.h<br/>test_tls.cpp<br/><br/>11.00/unit-tests/tls/encryption<br/>--><br/>
The error correction doesn't work if it is necessary. It results in missed document parts or failed connections. This is fixed now.<br/><!--<br/>Changed files:<br/>facsimile_ECM.h<br/>facsimile_ECM.cpp<br/>-->
== PBX: CFB on Trunk or Gateway did not work if the call was cleared with DISC ==
A recall after a transfer should also use the twin phone algorythm. For example if one of the phones is busy, the call should be sent to the busy phones only.<br/><!--<br/>pbx.cpp<br/>-->
== SIP: New config option /send-deprecated-diversion-header ==
Diversion header is not sent anymore since v11r1sr5 / v11r2sr1 / v10sr24 / v9hotfix50.<br/>For interop reasons this config option is added.<br/>If set the old and deprecated Diversion header is sent.<br/><br/><!--<br/>sip.cpp<br/>siptrans.cpp<br/>-->
Font-family of input, select, textarea, button did not inherit body style.<br/>Using now "font-family:inherit" to have same font-familiy all over.<br/><br/><!--work.css-->
== IP-DECT: Fix for "Wrong name with reverse phone book search" ==
Since the fix "Wrong name with reverse phone book search" it doesn't work. This is fixed again.<br/><!--<br/>Changed file:<br/>dectradio_mastercall.cpp<br/>-->
== IP2x P30x IPxx10: Tone is sometimes not switched off ==
If an users logs in a handset and a previously used handset is logged out, the cipher key index for early encryption isn't saved for this handset. This is fixed now.
== CF/SATA driver: Disturbs Linux SATA driver at start-up ==
The innovaphone CF/SATA driver can disturb the Linux SATA driver at Linux start-up, Linux recognizes a spurious interrupt and disables wrongly the SATA interrupt. The SATA device doesn't work or works slowly. This is fixed now.
4409 - Fixed comparison of FTY_IM_MESSAGE in unit tests
Ticket #126338. In fty_event_im_message::is_identical fty_event_im_message::data was handled as a null terminated string but it is a bufman buffer.
3725 - SRTP: Better diagnostics for SRTP on IP6000 IP2000
Ticket #105043. !mem info srtp_socket
1436 - V9 und V8 PPP port configuration geht nicht richtig auf ip6010
Ticket #63427. Da erscheint in V9 bei Auswahl von PPPOE das ISDN Menu, dafür ist bei <none> jetzt kein weiterführendes Menü sichtbar. Das sollte auch bei V8 so sein.
Bei V8 erscheint derzeit auch bei <none> das ISDN Menu, dafür ist bei V8 das PPPOE Menü korrekt vorhanden wenn man PPPOE0/1 als Schnittstelle auswählt.
V9 Hotfix 1 (90600.01)
1562 - A Bootcode Update could disrupt the Media stream for some seconds
Ticket #64631. This was observed on phone devices with relatively slow flash memory when a bootcode update took place while a call was active.
1557 - A PBX user with "Full PBX Administration" Rights could not edit phone configuration
Ticket #64572. The configuration pages could be opened once but after changing an item the input was disabled
1656 - AD Replication, Alphanumeric symbols were rejected in maps
Ticket #65499. A bug introduced by code-adaption from v8 to v9. I.e. this problem is new with v9 and does not occur under v8.
1680 - Appended called number to CF at gateway object needs to be adjusted acording to destination node
Ticket #65754. Reroute from master to slave in an e164 scenario for IP down did not work
1639 - Associating a call from a Gateway object with Gateway object Node did not work if CGPN represented an incomplete Number
Ticket #65359. This can result in not finding the correct Node Extern interface
1563 - Bootcode: Support for RGB LED added
Ticket #64636. ...
1487 - Calling Tone garbled when using G722 with a framesize > 40 milliseconds
Ticket #64015. 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.
1454 - calling tone missing in some cases
Ticket #63667. 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.
1502 - Conference on IP6000 Hardware 200 and lower not working with v8hf14 and v9
Ticket #64132. The ADSP serial port has been changed from SPORT1 to SPORT0 for the IP6010. Old IP6000 hardware has the SPORT0 not connected, so now SPORT1 is again used on IP6000.
1455 - config download may trap when malformed LDAP config data has been uploaded
Ticket #63678. 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.
1723 - Config upload error message changed to show all skipped lines
Ticket #66133. Request from Ascom
1707 - debug.xml, Tracing Modified
Ticket #66004. A few trace options were altered -SIP -LDAP
1726 - DECT: Call waiting without Feature Codes
Ticket #66153. 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.
1759 - DECT: Configuration option for OEM device
Ticket #66425. New configuration option 'Output Power' in DECT Radio for an OEM device.
Ticket #66174. Now duplicate LDAP 'pbx' <gw> items are deleted by the DECT users module.
1691 - DECT: Device state info for OEM GUI
Ticket #65864. Device state info for OEM GUI added.
1521 - DECT: File upload for user import
Ticket #64264. File upload fixed. Used for user import by an OEM device.
1518 - DECT: forward info message to radio
Ticket #64245. 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.
1717 - DECT: Gateway menu missed
Ticket #66077. Now, the Gateway menu is added again in V9 for the IP1200.
1571 - DECT: Increase number of anonymous RAS registrations
Ticket #64761. Now the number of the allowed anonymous RAS registrations is increased to 30.
1529 - DECT: Internal endpoint information 'assigned-by'
Ticket #64316. 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.
1674 - DECT: License check for OEM
Ticket #65727. The DECT Master checks the DECT Radio license for an OEM device and PBX.
1532 - DECT: Location update option for UTF-8 support
Ticket #64371. 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.
1610 - DECT: Master Call GUI close/clear for OEM
Ticket #65137. The link 'close' and 'clear' of the DECT Master call page did not work for an OEM device and is fixed now.
1456 - DECT: menu in IP6000/IP6010/...
Ticket #63679. The DECT menu is visible in the IP6000/IP6010/IP0010/IP1060/IP3010 devices, if the required DECT multi-cell license is available.
1473 - DECT: OEM device user configuration
Ticket #63883. For an OEM device the user data handling must be different. This does not concern the IP1200.
1600 - DECT: Option not to use TLS
Ticket #65053. for easier debugging
1669 - DECT: Option short busy tone
Ticket #65693. New option 'Short busy tone' for the DECT system added. If this is enabled, the busy tone after remote hangs up is shortened.
1572 - DECT: Registration string for test registration
Ticket #64766. The registration string for the test registration is fixed to work with V9.
1670 - DECT: Trap during registration up handling
Ticket #65698. Trap in DECT Master fixed. It occurs if the master endpoint is in delete state and a RAS registration up event is received.
1527 - DECT: User import file convertion
Ticket #64297. Support for latin1 to UTF-8 conversion for user import added. The import feature is only used in an OEM device.
1603 - DECT: Wrong packetsize sometimes used after renegotiation
Ticket #65086. This could cause interop problems with some PBX, which was not able to receive media packets above a certian size.
1516 - DHCP configuration with secondary but no primary gatekeeper not possible
Ticket #64232. 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.
1528 - DNS/Hosts: SRV-Record, Port Default 5060 Confusing. Removed.
Ticket #64312. Edit field is now going be empty
1616 - firmware of a certain image size did not boot from scattered flash segments
Ticket #65154. 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.
1703 - Gateway SIP Interfaces: Could not configure internal registration for a disabled interface
Ticket #65975. and if a interface was disabled afterwards, the config for the internal registration was lost
1493 - Gateway: Allow interface maps for analog interfaces as well
Ticket #64068. Was prohibited in the past, but there are uses for this.
1650 - Gateway: Allow sending of Date/Time in Connect on ISDN interfaces
Ticket #65445. Was missing in the User Interface, so it could not be configured
1631 - Gateway: Configured signaling port got lost, when ediiting interface maps
Ticket #65303. The signaling port was reset to the standard port when saving interface mappings
1719 - Gateway: New interface option "No Early Media" for SIP interoperability
Ticket #66092. New config option "No Early Media" in section "SIP Interop Tweaks". Required for LYNC interoperability to ignore SDP answer in 183 Session Progress, since no inband progress information is provided by LYNC.
1630 - Gateway: Not possible to enter wildcards ('.') in interface maps
Ticket #65280. wrong check for correct value
1439 - Gateway: Outgoing Call Completion did not work when outgoing call was routed through TONE interface
Ticket #63517. Outgoing CC request did not went out to ISDN interface.
1649 - Gateway: Record URL at SIP interface was lost when Internal registration was configured
Ticket #65443. UI problem
1474 - GW Route Configuration: XML Content Not Well Formed When Inserting A Map
Ticket #63884. Could happen with certain types of routing maps
1497 - H.323 limit registration redirects to avoid high load on misconfiguration
Ticket #64116. 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.
1621 - H.323: Display call state in "Signaling Timeout" error log
Ticket #65194. To provide better indication about the nature of the problem
1609 - H.323: Don't forward G.729B capability if silience compreession not enabled
Ticket #65133. This solves quality issues some SIP provider have with G.729B.
1714 - H.323: Media Negotiation problem, with reverse media calls thru media-relay/exclusive media interface
Ticket #66037. This could result in no media with mobility calls to sip provider
1640 - HTTP /home did not work for URLs containing more then one '/'
Ticket #65364. /home could be used for PBX0/MY/start.htm, but this did nor work because of multiple '/'
1501 - HTTP server: Query string is lost on redirect for "Force HTTPS"
Ticket #64128. The location header of the redirect does not include the initial query string.
files: http.cpp, http.h
1438 - HTTP-Client: Bad encoding of uri parameter in digest authentication
Ticket #63469. Uri parameter in digest authentication was not URL encoded
1556 - HTTP-Client: Trap in minifirmware on requests using a domain name
Ticket #64568. The HTTP client does not check if the DNS module exists before using it. In the minifirmware there is no DNS module.
Ticket #66166. The processing of the configuration form accidentially produced a too long command line.
files: http.cpp, http.h
1728 - HTTP-Server: Restrict number of filters and paths on configuration page
Ticket #66167. 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.
files: http.cpp, http.xsl
1584 - Incorrect rpcap timestamp after TRACE LOST messages
Ticket #64915. The RPCAP timestamp (Wireshark) after a TRACE LOST message was incorrect, as the TRACE LOST message contained an incorrect timestamp.
1535 - IP200A IP230 IP240: manufacturing test not working
Ticket #64378. write fixed bootheader fails
1568 - IP222/IP232 Bootcode: Support for hardware 300
Ticket #64701. USB-Clock is generated from PRE-USB Clock, needs a 25 pre-divider
1667 - IP222: Alpha input using the num block
Ticket #65681. Alpha input using the num block
1594 - IP222: changes for new hardware
Ticket #64967. * Reduce average current through IR-LED * Re-init LCD orientation (fight against resets of the display controller)
1666 - IP232: Implemented touch keyboard
Ticket #65680. Implemented touch keyboard
1601 - ip241 - G722 3party conference did not work
Ticket #65078. 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
1520 - IP241 - VLAN frames > 1514 bytes were not accepted
Ticket #64251. The default maximum receive length of 1518 does not permit for a 1516 byte frame. With the maximum receive length set to 1536 such frames are received.
1452 - IP241 DSP may freeze under heavy load
Ticket #63648. 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.
1534 - IP241!: IP230X not working
Ticket #64376. wrong UART used
1629 - IP241: Call list does not show display name if call was diverted
Ticket #65269. Display name of the diverted-to party was not shown in call list. Also truncate names and numbers to avoid interference with date and time.
1661 - IP241: Connected call was displayed as transferred call
Ticket #65641. When name was dialed instead of number, the connected number (received with connect message) caused display to show transfer condition.
1551 - IP241: Directory entries did not show number tag
Ticket #64555. Number tag indicates type of number (M = mobile number)
1575 - IP241: Diverting/Transfering party was not shown
Ticket #64775. If name (not number) was called and call was diverted/transferred the diverting/transferring party (dialed name) was invisible.
1696 - IP241: Do not hide all fkeys in active state
Ticket #65923. Keep these fkeys on display that do not interfere with call controls.
1665 - IP241: Fix for display rendering
Ticket #65678. Zeroize png object when image loading fails.
1668 - IP241: Fix for multi line editor
Ticket #65686. Fix for multi line editor
1676 - IP241: Fix for XML screen dump
Ticket #65737. XML screen dump is only used for automated testing.
1530 - IP241: IP230X not working
Ticket #64355. wrong UART used
1590 - IP241: Multi line text ctrl must support line break
Ticket #64938. Multi line text ctrl must support line break when displaying received text messages or when composing new text messages.
1519 - IP241: Number information interfere with status information on call ctrl
Ticket #64250. Truncate number information to match screen metrics. Show status information if space available only.
1617 - IP241: Show active CFU destination on idle screen
Ticket #65176. Show active CFU destination in header bar of idle screen. Not only a symbol on status bar.
1645 - IP241: Time and date not refreshed while menu screen is active
Ticket #65394. Refresh staus bar in menu screen
1673 - IP241: Trap on display rendering
Ticket #65726. When entering too many digits while preparing block dialing.
1721 - IP3010/6010: fax problems
Ticket #66110. * CED is not transfered * Wrong T38 encoding in V8
Ticket #65905. only the counter names but no values were displayed on the ethernet statistics page
1711 - ip6010 - frame loss on ethernet ports running in a VLAN
Ticket #66028. 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
1699 - ip6010 - same MAC address was assigned to ETH0 and ETH1
Ticket #65939. this results in problems when both interfaces are connected to the same LAN segment
1491 - ip6010 family : CFLASH overcrowds trace buffer if trace flag is turned on
Ticket #64061. dto
1559 - IP6010, IP6000: Use optimized memcpy
Ticket #64587. Use of load/store multiple and shifts for 32 bit alignment speeds up memcpy by a factor of approx 2
CPU load with the test test/9.00/box/dsp/ip6010 shows approx 1% lower CPU load. Enet test test/9.00/box/enet/ip6010 shows 10638Kbyte/s transfer rate, compared to 9708Kbyte/s with the old memcpy.
With ECC enabled the CPU load was 19% / 21% without SRTP and 31% / 33% with SRTP With ECC Enet test test/9.00/box/enet/ip6010 shows 10638Kbyte/s transfer rate10309
1742 - IP6010-CF: Kingston compact flash was not recognized
Ticket #66269. the card was not recognized because a register was wrongly initialized.
1579 - ip800 trace telling wrong information about power source
Ticket #64826. PCBs since V300 cannot detect POE power and trace therefore told 'not powered'.
1752 - IPVA, Double-Fault Handling Should Skip Trace History
Ticket #66362. Otherwise a triple fault occurs in case of a corrupted trace buffer
1552 - IPVA, Include NAT
Ticket #64557. Include the NAT0 module as in other innovaphone products
1712 - IPVA, More Watchdog Info
Ticket #66029. a bit more diagnostics info is written in case of a watchdog
1593 - IPVA, RAM-Dependent Flash Limitation
Ticket #64961. Flash Utilization must be limited with RAM <= 64MB To prevent loss of flash data when uploading IPVA v9hf1: 1)Perform Maintenance/Config/Download 2)Upload the new ipva.bin 3)Perform Maintenance/Config/Upload with the step 1) config.
Note: -The flash directory space is 13MB with 64MB Ram -The max. flash directory space is available from on 192MB Ram
1733 - ISDN: New interop flag to forward network provided or checked cli only
Ticket #66183. Useful if the real calling number is needed and not a number provided by CLIP no screening
1751 - Kerberos: Disable RC4 encryption by default
Ticket #66352. RC4 encryption is now disabled by default. It can be turned on on the General/Admin page.
1658 - Kerberos: Do not allow registration of multiple databases for one realm name
Ticket #65589. This happened when a box hosted multiple PBXes with the same system name.
Ticket #64877. 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.
1577 - Linux: Reset required links
Ticket #64813. Reset required links in the Linux XML pages fixed.
1499 - Media negotiation problem H.323 to SIP call, with reverse Media and media relay
Ticket #64124. a call without Media in both directions could be the result
1542 - memory leak check missing for last parked call info
Ticket #64445. 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.
1694 - Message Waiting Interrogation: Result message coding wrong
Ticket #65912. a malformed message was displayed in wireshark
Ticket #64055. changing i.e. the ,NT-Mode' config option didn't show the 'reset required' link button after pressing 'OK'.
1679 - MWI does not work in various Node/Pbx combination
Ticket #65750. MWI does not work in various Node/Pbx combination
1506 - myPBX launcher: Hotkey does not work for same number two times in a row
Ticket #64169. files: TextPeeker.cs
1459 - myPBX launcher: Hotkey should always bring myPBX to front
Ticket #63718. Not only if a number is marked
Files: ctiwin.cs client.js
1465 - myPBX launcher: Map all variants of the Norwegian language to "no"
Ticket #63771. There are three ISO 639-1 language codes for the variants of the Norwegian languages.
Norsk (Bokmål) (nb) Norsk (Nynorsk) (nn) Norsk (no)
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.
Files: ctiwin.cs, start.js
1522 - myPBX launcher: Open configuration dialog if no URL is configured
Ticket #64265. On startup the config dialog should show up if no URL is configured, yet.
1498 - myPBX: Save button for contacts stays disabled when using autocomplete
Ticket #64117. This can happen using Firefox. Turn autocomplete off and use the oninput event additionally.
files: client_contact.htm
1460 - myPBX: Script error on line 1351
Ticket #63744. A script error is thrown when there is an unexpected missed-call-auth: client.js, Line 1351, this.manager is null
files: client.js
1613 - myPBX: Sessions sometimes get stuck
Ticket #65143. Session IDs used the full dword scope but btree_compare can only handle values in the positive integer scope.
files: pbx_client.cpp
1678 - myPBX: Show allows that are configured using templates
Ticket #65746. So that the user can see all access rights
files: pbx_client.cpp, client_allows.js
1511 - myPBX: Show login screen after logout instead of closing window
Ticket #64194. files: client.js
1596 - myPBX: Usability improvements for call transfers
Ticket #65032. Blind transfer of incoming alerting calls. Transfer of outgoing alerting calls. Autoselect new outgoing calls. Hide transfer buttons on the selected call.
files: client.js
1508 - myPBX: Use different icon for logout
Ticket #64184. So that users can better see the difference between logging out and closing the window.
files: window_action_logout.png
1672 - myPBX: Use displayname as sender name of instant messages
Ticket #65704. The cn was used before. Now one of the following attributes is used: dn, cn, h323
Ticket #65142. The network (ETH0/ETH1) Link page was not reloaded after pressing 'ok' or ' clear' in IP800 and IP1200 and is fixed now.
1636 - new License Mechanism <type>@<num>=n
Ticket #65339. For hosting provider
1466 - no dialtone and no audible connection on ip24, ip22 and ip28 with V9
Ticket #63780. even in both directions
1489 - No received cause code should be treated as 'normal clearing'
Ticket #64043. 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.
1488 - One-way audio when using G711u
Ticket #64036. Playback of G711u RTP audio may fail and error log "Wrong Payload Type received" is set in some cases.
1517 - Out Of Memory Trap when running VM without prompt files
Ticket #64243. When calling into a Voice Mail object without prompt files memory objects are allocated at high rate without being freed. Memory is freed at disconnect. This may cause a OOM trap when call stays connected for a longer time.
1715 - PBX Admin UI: When editing allowed access, show configuration inherited from templates
Ticket #66059. Result was sometimes confusing, because it was very hard to find templates from which the current object inherited config
1753 - PBX administration: Slave password is lost when configuring myPBX
Ticket #66380. When the configuration of myPBX was submitted on the PBX/Config/myPBX page the slave password was reset.
1628 - PBX Broadcast: CFB configured at broadcast was always executed if "Execute member diversions"
Ticket #65261. If "Execute Member Diversions" was checked a call to Broadcast was also sent to CFB destination
1713 - PBX Broadcast: CFNR was executed only after No Response Timeout even if no member
Ticket #66032. If there is no member in the broadcast group, a CFNR configured at the Broadcast object should be executet immediatelly.
This was a collateral damage from hotfix
65261: PBX Broadcast: CFB configured at broadcast was always executed if "Execute member diversions"
1544 - PBX CDRs: Dummy call to endpoint because of SOAP UserCall generated CDR
Ticket #64449. 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
1618 - PBX CDRs: Missing transfer-to on operator connect, when calling waiting from other PBX
Ticket #65178. This was a general problem with transfer on another PBX. Happens with normal transfer on other PBX as well
1592 - PBX CDRs: On Transfer to Waiting, the "connected" message did not contain peer
Ticket #64955. Happend with transfer with consultation to a waiting queue
1587 - PBX CDRs: Transfer not indicated if transfer initiated by myPBX
Ticket #64924. in case of transfer with consultation
1657 - PBX CDRs: Transfered call generated 2 cdrs. This was confusing
Ticket #65521. 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
1585 - PBX CDRs: Wrong Display Name sometimes with Pickup
Ticket #64917. when calls routed between locations were involved
1598 - PBX CDRs: Wrong information if forward on other PBX
Ticket #65041. The leg1 info from the other slave was not evaluated for CDRs
1738 - PBX External UC: Update presence of users with correct External UC set only
Ticket #66223. 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
1725 - PBX Mobility: Min-Alert/Max-Alert timer on configured mobility
Ticket #66152. This way it can be prohibited that an announcement from the mobile provider accepts a mobility call
1597 - PBX Mobility: Use of RTP-DTMF was prohibited even if media terminated in PBX itself
Ticket #65036. 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.
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
1483 - PBX Root Node Extern on Slave for calls from Master
Ticket #63995. If a Root Node Extern interface was configured on a slave this was currently only used in case this was a "License Only" slave.
Now this interface is also used for calls from the master with no destination found. This can only happen in non-replicated environments.
1642 - PBX UI: Registration with Admin Pwd misleading, PBX Pwd is better
Ticket #65370. The Admin Pwd could be understood as the General/Admin password. PBX Pwd should be pretty clear what this is.
1567 - PBX User Interface did not work with Groups containing XML reserved characters (&,<,>,...) or non-ascii
Ticket #64695. XML or URI encoding was missing in some paces. The browser could not display the page. This happend when using the left PBX/Group tree for nvigation
1689 - PBX User Interface: Text refers to pages PBX/General or PBX/Password which do not exist anymore
Ticket #65834. Should be pages PBX/Config/General and PBX/Config/Security
1570 - PBX v5 SoftwarePhones licenes did not work on v9 or v8 PBX
Ticket #64709. An old v5 SoftwarePhone license installed on a v9 PBX did not work for v5 SoftwarePhones
Ticket #65805. On an endpoint transfering a call two CDRs are generated: One for the original call and one for the call created by the transfer. The one created by the transfer has an event "transfered". This one can be omitted from a calls list.
1700 - PBX-SOAP: Don't provide caller number if CLIR was used on call to monitored endpoint
Ticket #65944. If this was an internal call, the PBX knows the calling number anyway, but it should not be sent on SOAP
1737 - PBX-SOAP: FindUser should not show hidden objects
Ticket #66216. Could be confusing
1646 - PBX-SOAP: Trap if initiating multiple outgoing calls from a Waiting object at the same time
Ticket #65418. Some applications do this to deliver voice messages
1702 - PBX-SOAP: UserDTMF did not send DTMF to Voicemail or Waiting Objects
Ticket #65958. It only sent DTMFs to a VOIP connection
1720 - PBX/Access: Visibility defined for a domain, should just add to the other definitions
Ticket #66104. Other then a definition based on a name, which should overwrite everything else
1647 - PBX: BC Conference instant messaging support
Ticket #65422. Instant messaging support with myPBX for the "Bc Conference" PBX object added.
1627 - PBX: Busy on ... Calls should be executed on calls from on object to itself
Ticket #65260. was disabled in 2008, no idea what this should be good for. Can be very confusing
1653 - PBX: Call-Intrusion could result in wrong name display
Ticket #65462. esspecially for silent intrusion
1472 - PBX: CDRs from Waiting Queue in case of call thru DTMF map wrong
Ticket #63882. The CDR just ended without the transfer and following events
1589 - PBX: Conference trap
Ticket #64936. A zero pointer trap in the broadcast conference PBX object is fixed.
1549 - PBX: Failover to WAN for calls to Slave PBX did not work
Ticket #64540. accidentally broken in v9
1494 - PBX: Group not configurable for PBX object
Ticket #64075. needed by SOAP applicatons (e.g. TAPI) to find slave PBXs
1558 - PBX: IM sessions to external UC did not work
Ticket #64583. The session was not initiated towards the external UC
1632 - PBX: Name Identification was missing when initaiting IM Session
Ticket #65304. No display name of initiating party was sent
1730 - PBX: Name not case sensitive anymore
Ticket #66172. 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.
1504 - PBX: Potential Trap on calls to exec, map or waiting object
Ticket #64135. under some rare circimstances, which are unfortunatly not known, there could be a NULL pointer access
1560 - PBX: Publish to external UC, From header wrong
Ticket #64591. A From header of anonymous was sent
1495 - PBX: Rej.External Calls not possible for all object types
Ticket #64111. when creating the objects it was sometimes possible to set this checkmark but later when editing the object the checkmark was not displayed anymore
1553 - PBX: Reporting licensing for standby operation did not work
Ticket #64560. Reporting did not work for endpoints registered in standby mode, because of license check failed
1686 - PBX: Routing did not work correct for calls between Nodes/PBXs for some Facility Calls
Ticket #65825. 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
1513 - PBX: Send 'Connected Name' in Connect
Ticket #64199. This is for example needed to display E-Mail Icon in myPBX for contacts in remote PBX
1492 - PBX: Transfer Recall timer was not started if destination was ringing after blind transfer
Ticket #64064. 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
1524 - PBX: Trap on security denial of 'unknown' registration
Ticket #64269. 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.
1573 - PBX:OEM Voicemail license did not work
Ticket #64769. collateral damage from supporting licenses from different versions on a PBX
1463 - phone ip241 - monitor mode (handset/headset + speaker) cannot be disabled once enabled
Ticket #63750. 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.
1505 - phone orchid: called party microphone muted after a G722 call has been connected
Ticket #64139. 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.
1543 - Phone trap when leaving User-Choose screen with Menu key
Ticket #64448. problem: Phone trap when leaving User-Choose screen with Menu key. Missing "SaveChages?" screen.
solution: fixed in code. TXT_EMPTY==SvgChg Id. Worst Case occured.
files: phone_user_config.cpp
products: All Telephones
risks: none
1611 - Phone UI: Truncate h323id on header of idle screen to reserve space for e164 number
Ticket #65140. Long h323id's may take all available space and keep e164 number from being displayed.
1625 - Phone UI: Truncate h323id on header of idle screen to reserve space for e164 number
Ticket #65258. problem: Long h323id's may take all available space and keep e164 number from being displayed.
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: 1) |123...| number too long 2) |123456| number just right 3a)| 12345| no name found or less than 2chars left for name 3b)| 1234| 4) |n... 1| name shortened with ellipsis, number complete 5) |nam 23| enough room for name and number 6) |name | no number - enough space for name 7) |name01| no number - enough space for name (just right) 8) |nam...| no number - name shortend with ellipsis
files: forms_impl.cpp
products: all inca telephones
risks: none
1660 - Phone: Bei "Nachricht antworten" wird von der gerade aktiven Reg gesendet
Ticket #65630. Es sollte aber von der Reg gesendet werden, auf der die Msgs empfangen wurde. Wird ja bei Voice-Calls auch so gemacht.
1638 - phone: blinking of central MWI-LED could not be disabled for a MWI function key without an own LED
Ticket #65354. 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".
1619 - Phone: Broken "Display Name on Pickup/Partner Key" Option
Ticket #65179. problem: Broken "Display Name on Pickup/Partner Key" : pickup function key always displaying numbers, never H.323 alias (name)
1709 - Phone: Changing current presence activity does not remove presence note
Ticket #66017. Remove presence note when changing presence activity.
1739 - Phone: Changing or resetting the presence activity leaves presence note as is - should delete it
Ticket #66246. 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.
solution: fixed in code, remove note upon activity chnage from function key or in user-menu
1741 - phone: DHCP provided configuration was not displayed under "Phone/User x/General/Options"
Ticket #66268. 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.
But instead of the active values the local values were displayed. 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.
1695 - phone: disable call intrusion via partner key when recording is active
Ticket #65918. Call intrusion cannot be performed while recording is active: - recording establishes a 3party conference between local party, remote party and recorder. - call intrusion establishes a 3party conference between local party and the two remote parties - 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.
Now if any kind of recording is configured call intrusion is neither offered in 'recall' menu nor performed via partner key.
1740 - Phone: Function key "Park" not displaying parked calls
Ticket #66258. problem: phone; Function key "Park" not displaying parked calls
1750 - phone: Hexadecimal values instead of descriptive texts were displayed for some rare disconnect causes
Ticket #66343. "0x57 - unknow cause" was displayed instead of "user not a CUG member". Mainly german descriptive texts were missing.
1484 - phone: improved czech display texts
Ticket #63998. now all texts are translated to czech, previous errors were fixed (translations provided by zakharova@annexnet.cz)
1675 - Phone: Message reply uses current registration instead of registration which received the incoming message
Ticket #65729. problem: Message reply uses current registration instead of registration which received the incoming message. Now message sending functions also take message source as parameter.
solution: fixed in code.
files: phone/app/* phone/list/* phone/dir/*
products: All Telephones
risks: none
1564 - phone: picking up a call failed sometimes
Ticket #64679. Sometimes pressing the partner, pickup or park key to pick up an alerting or parked call had no effect.
1710 - Phone: Pre-load more directory entries
Ticket #66026. Until to now the number of pre-loaded directory entries was defined by number of lines on screen. More entries have been loaded on demand when scrolling up or down.
1565 - phone: prevent the pc port of the ethernet switch from receiving frames directly from the phone firmware
Ticket #64689. 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 config add ETH0 /isolate-pc
1576 - phone: Secretary/Executive function of Partner key does not work when "Subscribe for Dialog Info" is checked.
Ticket #64783. 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.
1509 - Phone: trap upon deletion of user reg on pbx if registration information being viewed on the phone
Ticket #64186. problem: trap upon deletion of user reg on pbx if registration information being viewed on the phone
solution: fixed in forms code (unbind screen)
files: box/forms/lcd/forms_impl.cpp
products: phones
risks: none
1546 - Phone: trap when redefining function key (long press access) from destination to partner
Ticket #64498. problem: Phone traps when redefining function key (long press access) from e.g. destination number to partner
solution: fix in code
files: phone_user_func.cpp
products: All Telephones
risks: None
1580 - Phone: When activating a callback through function-key, the stored number behind the function-key is also called
Ticket #64830. 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
solution: fixed in code: now ignoring function key invocations through function-key-release if popup active
files: app_fkey.cpp forms.* forms_impl.*
products: all telephones
risks: none
1644 - phone_orchid: handsfree mode does not work with DSP Fimware 494E003ce4.640.11
Ticket #65379. the calling party was fully echoed when the called party was in handsfree mode.
1510 - PPPOE: specific configuration not reachable from config web page
Ticket #64192. 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
solution: fixed
files: ip_pppif.xsl (now check for PPPOE0, PPPOE1 and PPPOE2 types)
products: all (gateways effected)
risks: none
1461 - Presence functionality is not available when registered via H323 at a non-innovaphone PBX
Ticket #63745. 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. Thus no presence control calls must be sent to such a PBX.
1477 - Preserve number of calls on SIG_CONN without connected number
Ticket #63906. On calls across PBXes the number is discarded when the SIG_CONN does not contain a connected number.
1496 - propagating comma separated option values to multiple device dependent config lines did not work
Ticket #64114. 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.
1716 - QSIG: Avaya expect Progress Indicator with external calls
Ticket #66074. 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)
1507 - SDP: Wrong clock rate for G.722 codec
Ticket #64182. RFC-3551 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.
1595 - Setup wizard used wrong default password on some products
Ticket #65024. If no password was entered on - ip6010 - ip2010 - ip1060 - ip302 - ip305 - ip6000 - ip2000 A wrong password was configured
1620 - SIP-H323 calls with SRTP: No media after multiple Hold/Retrieve
Ticket #65185. After first Hold/Retrieve there was no SRTP, after the next Hold/Retrieve very often no media
1623 - SIP: add an "alias" header field parameter to the Via header field
Ticket #65235. According to RFC-5923 to allow TLS connection re-use
1470 - SIP: Allocated message size to small for INVITE redirect response (Avaya)
Ticket #63829. Memory allocation is a bit to tight to fit the message due to many Via headers.
Ticket #65880. Problems handling Event Notification Extension for Resource Lists. Bodies can grow very large.
1732 - SIP: Decode application/rlmi+xml for presence from external UC server
Ticket #66175. Decode application/rlmi+xml for presence from external UC server
1690 - SIP: Fix for SDP answer to SDP offer with "a:inactive"
Ticket #65863. Interop with CUCM. Should return RTP/AVP(inactive) if offer was RTP/AVP(inactive). Not not RTP/SAVP(inactive).
1648 - SIP: Gateway interfaces without registration need another config option
Ticket #65436. New config option "Local Domain" added. Required for URI processing on incoming calls. If domain part of destination URI matches the local domain, the domain part is stripped from the destination information.
1701 - SIP: Handling of NOTIFY(message-summary) on gateway interfaces without registration
Ticket #65946. Required for LYNC interaction
1548 - SIP: Handling on INFO inside instant message session
Ticket #64535. INFO sip:xxx@172.16.0.200:5060;transport=TCP SIP/2.0 Via: SIP/2.0/tcp 172.16.1.61:5060;branch=z9hG4bK.DF25F93ADB524A283550ECBD871BC73B From: <***>;tag=902D6B1B5622453B000172A9724270D4;epid=00000003 To: <***>;tag=1440372429 Call-ID: 5D4FA4260D42445F4071AEA032D4321B CSeq: 3 INFO Contact: *** Content-Type: application/xml Max-Forwards: 70 Supported: timer Content-Length: 82
1550 - SIP: Locating remote federation server with another SRV query
Ticket #64549. There are different service name decorations in use _sips._tcp.<domain> and also _sipfederationtls._tcp.<domain>
1705 - SIP: Media negotiation problem during hold/retrieve
Ticket #65995. Media negotiation problem during hold/retrieve on local media
1435 - SIP: Media negotiation problem during transfer to early media source
Ticket #63422. test\9.00\relay\early-media failed
1503 - SIP: Memory leak when receiving small datagrams on SIP port
Ticket #64134. Some SIP clients send small UDP datagrams in order to refresh mapping on NAT devices.
1440 - SIP: Message buffer too small for REGISTER request for re-try with authentication
Ticket #63539. On some installations a change-of-nonce at server side may cause volatile "Registration down error" on client side.
1633 - SIP: New config file option /add-cn-capability
Ticket #65313. Required for mediation server (lync) interoperability. Otherwise mediation server complains: "The Gateway peer does not support comfort noise"
1602 - SIP: New config file option /no-alert-info
Ticket #65084. Supress Alert-Info header for interoperability with Thomson ST2030 and Thomson TB30
1683 - SIP: New config file option /no-app-rerouting
Ticket #65801. Interoperability with Ingate Session Border Controller. Required when handling redirect responses.
1444 - SIP: Not accepting calls from alternative proxy
Ticket #63600. When being registered at a proxy with 2 ip addresses the gateway does not accept calls from the alternative ip address.
1526 - SIP: P-Asserted-Identity in UPDATE not working
Ticket #64289. PAI with changed remote party identification was not handled if also Session-Expires header was present in UPDATE request.
1426 - SIP: Session refresh was taken as session modification
Ticket #63310. Local SRTP key was re-calculated after re-INVITE for session refreh was received. Causes SRTP decode error at remote side. CUCM scenario
1697 - SIP: Set CLIR if display string of From-URI contains "Anonymous"
Ticket #65925. Not only if userpart of From-URI contains "anonymous".
1405 - SIP: SRTP re-keying when retrieving held call
Ticket #62620. Required for Cisco interoperability
1547 - SIP: Trap when mypbx changes presence and external uc is configured
Ticket #64504. Trap when mypbx changes presence and external uc is configured.
1704 - SIP: Trap when receicing provisional response with RSeq header
Ticket #65986. Trap when trying to send PRACK
1566 - SIP: Treat "Privacy:off" like "Privacy:none"
Ticket #64692. Treat "Privacy:off" like "Privacy:none" when receiving INVITE
1607 - SIP: Use DNS query without recursion on closed federation interfaces
Ticket #65126. Keep DNS server from doing automatic recursion when resolving domain names on closed federation interfaces.
1754 - SOAP, Send leg2Info.originalCalled Info
Ticket #66407. As CallInfo.No with type="leg2orig"
Ticket #63902. Sometimes the SoftwarePhone 3pty conference is not working. It is fixed now.
1467 - SoftwarePhone: de-registration at application end
Ticket #63781. The application sends a de-registration to the PBX, if the user closes it.
1447 - SoftwarePhone: default configuration
Ticket #63632. The default configuration is changed to coder G.711a and license type PBX.
1734 - SoftwarePhone: GN 8120 USB volume keys
Ticket #66204. The volume up and down key mapping for the GN 8120 USB is fixed.
1681 - SoftwarePhone: Jabra BIZ 2400 support
Ticket #65755. Now the Jabra BIZ 2400 is supported by the SoftwarePhone.
1688 - SoftwarePhone: Jabra LINK 320 USB support
Ticket #65833. Now the Jabra LINK 320 USB is supported by the SoftwarePhone.
1698 - SoftwarePhone: Jabra PRO 94xx support
Ticket #65933. Now the Jabra PRO 9400 serie is supported by the SoftwarePhone.
1687 - SoftwarePhone: Jabra SPEAK 410 USB support
Ticket #65832. Now the Jabra SPEAK 410 USB is supported by the SoftwarePhone.
1448 - SoftwarePhone: license information
Ticket #63633. If no valid license is received from the PBX, a information is printed directly now.
1445 - SoftwarePhone: reloading current configuration
Ticket #63607. The device configuration is correctly loaded in setup.
1446 - SoftwarePhone: ringing through device
Ticket #63608. Ringing through device fixed, if there are more output devices than input devices.
1539 - SoftwarePhone: Setup about logo
Ticket #64414. The setup about innovaphone logo was changed.
1450 - SoftwarePhone: trap with busy external call
Ticket #63642. If a call to external is busy and is hooked off with myPBX, a trap occurs. This is fixed now.
1512 - SoftwarePhone: Tray Icon
Ticket #64198. Now the SoftwarePhone runs as Windows application with an tray icon.
1531 - supress "Send Number" for calls triggered by a 'Dial' function key with 'Send as Control Call' checked
Ticket #64365. 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.
1554 - Timeout when calling Mobile endpoint which does not send alert
Ticket #64563. 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
1569 - TLS: Error on processing huge handshake messages
Ticket #64702. 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.
files: tls.cpp
1635 - Trap in rarely used OS function bufman::remove
Ticket #65338. could result in negative length of buffer
1464 - Trap when starting from flash_stick
Ticket #63752. and flash memory not yet programmed with bootcode
1457 - Trap with conference (ADSP driver)
Ticket #63706. 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.
1682 - Trap: When Dectmaster registers user at PBX using SIP protocol
Ticket #65798. Occurred on IPBL[4.1.22]
1536 - Truncating UTF-8 aware string copy function
Ticket #64380. New string library function to correctly truncate an UTF-8 string to fit in a buffer.
1533 - Trunk Park/Pickup (line keys) did not work anymore
Ticket #64373. Collateral damage from fix
fix: #61590: PBX: Boolean Function Key was not updated when joining group
1555 - User Interface: OK, Cancel, Apply buttons inconsistent
Ticket #64567. The sequence is sometimes different. Should always be OK, Cancel, Apply
1537 - VM, <prep-callinfo> doesn't apply with <pbx-xfer>
Ticket #64384. VM, <prep-callinfo> doesn't apply with <pbx-xfer>
1652 - VM, email attachments weren't sent for https URLs
Ticket #65458. VM, email attachments weren't sent for https URLs
1651 - VM, Project script didn't run for endpoints having "Send Number" configured
Ticket #65456. VM, Project script didn't run for endpoints having "Send Number" configured
1604 - VM, Session Guard Timer Not Stopped For <pbx-record>
1525 - X.509: Use random serial numbers for self-signed certificates
Ticket #64279. 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.
Use a random number in order to avoid collisions after long reset.
Files: x509.cpp, x509.h
V9 Hotfix 2 (90600.02)
1807 - additional choices for 'loop current' on ip22/24 ip302 and ip28 gateways
Ticket #67396. for special applications higher loop currents above 26mA are requested. Implementation adds 30mA, 35mA and 40mA values.
1784 - Allow configuration of SRTP crypto suite, to be used for media proposals
Ticket #66810. This is needed, because some equipment (e.g. Microsoft Lync) is not able to adapt to whatever proposal it receives
1811 - Compact Flash: possible trap when deleting directories
Ticket #67456. There has been a chance for a trap when deleting directories.
1762 - configurable DHCP client hostname to be sent in DHCP requests
Ticket #66487. 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. The MS DHCP server creates a DNS entry from this name and the assigned IP address. A configurable name may better meet existing naming conventions.
1786 - DHCP-client monitors ethernet link down/up events and revalidates current lease after link up
Ticket #67006. This prevents problems when a device is hot plugged to another network. Further this helps to overcvome a problem with certain cable modems.
1796 - Fax: Progress informations
Ticket #67228. The FAX interface provides progress and error informations, now.
1771 - Gateway: Trap in case of collision of hold and clearing from remote
Ticket #66642. This could happen on gateways with analog interfaces if the R-Key was pressed right when the other side hung up
1764 - H.323 Media Re-Negotiation Problem if reverse Media call to Media-Relay/exclusive Coder Interface
Ticket #66547. 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
1772 - H.323 potential trap if AlertingNumber is received
Ticket #66710. 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
1792 - IP110 enters boot loop after power cycle with v9hf1
Ticket #67158. ... other phone_incas like 200a 230 240 are not affected.
Ticket #67304. The new DSP code increases the needed memory, on boxes PBX (IP302/IP305) the memory may be already low. So we split the binary with the extra IP28 DSP code to save some bytes.
1795 - IP241: Disable LINK LEDs not implemented
Ticket #67225. on IP240 this function is available
1791 - IP241: embedded test: Function key LED test didnt work
Ticket #67142. init was missing
1756 - IP2x2x: Reset USB Hub, OHCI and LCD during softreset
Ticket #66418. was missing on softreset. Also enable/disable USB power added
1801 - IPxx10 traps when clicking on Linux stop
Ticket #67314. wenn Linux gestopped wird, trapt das Gerät
1789 - myPBX launcher: Accept language code from configured URLs
Ticket #67137. Use the language code from the URL (lang=it), if configured. Otherwise use the language of the current windows login, like in prior versions.
1790 - myPBX: Workaround for redrawing problems in IE9
Ticket #67138. IE9 does not redraw the presence note when the activity image stays the same.
This can be fixed by avoiding to replace an image or text by the same image or text.
1767 - PBX CF Filter for external calls did not work as expected in case of chained CFs
Ticket #66599. A filter for external calls did not match if the external call was forwarded already by an internal user
1793 - PBX CF Filter worked different if previous divertion with CFU or CFNR
Ticket #67212. This was confusing
1769 - PBX-CDR: Generate CDRs for Broadcast objects
Ticket #66628. No CDR was generated from Broadcast object.
1774 - PBX-CDR: In CDRs generate for a Gateway object with prefix, the prefix was removed from the number
Ticket #66719. 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.
1779 - PBX-SOAP: Don't show dummy call, which is generated by UserCall
Ticket #66748. 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
1766 - PBX-SOAP: Support for Fax Server interface
Ticket #66590. UserFindDestination function added
UUI Protocol Discriminator fixed
1782 - PBX-SOAP: Support receiving of UUI
Ticket #66758. needed for Fax Server
1765 - PBX-SOAP: When monitoring a Gateway object, dialing information received with overlap receiving was missing
Ticket #66559. Applications, which monitor a gateway object, and want to connect the call depending on the number dialed, need this information
1794 - PBX: Syslog for registration attempt to unknown user wrong
Ticket #67224. The reason for the reject was provided as "Missing Authentication". It is now changed to "User unknown".
1788 - phone orchid: Audible Signal after alerting for partner/pickup key did not work
Ticket #67094. audible signal was triggered but not played
1787 - phone orchid: DTMF digits typed in handsfree mode sometimes duplicated when speaker volume is high
Ticket #67092. 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. Now this detection is disabled while typing DTMF digits.
1773 - phone: local/default coder config was overriden by unsupported or wrong DHCP config
Ticket #66716. DHCP provided coder config was not checked before overriding the locally defined configuration or the default values. Now a ';' separated list of coder configurations can be specified as well for the general as for the local coder, for example G729A,60,x;G723-53,60,x/G722,30;G711,30 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.
1798 - possible noise in PRI connections with ip6010 ip3010 ip1060
Ticket #67302. some few gateways may produce noise when using the PRI ports. This can be fixed with a new CPLD code contained in future firmware.
1770 - Relay: Forward facilities to local destinations
Ticket #66629. The relay forwards all facility events if interworking is disabled and the destination is local. This is needed for some interfaces.
1777 - Release not forwarded in quick dial object
Ticket #66728. If the called party released the call, the remote party didn't get the release.
1778 - SIP: Always add payload type 13 (CN) to SDP
Ticket #66735. Config file option /add-cn-capability is obsolete and replaced by /rem-cn-capability now.
1775 - SIP: DNS resolving failed on no-reg interfaces if DHCP takes longer than 10sec from boot
Ticket #66720. DNS resolving does not work as long as local IP config is not complete. DNS resolving is part of initialization process of SIP interfaces. SIP interfaces give up on DNS after 10 secs of retrying. Local IP config may take longer if DHCP is used.
1808 - SIP: Handling if transport errors on connection oriented transport types
Ticket #67398. Handling of transport error was added for SIP/TCP and SIP/TLS. Transaction timeout (32secs) is no longer required to elaps.
1780 - SIP: Lync interoperability
Ticket #66750. Lync uses same Rseq number on several provisional responses. Must send PRACK for each of it.
1810 - SIP: NOTIFY sent after 302 moved temporarily
Ticket #67450. After processing "302 moved temporarily" on an outbound call a NOTIFY (sipfrag) was sent.
1804 - SIP: Presence: Support for presence/person/activities/note
Ticket #67363. Not only decode presence/person/note
1768 - SIP: SIP/TLS registration was not accepted
Ticket #66625. Inbound SIP/TLS connection was not accepted since client certificate was checked. Usually client certificates are self signed and not trusted by innovaphone PBX. Mutual TLS was done.
1805 - SIP: Trap when receiving unsolicited NOTFIY with dialog-info
Ticket #67368. Trap when not subscribed for dialog-info
1783 - SoftwarePhone: Muting ringer from context menu
Ticket #66801. 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.
1800 - Web-UI: Vertical alignment of ui elements
Ticket #67313. Diagnostics/Counters and others
1755 - X.509: Add key usage to certificate requests
Ticket #66413. The Microsoft CA (standard) does not write the key usage into the certificate if it is not specified in the request.
V9 Hotfix 3 (90600.03)
1923 - ac_dsp3: support echo canceller trace
Ticket #68649. required to analyze echo canceller problems
1909 - body onload attribute can be extended and tab_active method doesn't crash anymore
Ticket #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.
1825 - Call to Voicemail did not work anymore with SRTP
Ticket #67672. This was a collateral damage from fix
1812 - DECT: IP6000/IP6010/... default config Master mode off
Ticket #67479. Now the Dect Master is in mode off by default for the IP6000/IP6010/...
1939 - Denial of Service filter in ethernet library did not work
Ticket #68907. this filter can be useful to prevent DOS attacks on non routing devices
1903 - DTMF user configuration with invalid checkbox check for presence setting
Ticket #68383. The check of the checkmark of the presence setting was wrong.
1882 - Gateway: Allow configuration of username and password for ENUM/SIP interfaces
Ticket #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.)
1839 - Gateway: Interface Name with multiple "Umlaute" did not work
Ticket #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
1887 - Gerneral/Admin page was broken if too many authentication servers were configured
Ticket #68231. The number of authentication servers is now restricted to 10.
1847 - H.323 Malformed packet
Ticket #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
1846 - H.323: A name_id of length 0 resulted in invalid H.450 coding
Ticket #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'.
1916 - HTTP-Client: Allow user names longer than 16 characters
Ticket #68499. Now user names with up to 64 characters are allowed.
1853 - improved test for Timeslot Switch Chip of ip6010 ip3010 ip0010 ip1060 and ip6000
Ticket #67840. intension is better analysis of hardware problems
1900 - IP-DECT: Adding OEM radios to Kerberos realm did not work with passwords containing special characters
Ticket #68377. The password was not URL-decoded when reading it from the UI.
1918 - IP-DECT: Allow setting empty text for idle display
Ticket #68553. Some handsets will not show signal strength and battery symbols if idle display is defined, they must be set to empty string.
1859 - IP-DECT: Changed channel option SRTP crypto suite
Ticket #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.
1952 - IP-DECT: Configuration of Media preferences did not work anymore
Ticket #69056. Media preference were ignored. This was a collateral damage introduced with fixes from the last hotfix
1857 - IP-DECT: Configuration XML data for OEM device
Ticket #67858. A new configuration XML attribute is added for changing a OEM GUI.
1874 - IP-DECT: OEM registration string
Ticket #68058. The Radio registration string of an OEM device is changed to check a OEM license in the DECT Master.
1840 - IP-DECT: Packetization could change after handover
Ticket #67738. On the new radio the RTP should be sent using the same packetization as was negotiated with the original call
1907 - IP150: dimming of key LEDs and LCD backlight
Ticket #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.
1870 - IP150: OEM specific WEB GUI modifications did not work
Ticket #67992. manufacturer specific stylesheets have to be be adjusted to the new GUI style
1823 - ip200a/230/240: handset conversations can be monitored in a directly connected headset
Ticket #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
1849 - ip22/24/28/302/305: DSP affected by certain fax tones
Ticket #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.
1946 - IP222/IP232: Hookswitch on some PCB not working
Ticket #69001. Reduce LED frequency from 12kHz to 6 kHz
1905 - IP222: Alpha input using the num block
Ticket #68398. Automatic switching back to numeric mode when leaving input ctrl.
1912 - IP241 - handsfree speaker volume to low
Ticket #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
1897 - IP241: Activate external background image from phone menu
Ticket #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.
1885 - IP241: Fix for display rendering
Ticket #68181. Display of diversion destination was corrupt. Pickup fkey labeling was wrong when party information needed truncation.
1894 - IP241: New fkey backgrounds
Ticket #68304. Less transparency to avoid interference with personalized background images
1858 - IP241: Remote party information truncated more than necessary
Ticket #67872. Line 3 of call ctrl
1889 - IP241: Updated some display text information
Ticket #68234. No need to strongly abbreviate disconnect cause text.
1854 - IP2x2: Handset microphone gain increased
Ticket #67843. changed from 22.5db to 30db
1938 - IP30x V9 hf2 media problem: Echo canceller not reliable
Ticket #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.
1850 - IP30x, IP1060, IP2010, IP6010: Fax did not work if rerouted from ISDN interface to a Voip destination
Ticket #67823. DSP was not configured correctly
1797 - IPxx10: error handling in sata driver
Ticket #67229. Old cards are producing DMA errors that were not handled properly. Try again read/write operation after error recovery.
1843 - Leak checking improved
Ticket #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.
1911 - License: Be safe against factory reset during license invalidation
Ticket #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.
1863 - myPBX: Always send an ID with CT_INITIATE
Ticket #67932. This is needed for interoperability with phones using v9hotfix2 or older. (See #67803)
1895 - myPBX: Chunked response from application sharing provider did not work
Ticket #68311. After receiving HTTPCLIENT_RECV_RESULT, pbx_client did not send another HTTPCLIENT_RECV.
1898 - myPBX: Hide passwords for application sharing and reporting in config
Ticket #68352. The passwords are moved from the config line to the VARS.
1948 - myPBX: Login did not work with system names containing special characters
Ticket #69033. Missing URL encoding when sending the connect-request message.
1831 - myPBX: Show active sessions on the admin interface
Ticket #67708. The active myPBX sessions are shown on the page PBX/myPBX.
1883 - myPBX: Support for WebEx meeting passwords
Ticket #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.
1942 - myPBX: Use display name for sending meeting urls and conference numbers
Ticket #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.
1893 - myPBX: Wrong charset was used when authenticating against reporting or changing user password
Ticket #68258. Javascript uses Unicode. The PBX and the reporting use UTF8. Therefore strings have to be converted to UTF8 before encrypting them using RC4.
1951 - PBX Mobility: Mobility object hanging if call establishment with SOAP/myPBX canceled
Ticket #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.
1852 - PBX Trunk/Gateway: Round robin within registrations to same device, different devices sequentially
Ticket #67835. This way both round-robin or sequential usage of gateways can be configured
1834 - PBX Trunk: Problem with Forking to trunk if multiple GWs are registered to Trunk
Ticket #67720. If one of the gateways rejected the call (no channel, not connected, ...), the original call from which was forked was disconnected
1866 - PBX Waiting: A call rejected on a primary operator was never sent to all operators
Ticket #67945. if again and again rejected on all primary operators. Problem happend also if rejected by "Do not disturb" on operator phone.
1827 - PBX Waiting: Call forwarded with DTMF mapping was shown in myPBX for each registration
Ticket #67682. The call was not sent with the original conferenceID, so myPBX could not detect that it was in fact the same call
1869 - PBX-CDR: Mobility calls to Trunk, external number was missing in CDR
Ticket #67978. only the number of the trunk itself was available
1873 - PBX-SOAP: Support UserClear for pending outgoing calls on Waiting Queues
Ticket #68009. UserClear was ignored in this state
1851 - PBX: Allow CFB on Gateway Type Objects
Ticket #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
1945 - PBX: Conference trap
Ticket #68998. A zero pointer trap in the broadcast conference PBX object is fixed.
1871 - PBX: End of call intrusion was not signaled to the phone
Ticket #68007. The call intrusion tone was generated even if the intrusion was terminated
1837 - PBX: License accounting did not work with Unknown Registrations under some special conditions
Ticket #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
1829 - PBX: License accounting in centralized licensing scenario wrong if master not available
Ticket #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.
1867 - PBX: Master/Slave compatibility problem with version 9 and version 8 and non-ascii characters in PBX name
Ticket #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.
1842 - PBX: New presence activity "do not disturb"
Ticket #67777. If a users sets this presence activity no calls are passed.
1824 - PBX: Translation of Cause "Call Rejected" to Cause "User Busy" for endpoint objects only
Ticket #67668. If the cause is received from a gateway, it is forwarded transparently now
1872 - PBX: User Interface Inconsistency. Same thing was sometimes call 'Response Timeout'/' and 'No Response Time'
Ticket #68008. Only Response Timeout used anymore
1860 - Phone: "Function keys not modifiable on the phone" mask should disable creation of new function keys of masked type
Ticket #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.
1933 - Phone: "Keep Calling Party Info on Pickup Key" option from Phone->Preferences not working, fixed
Ticket #68820. Phone: "Keep Calling Party Info on Pickup Key" option from Phone->Preferences not working, fixed
1865 - Phone: Added command line option to hide Administration Menu and/or MAC/Serial completely
Ticket #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.
1890 - Phone: Added loud note to web-ui that bool funnction key can only toggle if boolean object addressed by number
Ticket #68240. Description: Phone: Added loud note to web-ui that bool funnction key can only toggle if boolean object addressed by number
1836 - Phone: Added new function keys to Phone-UI (Toggle & Prepare Override)
Ticket #67729. Description: Phone: Added new function keys to Phone-UI (Toggle & Prepare Override)
1821 - Phone: Automatic key repeat did not work on IP222/IP232
Ticket #67661. Automatic key repeat did not work on phone w/o alpha keyboard
1917 - Phone: Call forwarding (always, busy, no reply) destination now choosable from dial-menu
Ticket #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.
1835 - Phone: Calls received with CLIR appear in call list with an empty entry
Ticket #67722. Description: Phone: Calls received with CLIR appear in call list with an empty entry; now fixed to display the CLIR text: anonymous/unbekannt/...
1929 - Phone: Changing config option /sip-hold does not call for reset
Ticket #68691. Reset is required and 'reset required" must be displayed.
1832 - Phone: CLIR on text messaging did not work
Ticket #67710. CLIR on text messaging did not work
1919 - phone: DHSG headset not reset to idle after a hookswitch signal in idle state
Ticket #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.
1937 - Phone: DnD Absence Message containing newline dumped to config-file breaks phone upon config-file upload
Ticket #68898. Solution: All flashdir entries cotaining newline now dumped as binary.
1845 - Phone: Enable "Activate Registration" without user/password authentication if "Protect Configuration at Phone" set
Ticket #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.
1904 - phone: Finnish translations updated
Ticket #68395. no english placeholders anymore, Texts begin with upper case letter
1943 - Phone: Hide calling party on Pickup key fixed
Ticket #68995. Phone: Hide calling party on Pickup key fixed
1891 - phone: intrusion call started in handset mode is not terminated when going on hook when TAPI or operator run on PBX
Ticket #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.
1826 - Phone: long function key titles hide idle screen information
Ticket #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)
1818 - Phone: Main menu scrolling below last item broken
Ticket #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).
1886 - Phone: Message function key
Ticket #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.
1931 - Phone: On IP240, make OK key a headset activation key (along with Space Key)
Ticket #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.
1892 - Phone: Switch presence fkey light on if presence activity is set
Ticket #68256. Switch presence fkey light on if presence activity is set
1878 - phone_inca: "ETH0/Isolate PC Link" checkmark could not be cleared via WEB UI once set
Ticket #68098. Only a WEB UI problem, a "config rem ETH0 /isolate-pc" did help.
1947 - phone_orchid - builtin speaker test did not work
Ticket #69026. works only over channel 0
1908 - phone_orchid: begin of voice mail prompt was cut off sometimes
Ticket #68416. Connection of Voip to DSP channel was delayed and thus the first packets of the RTP stream were lost.
1856 - phone_orchid: builtin color display test
Ticket #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
1896 - phone_orchid: call pickup via partner/pickup key fails when the key is pressed immediately after audible signal
Ticket #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.
1876 - phone_orchid: DTMF Tones detected in voice data from microphone were propagated to remote side
Ticket #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.
1830 - phone_orchid: microphone is not mute on a call intrusion in silent monitoring mode / microphone cannot be muted in a conference
Ticket #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.
1920 - SIP/TCP: Transport error when connection is closed by client
Ticket #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.
1881 - SIP: Diagnostics of transport error "Remote server certificate mismatch"
Ticket #68142. If remote sverer certificate does not match the destination domain name signaling connection is refused by client.
1861 - SIP: DNS problem when SRV response provides no additional records
Ticket #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.
1899 - SIP: Failed to register on dynamic PBX
Ticket #68362. Attempt to register on a dyn PBX with SIP protocol was rejected with "301 Moved Permanently".
1816 - SIP: Fast re-routing on gateway interface w/o registration
Ticket #67593. OPTIONS is used to check availability of remote peer. If OPTIONS fails, not trying to send INVITE.
1921 - SIP: Fix for Dialog-Info notification
Ticket #68581. Send an empty dialig-info XML after inbound subscription. Required for interop with Grandstream GXP2010.
1841 - SIP: Fix for early media from Waitng Queue
Ticket #67775. PROGRESS after ALERT was not handled by SIP stack. Now 183 Session Progress with SDP is send after 180 Ringing w/o SDP.
1875 - SIP: Fix for media negotiation with SRTP
Ticket #68067. Exchange of SRTP key may fail after hold/retrieve
1927 - SIP: Fix for REFER handling
Ticket #68663. Support for attended transfer between two calls from/to different SIP endpoints. As long as both calls belong to the same signaling interface.
1868 - SIP: Generate new SRTP key on every incoming re-negotiation
Ticket #67958. Required to have different encryption after transfer.
1844 - SIP: Generate/add SRTP key on media-relay interfaces
Ticket #67789. This enables even partial SRTP (SRTP on one side of media-relay).
1924 - SIP: Handling of 491 response on reliable transport
Ticket #68652. Re-try of re-INVITE was missing.
1884 - SIP: Interoperability with LinkSys SPA3102
Ticket #68174. LinkSys SPA3102 gives "g729a" as RTP payload type mapping:
Ticket #68232. Media negotiation problem on calls coming from on Lync client and getting forwarded to another Lync client.
1820 - SIP: Interoperability with Lync and media-bypass
Ticket #67645. Ack contained wrong To-Tag when calling a lync client in media-bypass scenario. Results into call drop after 30 seconds.
1833 - SIP: Interoperability with Lync: Handling of REFER from Lync
Ticket #67713. REFER for blind transfer was rejected with "406 Not Acceptable", due to absence of user part in refer-to URI.
1864 - SIP: Must answer every request - even unknown/unsupported methods
Ticket #67935. Lync sends proprietary NEGOTIATE request and waits for response. Otherwise transport connection is blocked for any upcoming request.
1822 - SIP: New config file option /no-ms-acceptedby
Ticket #67665. Some IP phones get confused by this Microsoft extension. Cisco SPA 303 and Cisco SPA 962 don't stop ringing anymore
1932 - SIP: Out-Of-Memory trap if Group Indications are enabled
Ticket #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.
1901 - SIP: Pending control calls on gateway
Ticket #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.
1928 - SIP: Problem decoding INFO(application/dtmf-relay)
Ticket #68667. DTMF digit was not decoded from message body if whitespace between EQUAL and DIGIT. E.g. Signal= 5
1813 - SIP: Send OPTIONS at configurable interval
Ticket #67519. For keep alive pruposes on interfaces without registration. Required for Lync interoperability. (config change TSIP /options-interval 30)
1877 - SIP: Trap handling 491 response on reliable transport
Ticket #68093. Trap with MAX_BUSY_TICKS
1862 - SIP: Trap when configuring STUN server on a SIP/TCP or SIP/TLS interface
Ticket #67923. STUN is for SIP/UDP only.
1941 - SIP: Trap when handling SUBSCRIBE on federation interfaces
Ticket #68976. SIP client object may be deleted while DNS query is pending. But only under critical timing conditions.
1879 - SIP: Trap when outgoing SIP call is cancelled while DNS resolving is ongoing
Ticket #68099. Trap when outgoing SIP is cancelled while DNS resolving is ongoing
1906 - SIP: Trap when using TLS as transport
Ticket #68410. Only if remote side closes transport connection while requests are pending.
1925 - SIP: Treat domain part of SIP URI case-insensitive
Ticket #68653. According to RFC3261(19.1.4)
1817 - SIP: Uninitialized data in SDP offer/answer
Ticket #67617. Applies to G.726 exclusive calls only.
1848 - SIP: Unwanted media-relay sessions when using forking/broadcast/multi-reg
Ticket #67819. If in incoming SIP was routed to multiple destinations the final session could be media-relay although not configured.
1922 - SIP: Use secondary proxy address on interfaces without registration
Ticket #68635. When secondary proxy address resolved by DNS it should be used to send INVITE to.
1930 - SIP: Web UI for cause code mapping (fix)
Ticket #68772. Did not work as expected
1814 - SoftwarePhone: DTMF to voice mail
Ticket #67563. Now, sending DTMF with myPBX is correctly working. This fixes sending DTMF to voice mail.
1944 - SoftwarePhone: No CT setup with remote connected calls
Ticket #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.
1838 - User Interface: Alignment of tables fixed
Ticket #67734. Some strange alignments were introduced with the last hotfix
1815 - VM: Trap while processing self-forwarded call
Ticket #67570. VM: Trap while processing self-forwarded call
1902 - WEB GUI page cannot be scrolled completely when height of left hand logo is too big
Ticket #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.
1915 - wrong link to PPP Interface State Info Help
Ticket #68463. Link out of V9 namespace
1910 - X509: Fix for reading innovaphone info from flash
Ticket #68435. Parsing the innovaphone info text was incorrect
V9 Hotfix 4 (90600.04)
1989 - auto complete dtmf feature codes with '#' after 2 seconds
Ticket #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 '#'.
1967 - CX0-Wave-Encoding sometimes produced white noise
Ticket #69286. CX0-Wave-Encoding sometimes produced white noise. Also a memory leak was eliminated.
1974 - G726 codec obsolete
Ticket #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.
2000 - H.323: Media Negotiation problem with Lync interop and SRTP
Ticket #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.
1955 - Improved protection against Denial of Service attacks
Ticket #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.
2008 - IP-DECT: Display update
Ticket #69770. Now, the endpoint's display is updated if the registration forwards a display update.
1970 - IP-DECT: Hanging call after call transfer
Ticket #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.
2014 - IP-DECT: Location update (OEM)
Ticket #69863. Location update with message waiting information is fixed if the endpoint roams. This is only used by OEM devices.
1979 - IP-DECT: New radio BMC firmware PCS05Ak
Ticket #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.
1993 - IP-DECT: Reset link
Ticket #69584. The reset link of DECT System Config GUI page is fixed.
1999 - IP200: Manufacturing test fails is hwbuild is reprogrammed
Ticket #69662. Old hwbuild was not cleared, because on INCA devices the hwbuild is located on a higher address was not cleared.
1963 - ip22/24/28/302/305: DSP affected by certain fax tones causing a trap
Ticket #69250. Assert removed for testing
1962 - IP222/IP232: residual echo in handset mode
Ticket #69249. Change Codec config
1998 - IP222: Alpha input using the num block
Ticket #69646. Input mode changes back to numeric in screen "Wahlvorbereitung". Should stay in alpha mode.
1976 - IP222: Call reroute did not work
Ticket #69455. Using redial key on in incoming not-connected call did not work.
1975 - IP222: Input ip addresses does not work
Ticket #69393. Entering an ip address on the phone's menu does not work. All digits are doubled.
2013 - IP6010: SRTP using AES-192 and AES-256 did not work
Ticket #69828. Due to a bug in the encryption driver of the IP6010, only AES-128 worked on this platform.
1960 - IPVA didn't run on vSphere5
Ticket #69232. IPVA didn't run on vSphere5
1996 - IPxx10-sata: trap after config /trace /track activation
Ticket #69642. Instruccion was accessing uninitialized pointer.
1957 - ISDN interop issue with SecuGATE LI 30 from Sirrix
Ticket #69168. The SecuGATE LI30 is sending/receiving ISDN INFO messages in Call Proceeding State (State 3 and state 9), which was not supported
1934 - Kerberos: Protect against ping pong attacks
Ticket #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.
2015 - Logging: Threshold for error event "SRTP authentication failed"
Ticket #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.
1953 - Memory leak with each new telnet session
Ticket #69157. Memory leak with each new telnet session (last packet upon exit not cleaned up)
1980 - new: DHCP manufacturer specific option 'boot-cfg' provides an URL to read config from after any reset except 'creset'
Ticket #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".
1972 - Pass DNS-TTL to SIP
Ticket #69374. In order to do load-balancing SIP needs to ge hold on the TTL of DNS resource records.
2016 - PBX Calls Page/SOAP wrong number
Ticket #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
1964 - PBX Mobility: Support of transfer on mobile side, not using mobility
Ticket #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
1958 - PBX Twinning: Blind transfer to other phone in a twin phone configuration was not possible
Ticket #69170. Only normal call to other phone or consulation call was supported
1985 - PBX Waiting: In case of "Announcemen w/o Connect" together with Alert-Timeout 0, DTMF dialing was not possible
Ticket #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.
1987 - PBX Waiting: No ringback when doing two-stage dialing to a Gateway/Trunk object
Ticket #69531. A local ringback is now switched on, when receiving ALERT from called party
1983 - PBX Waiting: Not possible to send DTMF to Waiting Queue from myPBX Numeric Keyboard
Ticket #69482. The Waiting Queue prohibited the facilities used for this to be sent to calling phone
1977 - PBX: Call Diversion/Forward to '-' did not always show expected result
Ticket #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"
1984 - PBX: Filter for internal or external calls at CFs did not work CFB or CFNR if call already diverted
Ticket #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)
1990 - PBX: HTTP request to initiate call for mobile phone
Ticket #69570. To improve GSM client functionality
1981 - PBX: No CLIR on internal calls did not work for SOAP
Ticket #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.
1965 - PBX: Phone config was not sent to phone, if phone was power cycled shorty after registration
Ticket #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
1982 - PBX: Reject calls without media, if no known facility
Ticket #69477. Fixes compatibility issues between versions. For example presence subscription sessions from v8 phones being forwarded to voicemail
2001 - PBX: WSDL for SOAP API readable from box
Ticket #69688. Simplifies Application development in some environments
2017 - phone ip222: MWI LED not working
Ticket #69924. The MWI LED was not switched on for pending voice mails and/or pickups
2012 - phone: "Keep Calling Party Info on Pickup Key" option from Phone->Preferences did not work with very long numbers/names
Ticket #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
1992 - phone: assume an outbound call to be an external call if connected number info is missing in connect event
Ticket #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.
1995 - phone: VLAN signaling priority could not be configured via phone menu
Ticket #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.
2020 - phone_orchid: checkmark "Phone/User-x/General/Options/No DTMF Detection" has no effect
Ticket #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.
2004 - phone_orchid: Calls received with CLIR appear in call list with an empty entry
Ticket #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
2003 - phone_orchid: wrong volume setting when monitor mode is entered
Ticket #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.
1956 - PPP connection fails after LCP renegotiation with different authentication methods (PAP -> CHAP)
Ticket #69167. the active authentication entity was not stopped when another authentication was started and and signalled layer down after some timeout
2007 - Relay: Forward facilities to local destinations
Ticket #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.
1940 - RTP-DTMF: Start handling of RTP-DTMF on reception of END event
Ticket #68938. Workaround for Bug in MS Lync / Mediation server. Mediation server changes destination port while sending RTP-DTMF redundancy retransmissions.
1988 - SIP: Bug when decoding SRTP keys from SDP
Ticket #69545. Bug in decode base64.
1994 - SIP: Do not check remote certificate name when calling PBX client with TLS
Ticket #69598. Check is reqired only when calling an unregistered client or when giving INVITE to registrar.
2006 - SIP: Get display information from Call-Info header in register refresh response
Ticket #69758. Get display information from Call-Info header in 200/OK
1961 - SIP: Interoperability of INFO(application/dtmf-relay) with Polycom
Ticket #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".
1954 - SIP: Media negotiation problem
Ticket #69159. On media-relay with exclusive codec we can answer an incoming SDP offer right away without passing to app.
1969 - SIP: New config file option /hold-notify-as-inactive
Ticket #69293. If set, holdNotific is interworked into "a=inactive". If not, holdNotific is interworked into "a=sendonly".
1978 - SIP: New config file option /prefer-pai2
Ticket #69459. Interoperability with Telepo: When receiving INVITE, get calling party id from second P-Asserted-Id header.
1949 - SIP: Presence interoperability with ESTOS UC server
Ticket #69050. 'Do Not Disturb' signaling without presence/tuple/contact element.
1986 - SIP: Send BYE with Reason header with "Q.850 Recovery on timer expiry"
Ticket #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"
1950 - SIP: Support for Mediation Server Cluster
Ticket #69051. Load balancing and fail-over acc. to spec.
1973 - SIP: Support of maddr parameter in redirect response for REGISTER
Ticket #69386. Set when redirecting REGISTER. Read when processing redirect response.
2005 - SIP: Take SDP id and version as 64bit integer
Ticket #69738. o-line of SDP offer/answer is defined as 64bit integer Interoperability with MX-ONE
2009 - SIP: Trap when handling NOTIFY(application/qsig)
Ticket #69771. Traps if no progress indicator present in tunneled DISCONNECT message.
1971 - SRTP: Bad key exchange on H.323/SIP interworking
Ticket #69353. MKI was added in SDP due to uninitialized data object.
2018 - Trap during fax transfer
Ticket #69950. Seen on Ip28, can happen on AC_DSP3 ( IP22/24/28/302/305/1060/301/6010 )
1968 - USB Driver merge from v10 to v9
Ticket #69288. First step to support USB headsets. This is not functional yet.
2010 - Voicemail <pbx-query-obj> queries new properties
Ticket #69781. "user_type": returns "1"(endpoint) or "2"(other object like gateway)
Ticket #70697. as <pbx-getcallinfo out-leg2-reason="..."/>
2037 - added support for USB tracing with Wireshark
Ticket #70373. USB traces are now supported by the debug class and can be opened with Wireshark and a new innovaphone.dll.
2078 - CX0 Wave-Encoding Not Working If Fact-Chunk Present In Header
Ticket #71290. CX0 Wave-Encoding Not Working If Fact-Chunk Present In Header
2079 - DHCP Server Identifier could not be cleared via WEB interface
Ticket #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.
2045 - ethernet broadcasts bearing unicast IP packets with an arbitrary destination address were passed to the local IP-stack
Ticket #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.
2047 - Gateway: Forward Display Info received from ISDN Setup to H.323
Ticket #70562. needed for compatibility with SecuGATE LI30
2086 - H.323 re-negotiation: Don't reuse media proposals if a select was already sent
Ticket #71435. This fixes compatibility issues with SIP, especially when SRTP is used.
2085 - H.323 Signaling Timeout event was generated for 'normal' RAS re-transmissions
Ticket #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.
2102 - H.323: No Alarm/Event should be generated by shutting down registration due to reset
Ticket #71545. no event is sent to the application about this kind of unregistration
2043 - H.323: Renegotiation to Fax did not work under some conditions
Ticket #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
2076 - IP-DECT: CSS changes for new OEM device
Ticket #71282. For a new OEM device with a changed style some CSS classes are added.
2023 - IP-DECT: Default frame size to 30ms
Ticket #70140. Now the default frame size is 30ms.
2056 - IP-DECT: Hanging call after failed call transfer
Ticket #70756. The hanging call in dectradio is fixed which occurred if a user does an unattended call transfer and the call transfer fails.
2062 - IP-DECT: License restriction for OEM device
Ticket #70887. A license restriction for an OEM device is added. It is not used in the IP1200.
2026 - IP-DECT: Master radio list sync highlighting
Ticket #70150. In the IP-DECT Master radio list the active sync highlighting is fixed, now.
2105 - IP-DECT: Ring back tone after transfer
Ticket #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.
2029 - IP222/IP232: Handset volume with 90772 andHF4 too low
Ticket #70226. This is a side effect of the echo canceller setting, volume in IP222/232 increased. IP241 uses old settings.
2033 - IP222/IP232: Sporadic Problem with the LCD Display after soft-reset
Ticket #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.
2048 - IP222: Cannot leave menu screen with ESC when IP address ctrl was active
Ticket #70569. ESC was taken as BS (backspace). ESC must be handled as ESC when BS is no longer possible (empty input).
2053 - IP222: Use R-key as BACKSPACE on edit fields
Ticket #70664. Not ESC-key. ESC-key is for leaving screen only.
2110 - IP232: Fix for touch handling
Ticket #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.
2134 - IP232: Redesign of touch keyboard
Ticket #71968. Redesign of touch keyboard
2024 - ip24/ ip6010/phone_orchid: in-band ring back tone generation into voice channel did not work
Ticket #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.
2099 - IP241: Added flag to hide background-image configuration from phone menu
Ticket #70377. Using UP/DOWN/LEFT/RIGHT keys on the "Call Diversion" screen makes ON/OFF changing it state.
2089 - IP241: Show both parties of each call on screen "Pickup List"
Ticket #71455. Currently only calling party infomation is displayed.
2080 - IP241: Show ISDN display information in full length
Ticket #71330. Use scrolling if required
2119 - IP241: Slow screen update when changing fkey type on fkey configuration screen
Ticket #71831. Removing and adding config controls is very slow.
2036 - IP241: Status icon for conditional call diversion
Ticket #70363. Now having 2 different status icons for diversion. One for unconditional and one for conditional forwarding.
2032 - IP241: Use menu-key to enter tftp mode
Ticket #70263. tftp was activated with alt-key, on on IP240 menu-key is used
2133 - ip6010 - calling & dtmf tone timing did not work
Ticket #71966. the tone time was calculated much too short
2065 - IP6010: Wrong timer under high load
Ticket #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.
2059 - IPVA, Keyboard Console, Credentials With Special Chars Couldn't Be Entered
Ticket #70873. The Scancode table wasn't finished
2118 - IPVA, Reset didn't work on VMware Player 4.0
Ticket #71818. IPVA, Reset didn't work on VMware Player 4.0
2137 - IPVA, Trap After Failed FW Upload
Ticket #71978. If a firmware upload failed with "wrong checksum" the box trapped occasionally, leaving the boot disk in an inconsistent state.
2074 - ipva, update to _platform_tracing.xsl
Ticket #71274. ip6 missing
2022 - Linux: VLAN GUI removed
Ticket #70131. It is not necessary to configure a VLAN to Linux. Now the GUI page Linux VLAN is removed.
2100 - Media: Discard RTP packets from wrong source
Ticket #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'.
2051 - myPBX launcher: Hotkey does not work with some applications
Ticket #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.
2041 - myPBX: Closing a chat window sometimes causes a Java Script error
Ticket #70439. client.js, line 2033
The Problem is a collision of closing the window and incoming messages for that window.
2058 - myPBX: Interface for IM provider
Ticket #70867. Interface between the JavaScript client and the myPBX launcher.
2104 - myPBX: Send existing invitation links to users that join the chat session later
Ticket #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.
2072 - myPBX: Show implicit visibility in groups
Ticket #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.
2081 - NAT: Mapping to different internal UDP port did not work
Ticket #71339. Configured destination port got lost after configuration
2174 - PBX Mobility: Potential Trap when initiating a call with myPBX or SOAP while another call is waiting
Ticket #72637. A waiting means a call which was received while another call was active and the active call being disconnected and the waiting call not yet sent, or a call waiting for recall.
2101 - PBX: Blind transfer with consultation to BC-Conference failed
Ticket #71540. The call was disconnected
2094 - PBX: CFU was executed on PRESENCE_PUBLISH/SUBSCRIBE calls
Ticket #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
2050 - PBX: Conference presence info
Ticket #70581. Now the broadcast conference object updates the presence info and is shown as callable in myPBX.
2035 - PBX: Conference trap
Ticket #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.
2087 - PBX: New User property 'Do not Disturb'
Ticket #71439. No calls are sent to the user if set.
Can be set by 'External UC' applications (e.g. Estos Procall)
2049 - PBX: Presence update for PBX objects
Ticket #70577. Now the presence is updated in myPBX for other PBX objects if the access is changed to allowed.
2108 - PBX: Send Name Identification with CLIR calls if "No CLIR on internal Calls"
Ticket #71643. The feature "No CLIR on internal Calls" did not work completetly. The number was sent, but the Name Id was still suppressed
2097 - PBX: Support up to six devices for a user
Ticket #71506. Sometimes 4 devices are to little. A little bit of cleanup on the user interface was done as well with this change
2098 - PBX: Tooltip on "PBX/Config/Log Calls" checkmark wrong
Ticket #71507. It read "If not checked PBX calls are logged", should be "If checked PBX calls are logged"
2092 - PBX: Wrong web page when submitting an object and an error happens
Ticket #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.
2103 - phone-orchid: micro-speaker loop of embedded test did not work
Ticket #71566. Der Test war im Treiber nicht freigeschaltet
2068 - phone: dialog and presence subscriptions sometimes got lost after PBX restart when phone config was stored on PBX
Ticket #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.
2115 - phone: display info provided by SETUP or CONNECT was ignored
Ticket #71727. only the display info provided by an INFO event was handled
2071 - phone: LED mode of Join Group function key can be set both for idle and for active state
Ticket #71247. sometimes the "not in group" state must be signaled as the exception
2114 - phone: Mic Off/On controllable via Soap:UserRc(<call>,14/15)
Ticket #71721. To allow Soap app's control of the mute key
2096 - Phone: Presence-Fkey did not always show presence set by myPBX
Ticket #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.
2082 - phone: send config to PBX only when the config was edited on phone
Ticket #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.
2091 - phone: when a menu item was selected by the adjacent key sometimes the function assigned to this key was also executed
Ticket #71469. The function assigned to the menu item (for example pickup) was executed first and then the key-function (for example dial).
2116 - phone_orchid: continuous dialing tone stops after 10 seconds and is not restarted anymore
Ticket #71753. the continuous dialing tone as used in most countries must be played as long as no digit is entered.
2109 - phone_orchid: Ring Back tone missing when Silence Compression is enabled
Ticket #71670. When Silence Compression was enabled the tone generation was not triggered
2069 - Possibility to clear the call list in mypbx
Ticket #71202. A new button is now available in mypbx to clear the call list.
2107 - Ring Back tone missing after transfer when in-band tone was provided before but not after transfer
Ticket #71612. The initial local ring back tone must be restarted when no data is received after transfer.
2057 - SIP: Allow STUN to be used to map local media ports on SIP/TCP interfaces
Ticket #70809. STUN cannot map signaling TCP port, but UDP media ports.
2030 - SIP: Cannot use SIP phone with myPBX
Ticket #70252. Blind transfer (sending REFER) did not work since v9hotfix3.
2063 - SIP: Diagnostics of error "Registration expired"
Ticket #70938. Missing registration refresh
2060 - SIP: Diagnostics of transport error "SIP Overload"
Ticket #70883. Limitation of buffer allocation
2042 - SIP: DNS resolving _stun._udp.xten.com did not work
Ticket #70449. Because of unusual DNS response. Target attribute of SRV answer records did not contain host domain name, but ip address.
2052 - SIP: Do not try to map local listen port with STUN when SIP/TCP is used
Ticket #70654. Mapping of local listen port is only required fur SIP/UDP
2073 - SIP: Endpoints behind NAT could not register at public PBX
Ticket #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.
2084 - SIP: Interop with Nortel CS1000 SIPLine GW
Ticket #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.
2095 - SIP: Interoperability with MX-ONE
Ticket #71480. A semi-attended transfer fails if MX-ONE sends INVITE(Replaces) instead of 200/OK when connecting a call.
2066 - SIP: Media negotiation problem
Ticket #71009. Internal re-negotiation during early media on incoming SIP call. Provide received offer to app again.
2021 - SIP: Message decoding error
Ticket #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"
2077 - SIP: No media after accepting a waiting call
Ticket #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.
2064 - SIP: No route processing if neither Record-Route header nor Contact header is present
Ticket #70971. Misleading trace message: sip_call::process_routing(0xA8) Unsupported transport protocol: sip:user@domain.com;user=phone
2027 - SIP: No update of name presentation after transfer on QSIG interworking
Ticket #70155. Fix for QSIG/SIP interworking. Display name of callTransferComplete was not passed to SIP.
2067 - SIP: Offer CLEARMODE only if bearer capabilities are "Unrestricted Digital Information"
Ticket #71162. On "Unrestricted Digital Information" only CLEARMODE is offered (no audio codecs). On other bearer capabilities no CLEARMOE is offered (only audio codecs).
2040 - SIP: Prefer P-Asserted-Identity with tel-URI
Ticket #70417. Prefer P-Asserted-Identity with tel-URI over that with sip-URI. In case there are two P-Asserted-Identity headers in INVITE.
2031 - SIP: Presence interoperability with ESTOS UC server
Ticket #70257. Another fix for "Do not disturb".
2025 - SIP: Problems with CLEARMODE
Ticket #70149. CLEARMODE was not offered as prefered codec outgoing INVITE. Destination accepted G711a instead of CLEARMODE:
2028 - SIP: Support for transparent message headers on transfer
Ticket #70163. Interoperability with Telepo.
2111 - SIP: Trap on timer expiration during call release
Ticket #71699. Media negotiation watchdog timer expired after final SIG_REL went to app. But before app deleted the call object.
2135 - SoftwarePhone: New Tray Icon
Ticket #71969. The SoftwarePhone has got a new tray icon differs from the myPbx one, now.
2054 - Trap: When Dectmaster registers user at PBX using SIP protocol
Ticket #70675. After closing regstration Dectmaster starts another call. Call is rejected, but signaling enity is deleted before call object.
2093 - v8 to v9 upgrade problem with gateway registration names containing non-ASCII characters
Ticket #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>
2039 - Webmedia: Recording of G.722 did not work
Ticket #70382. Empty file was created.
2070 - when editing a phone config template the dialing location inherited from a predecessor template was stored in the edited templat
Ticket #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.
V9 Hotfix 6 (90600.06)
2132 - Debug information on assertion
Ticket #71961. More debug information on default event handler.
2131 - H.323 RAS: Registration with authentication to account without authentication failed in a strange way
Ticket #71958. Information was missing that no authentication was supported, so it was continued to retry instead of giving up right away
2247 - H.323: Renegotiation to SRTP after dialtone failed
Ticket #73698. Happens with Escape Dialtones configured in PBX and calls with SRTP enabled
2167 - HTTP: Accept authentication with both Latin1 and UTF-8 coding
Ticket #72512. The server does not know what encoding is used in Basic- or Digest-Authentication. Therefore both encodings shall be tried.
2241 - IP-DECT: DECT endpoint update for multicast identities
Ticket #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.
2191 - IP222,IP232: Long key press on numeric block while being offhook does not open directory search
Ticket #72831. Long key press during offhook is expected to open directory search.
2223 - IP222/IP232: hand receiver volume too high
Ticket #73306. Die IP222/IP232 wird gelegentlich als zu laut empfunden. Die Prozentanzeige mit den krummen Balken sieht "komisch" aus. 0% (Slider ganz links) wäre gut. Mehr Werte wären gut.
2218 - IP222/IP232: voice quality in handset mode
Ticket #73263. * Neuer DSP code von Audiocodes ( patch Version 660 02 mit AMR_WB ). * Kein StrongEC mehr nötig * Codec PCM Bus läuft mit 16KHz * Wideband Codecs zur Zeit disabled * Konferenz erfordert noch Anpassung im Treiber
2163 - IP222: Take center key of 4-way-navigation control an OK key
Ticket #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.
2224 - IP232: Cursor positioning when touching ip address control
Ticket #73407. Make sure input cursor is at right side of touched octet.
2175 - IP232: Fix for touch keyboard handling
Ticket #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.
2141 - IP232: Input of capital Latin Extended-A
Ticket #72226. Input of capital Latin Extended-A
2251 - IP232: Memory leak in display rendering
Ticket #73864. Memory leak in display rendering
2192 - IP232: New touch control in menu header for screen exit
Ticket #72833. Re-design
2160 - IP232: Parameters for touch sensor fixed for latest sensor
Ticket #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)
2106 - IP232: Presence fkey shows current activity, but not current note
Ticket #71611. Show current note in second line of fkey label.
2144 - IP232: Touch keyboard key highlighting
Ticket #72268. Added highlighting for key like SHIFT, BACKSPACE, etc.
2179 - ip241 - monitor mode (handset + speaker) did not work in V9hotfix5
Ticket #72702. monitor mode (handset + speaker) was displayed but the speaker was mute
2245 - IP241,IP222,IP232: Extra digits dialed when using RIGHT key on screnn 'indrect dialling'
Ticket #73694. '123123' instead of '123' was dialed.
2161 - IP241,IP222,IP232: Make password configuration more convenient
Ticket #72456. When entering a passwort on the phone's ui content is displayed as "****". Show last enered character in plain text.
2243 - IP241,IP222,IP232: Missing "Recall possible" display text on call completion
Ticket #73628. Missing display text on 'recall possible' notification.
2209 - IP241,IP222,IP232: Show number of missed calls in status bar
Ticket #73140. New extended symbol is more eye-catching.
2195 - IP241,IP222,IP232: Support for hebrew and arabic presence notes
Ticket #72905. Hebrew and arabic text passages are rendered frmo right to left.
2194 - IP241,IP222,IP232: Support for unicode 0x308 (diaeresis/umlaut)
Ticket #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.
2240 - IP241,IP222,IP232: Symbol "new messages" and symbol "headset" do overlap in status bar
Ticket #73608. Moved "headset" symbol to right side of status bar.
2231 - IP241,IP222,IP232: Trap when scrolling down long call list
Ticket #73494. Out of memory trap when scrolling down long call list.
2199 - IP241,IP222,IP232: Update for CFU indication on idle screen
Ticket #73013. Not displaying "cfu:241" on header of idle screen but "Diverted to 241" or "Umgeleitet zu 241" or "Renvoyé sur 241" etc.
2187 - IP241,IP222,IP232: Wrong call state displayed
Ticket #72748. "Destination reached" displayed instead of "subscriber busy" if consultation call was rejected.
2149 - IP241: Disable Link LED not working
Ticket #72338. Timing problem with access to the paged MDIO registers
2168 - ip28 codec hang when TEL port configured to '600Ohms'
Ticket #72521. this option experienced problems in the past and should therefore not be selected. If selected the firmware will switch to default CTR21.
2156 - IP2x2: Fixed fkey handling of builtin test
Ticket #72442. Fkey test did not work for touchkeys, key symbols were misplaced on screen
2219 - IP2x2: Keyboard handling on call reroute
Ticket #73268. Redial key launches "Transfer" screen. After entering the destination number, Redial key should cause execution of call reroute.
2185 - IP4001/IP6000 -UART driver did not work
Ticket #72743. the hardware requires the registers to be written 32-bit wise which was not respected by a fix one year before.
2242 - Ip6010 DSP Allow calls to start with T.38
Ticket #73626. Needed for some T38 fax applications
2162 - Ip6010 DSP Allow coder change from T38 to voice and back to T38
Ticket #72457. ..
2153 - Ip6010 DSP Disconnect timeout after fax session
Ticket #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".
2238 - IPVA, crash-dump without backtrace, if tracing was switched off internally
Ticket #73574. IPVA, crash-dump without backtrace, if tracing was switched off internally
Ticket #71954. Specifically to reduce boot-time. Utilizing 64KB I/O accesses where possible.
2246 - ISDN Trunk: Transfer to ISDN Trunk with TONE interface failed
Ticket #73695. There was not media after the transfer
2239 - ISDN: Do not provide tones and no signaling of inband info for unrestricted digital information calls
Ticket #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).
2204 - LDAP replication from v7 could stop
Ticket #73052. LDAP replication from v7 could stop
2235 - Linux: Empty file check
Ticket #73554. Now, an empty file is not accepted as a successful file upload.
2211 - Media Negotiation: v9 XPARENT not compatible to v8 or earlier XPARENT
Ticket #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.
2229 - myPBX: Allow overriding implicit allows
Ticket #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.
2221 - myPBX: Allow spaces in last LDAP attribute
Ticket #73276. Currently "j doe innovaphone" finds all internal users starting with j. It should only find internal users that start with "j doe innovaphone".
2124 - myPBX: Cause codes for instant messaging
Ticket #71878. Display causes why a chat session has been terminated.
For example: Unassigned number, rejected, no response...
2173 - myPBX: Configure visibility of online state and presence separately
Ticket #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.
2147 - myPBX: Display invitation links in the chat window of the organizer
Ticket #72282. The organizer of an application sharing session or a audio conference wants to see all corresponding links in the chat window.
2136 - myPBX: Display reason why a browser is not supported
Ticket #71972. Currently it only says "Browser version not supported".
It should also tell what feature isn't supported: - XmlHttpRequest - WebStorage - PostMessage
2158 - myPBX: Display shortened links in chat messages
Ticket #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.
2202 - myPBX: Do not show bubble for outgoing chat calls and messages
Ticket #73039. Notifications are only needed for incoming calls and messages.
2146 - myPBX: Forward invitation links to incoming chat calls that are added to a chat conference
Ticket #72280. The same way like the invitation links (for application sharing and audio conferences) are forwarded to additional outgoing chat calls.
2205 - myPBX: Highlight incoming alerting calls
Ticket #73093. Highlight the alerting icon using an orange border.
2157 - myPBX: Improve presentation of chat conferences
Ticket #72444. Chat conferences should not be displayed as multiple calls but as a single item in the main window.
2125 - myPBX: Make call state icons clickable
Ticket #71886. Clicking a call state icon should have the same effect as clicking the text.
2126 - myPBX: Make more obvious that closing the chat window terminates the chat
Ticket #71893. Display warning message when the user tries to close the chat window. Show disconnect icon instead of close icon in the window.
2165 - myPBX: Name and Number Display not correct on IM sessions across PBXs
Ticket #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
2170 - myPBX: New icons for unknown phone and im status
Ticket #72536. The difference between open, closed and unknown was not understood well.
2189 - myPBX: New translations
Ticket #72768. For the other v9hotfix6 items regarding myPBX.
2127 - myPBX: Notification on disconnected chat sessions
Ticket #71918. Display a text that the chat has been terminated.
2142 - myPBX: Queue chat messages for im calls that are not yet alerting
Ticket #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.
2200 - myPBX: Simplify adding multiple users to a chat session
Ticket #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.
2203 - myPBX: Skript errors in chat window
Ticket #73040. Closing the chat window caused different script errors in Internet Explorers and Firefox.
2248 - Option added to wait for the DSP to disconnect fax, needed only for some fax devices.
Ticket #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.
2176 - PBX Config Templates: When editing config Templates wrong inherited values are displayed
Ticket #72655. A config Template inherits only from the 4 other Templates configured in it directly, whereas any other object inherits from the Config Template configured with it and the four other Templates this is refering to.
2208 - PBX External UC: Support of multisite setup
Ticket #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.
2178 - PBX Mobility: Better support for Opticaller
Ticket #72701. HTTP request to initiate call for mobile phone
2188 - PBX Waiting: Ringback missing when using DTMF to dial from one WQ to another which is alerting
Ticket #72766. This was a collateral damage for a fix for Waiting Queue announcements from a Boolean Object
2220 - PBX Waiting: When using a waiting queue for outgoing dialing, the announcement was cut off at the beginning
Ticket #73274. This was because the announcement was started already when the call was sent out and not when the call was connected
2184 - PBX: Called Name displayed when calling an object with forking was wrong
Ticket #72735. The name of the forking destination was displayed instead of the name of the called object
2210 - PBX: Called party number wrong at original called phone with forking
Ticket #73145. If the forking destination and the original called phone are assigned to different nodes
2169 - PBX: Cause for release/reject of a IM session not signaled
Ticket #72533. Better feedback to the User
2215 - PBX: Conference GUI update with IE
Ticket #73256. The PBX broadcast conference object window is not closed if IE is used. This fixed now.
2196 - PBX: Conference no media
Ticket #72915. There is sometimes no media for a conference member. This is fixed, now.
2222 - PBX: Forward original received ISDN display element to picking up or forwarded call
Ticket #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.
2145 - PBX: H.323 Names in some places (e.g. Waiting Queue Maps) containing non-ASCII charcters could break the user interface
Ticket #72277. This happened esspecially for Names as destinations for Waiting Queue maps, but it could happen at other places as well.
2237 - PBX: Navigating in locations and groups if non-ascii charaters were used, failed with internet explorer
Ticket #73567. Internet explorer buggy concerning url encoding of href attributes
2171 - PBX: New presence access flag 'online'
Ticket #72538. With this flag a user allows access to his online status
2186 - PBX: No Audio if call thru Waiting Queue DTMF destination, was transfered to BC-Conf
Ticket #72746. Problem caused by call state management error in PBX for calls connected without alert if alert was received later
2193 - PBX: Profiles/Access got lost when writing an User Object with SOAP
Ticket #72849. The respective tags were not allowed
2228 - PBX: Reporting licenses counting could be wrong if config templates were used
Ticket #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
2213 - Phones: Update of phone text data
Ticket #73211. Some translations missing (Polski,Eesti). Support for language typical letters (Polski). Some abbreviation removed (for color phones).
2201 - RTP-DTMF: Digit may get lost during media re-negotiation
Ticket #73037. Receiver starts handling on END event. Sender may stop sending before END event was sent.
2143 - Scheduling improved to avoid processes not being scheduled during long flashman operations
Ticket #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.
2183 - SIP: Add payload type for RTP-DTMF in case of media-relay
Ticket #72732. Add payload type for RTP-DTMF (telephone-event) to SDP offer in case of media-relay. Should help on Mobility scenarios.
2148 - SIP: Cleanup failed (resources leaking)
Ticket #72284. Call and channel objects were not freed sometimes when INVITE was followed by CANCEL very fast.
2227 - SIP: Configuration of SIP response code mappings did not work on IPxx10 and IPVA
2206 - SIP: Interworking with KIRK Wireless Server 300
Ticket #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.
2244 - SIP: Media negotiation problem on outbound call from BC Conference object
Ticket #73652. INVITE was sent without SDP offer. test/9.00/pbx/conf/SIP/002-conf-with-bcast.xml
2197 - SIP: Memory leak during transfer
Ticket #73003. Occured on internal testing only (002-conf-with-bcast.xml)
2230 - SIP: Overriding cause code mappings did not work
Ticket #73477. Mapping of SIP response codes into Q.931 cause codes could not be overwritten.
2216 - SIP: support for "Content-Type: multipart/alternative"
Ticket #73260. Lync interoperability
2140 - SIP: Trap on IP-DECT when re-configuring PBX link
Ticket #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
2250 - SIP: Using wrong remote port when registering
Ticket #73784. Only affects IP-DECT when handset is switched OFF and ON and if the SIP runs on non-standard port.
2139 - SIP: Workaround for delays in media negotiation
Ticket #72189. Workaround for delays in media negotiation caused by delayed Admission on H.323 calls.
2155 - SoftwarePhone: Crash dump
Ticket #72433. Now, the SoftwarePhone writes a crash dump file.
Ticket #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.
2154 - SoftwarePhone: Password encryption tool
Ticket #72431. A password encryption tool was added.
2225 - SoftwarePhone: Support for Jabra SPEAK 410 USB with product id 0x0410
Ticket #73424. Now, the SoftwarePhone supports Jabra SPEAK 410 USB with product id 0x0410.
2166 - SoftwarePhone: Trap and media channel after conference
Ticket #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.
2190 - Support for more automated Fax Tests
Ticket #72826. Switching from Audio to Fax to Audio to Fax. Special case which could happen if switch to Fax happened before connect.
2138 - TCP: Roundtrip measurement wrong in case of packet loss
Ticket #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.
2150 - Trap: When accessing web interface
Ticket #72362. Seen once. Not reproducable yet.
2182 - Trap: When Dectmaster registers user at PBX using SIP protocol
Ticket #72729. When Dectmaster registers user at PBX using SIP protocol
2232 - Trap: When using Diagnostics/Ping
Ticket #73536. When leaving Diagnostics/Ping or Diagnostics/Traceroute web screen a trap may occur.
2180 - update - scfg command could hang when the HTTP session was broken or prematurely closed by the server
Ticket #72708. in consequence update script processing was stopped until reboot
2181 - Web-UI: Wider input fields for domain names
Ticket #72723. When configuring - Remote Domain - Local Domain - Proxy - STUN Server on a Gateway interface there's need for more space.
V9 Hotfix 7 (90600.08)
2293 - debug method raw_ethernet
Ticket #74897. A new debug method to trace raw ethernet data.
2258 - Dect User Gui: AD-replicated objects weren't listed under certain conditions
Ticket #74051. Deleted object thate were re-replicated werent't visible in the Dect user GUI. A mandatory Ldap attribute was missing under such circumstances.
2339 - different port naming convention for new hardware
Ticket #75575. BRI1..5 instead of TEL1..4+PPP
2316 - Fix for MIPS counter
Ticket #75310. MIPS counter was incorrect
2274 - Flash Directory: Substring match could miss results
Ticket #74320. Searching by substring worked case sensitive in some cases, instead of functioning case insensitive.
2369 - Flashman: Show meaningful result after OEM certificate upload in production
Ticket #75958. "Certificate" instead of "-unknown-"
2322 - Gateway Interface Maps: Should be applied to leg1 info also
Ticket #75437. The same rules used for a Calling Party Number are applied to leg1 info
2406 - Gateway: Better handling of call-reroute requests
Ticket #76515. Apply interface maps to numbers in reroute request
2396 - Gateway: Conference interface, no voice
Ticket #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.
2276 - Gateway: Deleting of routes could result in duplication of routes
Ticket #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.
2327 - Gateway: Event "No Media Data Received" was created for each SWITCH-PCM call
Ticket #75483. There should be no event in this case, this is normal.
2393 - Gatway: Transfer handling may cause re-routing after regular hang-up
Ticket #76357. Re-routing is expected to take place on non-connected calls only.
2352 - H.323: A forwarded HopCount>32 could result in a very small HopCount
Ticket #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
2370 - H.323: Call to SIP Trunk with MediaRelay and exclusive coder renegotiation from TONE to early media did not work
Ticket #75959. No ringback and no audio after connect
2386 - H.323: Merges from v10 for Media Negotiation fixes and special OEM protocol features
Ticket #76301. - Problem with transparent forwarding of SDP - special SRTP key exchange - Video fixes - Unit Testing support
2304 - H.323: Renegotiating to Fax was rejected, if PCM switch and local networks were used
Ticket #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.
2255 - H.323: Renegotiation after PBX Node dialtone failed sometimes
Ticket #74020. This depended on timing. This happened if the call was sent after the dialtone to a master over a slow link.
2315 - H.323: Renegotiation to T.38 did not work for slowstart call thru media-relay to efc endpoint
Ticket #75305. A common scenario when this happend is an XCAPI (slowstart) calling thru a PBX with media-relay configured to an ISDN interface
2333 - H.323: Support for switch back to local media on endpoint
Ticket #75522. needed to resume normal call after a conference using external conference unit
2294 - Interop: Don't add an error log "SRTCP authentication failed" on non-RTCP packets
Ticket #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.
2388 - IP-DECT: Memory leak with rejected radio registration
Ticket #76308. Memory leak occurred with a rejected radio registration is fixed, now.
2392 - ip22/24/28/302/305/6010/3010/1060: Fax failed if tones are sent during the fax call
Ticket #76342. In some cases the Fax messages were missing due to tones send during the fxx call
2378 - IP222,IP232: Alpha mode did not work on multi-line edit controls
Ticket #76165. Could no enter alpha chars using the num block
2307 - IP222,IP232: Changes not saved in some cases
Ticket #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.
2291 - IP222,IP232: changes not saved when leaving menu screen with LEFT key
Ticket #74792. Even after confirming to save
2288 - IP222,IP232: Function key "Hotdesk" did not work
Ticket #74723. Function key "Hotdesk" did not work
2261 - IP222,IP232: Language configuration not saved
Ticket #74087. Language can be selected on phone menu, but change is neither saved nor applied.
2260 - IP222: Handset Microphone level too low
Ticket #74073. Handset micro increased by 6db Handsfree micro unchanged
2289 - IP232: Fix for long touch on fkey
Ticket #74743. Opens fkey config, but mist not toggle control on touch-off.
2340 - IP232: Hiding touch keyboard by touching a control
Ticket #75576. Touching a text control activates the touch keyboard. Touching the text control again now deactivates the touch keyboard.
2385 - IP232: Open call details when touching entry in call list
Ticket #76292. Instead of initiating call to remote party in call list entry.
2256 - IP232: Open phone menu by touch gesture
Ticket #74046. Touching header bar and moving vertically down opens the phone menu. No need to use center key of 4-way control.
2283 - IP232: Pickup fkey displays no information in active state
Ticket #74392. Pickup fkey displays no information in active state
2266 - IP232: Pixel noise with some color values
Ticket #74221. - changed to lower pixel clock
2336 - IP232: Touch gesture to open directory-search/indirect-dialing screen
Ticket #75549. Vertical upwards on idle screen opens directory-search/indirect-dialing screen. Vertical downwards on idle screen opens main-menu screen.
2269 - IP241,IP222,IP232: Added drop shadow to lower end of menu screens
Ticket #74278. To make end of list of menu entries more discernible.
2329 - IP241,IP222,IP232: Call duration display wraps after 100 minutes
Ticket #75508. Now after 60 minutes the display changes from mode [mm:ss] to [hh:mm].
2399 - IP241,IP222,IP232: Call held indication
Ticket #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.
2345 - IP241,IP222,IP232: CFU information in header bar
Ticket #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.
2278 - IP241,IP222,IP232: Could not activate first builtin background image from phone menu
Ticket #74367. Could not activate first builtin background image from phone menu if external background image is currently active.
2324 - IP241,IP222,IP232: Indirect dialing of names not possible
Ticket #75451. Could not dial a name from 'indirect dialing' screen by moving the focus to input field and going offhook. (Only fkey worked)
2254 - IP241,IP222,IP232: Limitation of background image size
Ticket #73957. Limit is 512 kBytes now. To keep phone from out of memory trap.
2300 - IP241,IP222,IP232: Mute status indication does not work
Ticket #74964. "MICROPHONE OFF" is displayed on status bar now.
2348 - IP241,IP222,IP232: Replace triangle by arrow to display diversion/transfer information
Ticket #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.
2383 - IP241,IP222,IP232: Show "New message from ..." on active phone screen
Ticket #76283. Show "New message from ..." on phone screen for 3 seconds when active with calls (non-idle). Like on b/w phones.
2279 - IP241,IP222,IP232: Show multiple diversions on incoming calls
Ticket #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)
2257 - IP241,IP222,IP232: Some diagnostics to check phone's ui memory consumption
Ticket #74049. Where have all the memory gone.
2384 - IP241,IP222,IP232: Toggle of display-name/name-alias/number was buggy
Ticket #76290. String termination was missing. Garbage data was displayed.
2355 - IP241,IP222,IP232: Two waiting calls are now displayed (instead of one)
Ticket #75789. Waiting calls are displayed below the two main calls. Only two lines of information and grey background.
2296 - IP241,IP222,IP232: Volume control not displayed when already at maximum
Ticket #74927. Volume control is not displayed when trying to increase while being at maximum.
2277 - IP241/222/232: Monitormode (Lauthören): Level too low
Ticket #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
2312 - IP28 Watchdog leads to endless interrupt and trap
Ticket #75186. .
2298 - IP6000 crypto driver: Trap when buffers are depleted
Ticket #74935. Avoid the trap and log an Event when the buffers are depleted.
2212 - Ip6010 DSP Allow coder change from T38 to voice and back to T38 with local DSP
Ticket #73193. ..
2313 - IP800/6000: Problem mit FAX und CNG detect: Option to disable CNG detect added
2366 - IPv6: Memory leak if packets received, which are not handled locally
Ticket #75920. E.g. an IPv6 multicast UDP packet, for some other application was not deleted.
2292 - IPxx10: Flashdir Segments Default to 129(was 51)
Ticket #74826. IPxx10: Flashdir Segments Default to 129(was 51)
2214 - ISDN: Enable fax detection only after connect
Ticket #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.
2271 - ISDN: Send HLC with mobility calls
Ticket #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.
2402 - LDAP-Expert, Edit Object Dialog: Could show more than one object
Ticket #76464. LDAP-Expert, Edit Object Dialog: Could show more than one object
2280 - myPBX launcher: Unhandled exception when accessing browser object
Ticket #74380. The myPBX launcher could crash when accessing the browser document because of an unhandled exception.
2342 - myPBX launcher: Use default configuration from local machine registry hive
Ticket #75634. If no user configuration is given, the following values are copied from the local machine hive.
2412 - myPBX launcher: Window position was not always remembered
Ticket #76626. When the user just moved the window without resizing it, the window position was not remembered for the next session.
2265 - myPBX: Allow asterisk as a wildcard in LDAP search strings
Ticket #74200. As it is done on our phones.
2334 - myPBX: Script error after waking the PC up from hibernation
Ticket #75532. Sometimes window.open throws an unhandled exception in this case.
2281 - myPBX: Support static URLs for application sharing
Ticket #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.
2285 - NAT: Permanent UDP forwarding did not work
Ticket #74549. Permanent UDP forwarding was unusable after first forwarded datagram.
2270 - No media for calls from SIP provider forwarded back to SIP provider in case of NAT and MediaRelay
Ticket #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.
2321 - PBX Exec Object: Leg2 information send to secretary not adjusted correctly for nodes
Ticket #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
2394 - PBX Number Map: Not possible to use overlap dialing to Number map with incomplete destination
Ticket #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.
2308 - PBX Waiting: Diverting leg1 info not correct when diverting to a Waiting Queue
Ticket #75091. If Waiting queue not defined in root node and escapes are used
2286 - PBX Waiting: If used as outgoing dialing device, coder list sent with the call was not correct
Ticket #74593. Should be based on the coders paramter in the configured URL
2358 - PBX Waiting: Leg2 information sent with calls to operators not adjusted correctly for nodes
Ticket #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
2407 - PBX-SOAP: Better error handling when using a Waiting Queue for outgoing calls
Ticket #76544. - call was hanging if an invalid number or user without registration was called - trap if outgoing call was done on operator connect
2341 - PBX-SOAP: UserHold without MOH to local User
Ticket #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
2391 - PBX-SOAP: When monitoring Waiting queue, the peer number was not indicated
Ticket #76315. The number was considered presentation restricted, because of some bit manipulation error
2337 - PBX-SOAP: Wsdl Versioning mechanism fixed
Ticket #75552. Better mechanism, to avoid constantly adding arguments to the Version function with new Versions.
2325 - PBX/IP-DECT: User password length
Ticket #75453. The maximum user password length in the PBX and IP-DECT is increased to 23.
2309 - PBX: Adjusting received leg1 info wrong
Ticket #75097. In context of nodes with escapes
2318 - PBX: Allow 'Max Calls' configuration of 0
Ticket #75362. So that no call at all is sent to the object but a CFB (if configured) is executed
2275 - PBX: CF at Gateway Type objects - additional dialed digits should be added to the destination
Ticket #74348. This way a CFNR at a trunk object can be used to reroute the call to another trunk.
2326 - PBX: CFNR Loop check detected loops that weren't
Ticket #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
2328 - PBX: Conference id prefix/suffix configuration
Ticket #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.
2267 - PBX: Conference to VM
Ticket #74222. Now a call to the VM as a conference member works again.
2282 - PBX: Tracing flag turns on tracing in all dyn PBX's as well
Ticket #74390. Helpfull to debug dyn PBX setups
2403 - phone - Headset Function Key can be configured now as enable/disable or as call control key
Ticket #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)
2401 - phone ip222, ip232 : USB headset support (beta)
Ticket #76463. general support for USB headsets added, some headsets are tested and working, others will follow soon
2368 - phone_orchid: after leaving a conference the active call was mute sometimes
Ticket #75955. The conference was not stopped internally in all cases.
2295 - phone_orchid: ethernet statics counters sometimes displayed as negative numbers
Ticket #74913. the unsigned counters were printed as signed integers
2287 - Phone_orchid: Residual echo in handset ( sporadic)
Ticket #74671. Echo canceller NLP sensitivity Mode changed to 2
2398 - phone_orchid: spurious trap in long conference calls
Ticket #76445. in long confernce calls the phone may trap because the DSP delivers an unexpected zero size packet.
2264 - Phones: Fine grained function hiding did not disable Fkey configuration
Ticket #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.
2335 - SIP: Check proxy availability did not work in any scenario
Ticket #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.
2381 - SIP: Contact not allowed in BYE/CANCEL/PRACK request
Ticket #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
2359 - SIP: Decoding of RFC-4412 definitions
Ticket #75848. The 'Resource-Priority' Header Field The 'Accept-Resource-Priority' Header Field The 'resource-priority' Option Tag 417 Unknown Resource-Priority response
2314 - SIP: Decrement Max-Forwards and interwork to H.323
Ticket #75288. Instead of sending always "Max-Forwards: 70"
2263 - SIP: Don't take NOTIFY(application/simple-message-summary) with "Do-Not-Disturb: no" as FTY_MWI_DEACTIVATE
Ticket #74162. Interop with Nortel CS1k PBX
2373 - SIP: Fix for Supported header
Ticket #76016. Re-implemented to easily add more tags
2367 - SIP: Fix for video negotiation
Ticket #75954. Fix for video negotiation
2362 - SIP: Handling of 422 Session Interval Too Small
Ticket #75873. Get Min-SE and re-try INVITE
2252 - SIP: Handling of collision of transfer and release
Ticket #73936. If one end releases a call while the other initiates an attended transfer, a "ghost call" may remain. Resource leak.
2356 - SIP: Handling of some more proprietary Alert-Info and Call-Info in INVITE
Ticket #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
2305 - SIP: Interworking of divertingLegInformation1 improved
Ticket #75025. Honour subscriptionOption
2311 - SIP: No interworking of "Q.931 CALL PROCEEDING" into "183 Session Progress"
Ticket #75140. Causes trouble on other vendor PBX's. (Aastra 5000, Advoco/Arcstel, Nortel SESM)
2302 - SIP: Rare problem with blind transfer
Ticket #74991. callIdentity must be zero when giving ctInitiate to PBX on blind transfer.
2319 - SIP: Send "305 Use Proxy" if INVITE is received from unexpected source
Ticket #75380. Applies to registered interfaces only (e.g. phones).
2363 - SIP: Send b=TIAS attribute in media description
Ticket #75881. Send b=TIAS attribute in media description
2380 - SIP: SIP message may exceed available buffer
Ticket #76203. Trace message like this can occur: ERROR: SIP message buffer (1025) exceeded! (9,14,801d781c) SIP: CSeq missing SIP: Message encoding failed!
2262 - SIP: Support for "Content-ID" in "multipart/alternative" bodies
Ticket #74161. Support for "Content-ID" in "multipart/alternative" bodies
2375 - SIP: Support for active call pickup acc. to RFC-3891
Ticket #76035. Sending INVITE with Replaces header to do pickup at SIP PBX's
2395 - SIP: Support for media recording
Ticket #76396. Media recording to HTTP URL
2361 - SIP: support for proprietary message header
Ticket #75864. Support for proprietary message header (CAL)
2379 - SIP: Trap when parsing very large History-Info header
Ticket #76198. Trap when parsing very large History-Info header
2409 - SoftwarePhone: Auto start configuration
Ticket #76595. The duplicate back slash occurred with the install directory is fixed now. Used by the run mode configuration 'At login'.
2323 - SoftwarePhone: Timer tick resolution
Ticket #75447. The timer tick resolution is increased. This fixes the call RTCP calculation transmitted to the PBX.
2408 - Spurious leaks in leak checking
Ticket #76589. A problem which mostly shows up in automated release testing and then requires manual investigation.
2301 - Support for new hardware
Ticket #74990. .
2330 - TLS: Duplicate alert message on malformed ClientHelloV2
Ticket #75509. Only one alert should be sent per session.
2303 - TLS: Flow control for incoming data
Ticket #75004. The TLS socket has to wait for the application to process incoming data before sending the next RECV.
2331 - TLS: Improved negotiation of protocol version
Ticket #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.
2320 - TLS: Possible trap when restoring sessions
Ticket #75394. Only in v9, depending on the certificate.
2332 - TLS: Skip empty records
Ticket #75511. TLS record layer should ignore records with zero length without doing anything.
2310 - VM: <pbx-upd-obj type="cfu"..> without effect when invoked multiple times
Ticket #75121. Statement <pbx-upd-obj type="cfu"..> failed to work properly after being used for diversion manipulation multiple times within a single script session.
2290 - Web-UI: Configuration of "Sync Server" does not return with HTML page
Ticket #74778. Configuration of "Sync Server" does not return with HTML page
2476 - ip6010/3010/1060: Ethernet receive packet sometimes delayed
Ticket #77781. Sometimes the rx descriptor are processed with the next tx event. Now the rx queue is processed completely in on interrupt.
2474 - ip6010/3010/1060: Ethernet transmit packet length is sometimes wrong
Ticket #77774. Sometimes old content of the tx dma descriptor was used by the ethernet MAC. Now the memory write buffers are drained before enabling the tx dma.
Ticket #77027. 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
2466 - late RTP data sent from DSP driver to media endpoint may cause a trap
Ticket #77642. this was observed in a phone_orchid test setup with media recording activated
2465 - Media: Send CN frame with minimal noise level
Ticket #77625. For NAT reasons innovaphone endpoints send up to 4 RTP packets to remote end in order to open mapping on NAT router. These CN packets carry one-byte payload with value of 0. Value of 127 turned out to be more clever.
2435 - Memory Leaking In 802.1x Module
Ticket #76960. Memory Leaking In 802.1x Module
2419 - myPBX launcher: Always save window position when closing the application
Ticket #76763. 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.
2416 - myPBX: Localised date and time format in call list
Ticket #76737. myPBX should show a localised date format instead of YYYY-MM-DD HH:MM:SS.
2377 - myPBX: Log event if application sharing failed
Ticket #76116. Log an event if the creation of a meeting failed.
2436 - PBX-SOAP: Call initiated by SOAP for softwarephone or IP-DECT was sent as transfered call
Ticket #76962. The result was that call diversions or busy on ... calls settings were ignored
2432 - PBX: Update of remote number after call-transfer did not always work across PBXs
Ticket #76942. 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.
The same wrong number was also indicated on SOAP/TAPI as well as on myPBX.
Ticket #77540. 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.
2446 - SIP: Bug when interworking UUI of 1-byte length
Ticket #77225. Results into bad formatted INVITE since User-to-User header contains uninitialized data.
2452 - SIP: DNS refresh is not executed according to DNS-TTL
Ticket #77309. Proxy IP address needs to be refreshed by DNS request according to DNS-TTL.
2414 - SIP: Do not offer CLEARMODE on exclusive coder config
Ticket #76654. Do not offer CLEARMODE if exclusive coder configured
2389 - SIP: Fix for REFER handling
Ticket #76309. Handling of blind transfer without userpart in Refer-To-URI.
2420 - SIP: Handling of 300 responses
Ticket #76771. Select Contact-URI according to q-value. Replace remote target URI and change remote transport address.
2458 - SIP: Media negotiation during blind transfer caused ERROR log 0x70007
Ticket #77385. Media negotiation during blind transfer caused ERROR log 0x70007 on the outbound call to transfer destination.
2449 - SIP: SDP answer must have the same number of media descriptions as received offer
Ticket #77277. SDP answer must have the same media descriptions as received offer
2450 - SoftwarePhone: Exception during start up
Ticket #77288. Fix for fix #75886.
2437 - SoftwarePhone: Support for Plantronics Savi 740
Ticket #76975. HID Keypad Mapping support for the Plantronics Savi 740 with the product id 0xac01.
2469 - Trap during HTTP request with missing user/pwd in basic authentication header
Ticket #77695. .
V9 Hotfix 9 (90600.11, withdrawn)
2299 - add programmable tftp retry limit to httpclient
Ticket #74936. currently there is a fixed retry limit of 4. some OEMs want to set this dependent on expected file size.
2509 - DHCP server leases with hostnames containing non-ascii latin1 characters (for example Umlauts) could not be displayed
Ticket #78382. In pre-V9 firmware hostnames were stored latin1-encoded. Names contaning non-ascii latin1 charaters must be converted to UTF8 before display.
2531 - Gateway: Handling of blind transfer gets stuck in routing
Ticket #78709. Routing of transfer call stops with: reason='incomplete' But during transfer no dialing digits will follow.
2499 - Gateway: Trap when interworking Call Completion
Ticket #78228. Trap when interworking Call Completion.
2487 - Gateway: Wrong Media info sent for calls forwarded from a VOIP interface to another VOIP interface
Ticket #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.
2390 - H.323: Media Negotiation problem with conferences on IP-DECT
Ticket #76314. A channel was not switched to the conference after a hold/retrieve cycle
2468 - H.323: Media Negotiation problem with transfer in Gateway (not PBX)
Ticket #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.
2505 - H.323: Potential Trap in case of special malformed RAS registration
Ticket #78368. A NULL pointer access could happen
2516 - IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax failure after transfer
Ticket #78487. channel was reconfigured on remote switch to T38, without a new activate fax relay command
2494 - IP1060 IP3010 IP6000 IP6010: T1 mode: wrong pulseshape for short lines.
Ticket #78157. Slew rate and level for T1 pulse for short lines was too high
2527 - IP222 IP232 IP241: Sporadic noisy sidetone
Ticket #78649. Analog and digital sidetonegain were enabled,now only the analog sidetonegain is used.
2518 - IP241,IP222,IP232: No RTP send on outgoing call in some cases
Ticket #78550. No RTP send on outgoing call in some cases
2485 - IP241,IP222,IP232: Show special symbol for call completion entries in call list
Ticket #77966. Show special symbol for call completion entries in call list to separate from usual missed calls.
2486 - IP241,IP222,IP232: Status message "Recording" not displayed
Ticket #77992. Status message "Recording" needs to be displayed during recording.
2498 - IP241,IP222,IP232: Trap in font rendering
Ticket #78210. Some characters at beginning of line may cause trap.
2526 - IP241,IP222,IP232: Wrong call direction indication on PARTNER fkey
Ticket #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)
2488 - IP241,IP222,IP232: Wrong display name in call list
Ticket #78047. Name identification of remote party is to be displayed, but name identification of diverting party was displayed instead on incoming calls list.
2442 - IP241: Headset noise during firmware update
Ticket #77111. .
2519 - IP6000: Traps in DSP driver under high load
Ticket #78591. under high load timing may change. Checks in driver relaxed to take this into account.
2533 - Linux: Enable/Disable support link
Ticket #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.
2510 - MyPBX: Changing password did not work any more
Ticket #78391. Collateral damage of fix #75453: PBX/IP-DECT: User password length
2517 - PBX Boolean: Avoid unnecessary load by boolean monitoring
Ticket #78504. A call was sent to the monitoring endpoint every 10s
2522 - PBX Filter: Maximum length increased from 13 to 17 digits
Ticket #78594. 13 digits could be too little if an international number including a prefix of an external line was to be checked
2410 - PBX Mobility: Dialed digits could get lost, when using Opticaller
Ticket #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.
2511 - PBX Mobility: No-Alert checkmark as workaround for provider which do not provide Alerting
Ticket #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
2506 - PBX MWI Object: Malformed H.450 interrogation result was sent if no message
Ticket #78376. It still worked, because the wrong message was usually interpreted as no message available, which was correct, but it does not look nice in wireshark
2492 - PBX soap method SetPresence
Ticket #78092. PBX soap method SetPresence added to set the presence of a PBX user.
2477 - PBX Twinning: When calling another (twin) phone, the call was sent to the original phone also
Ticket #77829. It still works, but this waiting call could be confusing
2480 - PBX Waiting: Name Id missing in calls initiated with SOAP
Ticket #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
2520 - PBX-SOAP: Limit on size for Admin call too small
Ticket #78592. Only 4k was allowed
2354 - PBX-SOAP: UserCall new option diversion override
Ticket #75773. To initiate calls, which are not diverted by a configured CFU.
Use Info[] element with type="fty-no-cf"
2503 - PBX: Dynamic group function keys did not work with non-ascii characters in group name
Ticket #78322. Name was not correctly converted to internal utf-8 representation
2478 - PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX
Ticket #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
2495 - permit hid usage report sizes of up to 64 byte, some devices send more than 32
Ticket #78158. Jabra LINK 14201-30 responds with 33 byte to certain requests
2493 - phone ip222, ip232 : USB headsets need not to be configured anymore
Ticket #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.
2448 - Phone->PBX LDAP Search returns normalised number to be dialled by phone
Ticket #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.
2490 - phone: the call which was the active call at start of a conference was not automatically cleared on a release from remote
Ticket #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.
2512 - phone: ip222, ip232: USB headset sometimes mute after disconnect
Ticket #78414. depending on isochronous transfer state at disconnect time. seems to happen more often with relatively cheap headsets.
2497 - phone: ip222, ip232: USB headset support - Plantronics C420, GN2000 USB - MS OC Version
Ticket #78201. .
2496 - phone: ip222,ip232: Muting USB headset micro via microphone key did not work but MICROPHONE OFF was displayed
Ticket #78159. .
2504 - phone: ip222,ip232: trace USB media data only if explicitely requested
Ticket #78326. use "config add USB-HOST AUDIO /trace" if data is needed
2502 - phone: ip241: memory leak when ip230x extension module is attached and used
Ticket #78311. one packet lost per keystroke on extension module
2459 - phone: Pickup fuction key is not displayed while all alerting calls are displayed on a Partner key
Ticket #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
2417 - phone: Ring Tone Titles containing apostrophes garble phone configuration
Ticket #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.
2491 - phone_orchid: remaining call mute after remote relase for the call which was the active call at start of a conference
Ticket #78091. did not happen after a remote release for the call which was on hold at start of the conference.
2482 - RTP: Potential random trap when closing channels
Ticket #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.
2529 - show linux shutdown warning on firmware reset page
Ticket #78665. If linux is running, a warning is now shown, that linux should be shutdowned before a firmware reset is performed.
2514 - SIP: Be save against sudden death of SIP caller
Ticket #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.
2426 - SIP: Record-Route handling on outbound subscriptions
Ticket #76831. Processing of Record-Route in SUBSCRIBE response. Simply missing.
2508 - SIP: Removed b=TIAS attribute from media description
Ticket #78379. Causes trouble on some SIP providers (neotel.at)
2415 - SIP: Support for "Allow-Events: ccnr" and "Allow-Events: ccbs"
Ticket #76658. For interoperability: Support for "Allow-Events: ccnr" and "Allow-Events: ccbs"
2521 - SoftwarePhone: Default ToS value for RTP
Ticket #78593. Now the default ToS value for RTP packets is 0xB8.
2484 - Web-UI: Visually separated commands
Ticket #77954. Visually separated commands 'Clear' and 'Save' on Maintenance/diagnostics/Events. Also 'download' and 'delete' on General/License.
V9 Hotfix 10 (90600.12)
2539 - Linux: Enable/Disable configuration
Ticket #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.
2547 - PBX: Trap if enabling Unknown Registrations
Ticket #78904. Collateral Damage from
fix: #77261: Phone->PBX LDAP Search returns normalised number to be dialled by phone
V9 Hotfix 11 (90600.14)
2525 - Active Directory Replication: Editfield for Poll Timer added
Ticket #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.
2604 - AD Replication stuck after connection loss
Ticket #79541. Didn't reconnect. Side effect of after-hf10 fix
2582 - ASN.1 BER: Support for constructed octet strings
Ticket #79261. needed for PKCS#12 only decoding, only one level of nesting
2581 - ASN.1 BER: Support for indefinite-length encoding
Ticket #79260. needed for PKCS#12 import only decoding
2561 - Cisco Softkey Features, Status updates and Park
Ticket #79030. suported_mask needs more flags Softkey Feature Park
2540 - Conference: Permanent noise in conference channel
Ticket #78838. Permanent noise in conference channel. Increasing with every new participant.
2559 - Edss1 Interworking: Sending divertingLegInformation2 as PublicPartyNumber(was UnknownPartyNumber)
Ticket #79025. Such that the type of number resulting from an interface cgpn-map can also be asn.1 encoded.
2602 - Gateway: 'cn' attribute for test interfaces
Ticket #79506. The 'cn' attribute is included in the test interfaces SIG0/1 for internal tests now.
2564 - Gateway: Mapping of Message Center Number of MWI
Ticket #79039. Mapping of Message Center Number of MWI according to CGPN mapping.
2515 - H.323/SIP: Avoid delayed SDP within outgoing calls as far as possible
Ticket #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'
2595 - H.323: Media problem if PBX rtp-proxy is activated for CFNR to external destination
Ticket #79430. no voice
2551 - H.323: Media problem with SIP call-completion on IP-DECT
Ticket #78970. There was no media on the successful completion call (SDP recived in Progress was not forwarded)
2473 - HTTP-Client: MD5-sess authentication
Ticket #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.
2596 - HTTP: Could not have sockets with same ports but different remote addresses
Ticket #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.
2558 - IP-DECT: Better handling when receiving a call, while a previous call is being released
Ticket #79022. Call to service number, which cause a callback can create problems.
2562 - IP-DECT: Cisco Park
Ticket #79032. Send Park softkey event
2556 - IP-DECT: Cisco SIP conferencing
Ticket #78988. Cisco proprietary softkey events implemeneted to start a 3pty conference
2560 - IP-DECT: Feature Code for calling predefined service URIs
Ticket #79028. User for Cisco compatibility
2567 - IP-DECT: Forward Cisco Status Line Updates not related to a call
Ticket #79104. .
2563 - IP-DECT: Forward Cisco Status Line Updates related to a call
Ticket #79036. .
2598 - IP-DECT: LDAP phonebook (IP1202)
Ticket #79483. The new feature LDAP phone book for the IP1202 is added now.
2555 - IP-DECT: Radio/Master calls switched to slowstart
Ticket #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.
2591 - IP-DECT: Signal waiting calls to handset
Ticket #79349. Was only acoustic information.
2566 - IP-DECT: SRTP for an OEM device
Ticket #79061. Now SRTP is configurable for an OEM device.
2594 - IP-DECT: Subscription could get lost randomly with logout/login cycle
Ticket #79397. Problem with user database handling
2601 - IP-DECT: Three party conference with innovaphone PBX
Ticket #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.
2554 - IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: DSP packet debug didnt show some packets, version endian was wrong
Ticket #78985. cleanup
2550 - IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax failure after transfer #2
Ticket #78969. tone flag was not disabled
2535 - IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax receive max packet size increased to 1024
Ticket #78803. DSP erhält zum falschen Zeitpunkt ein close-rtp.
2489 - IP222/IP232/IP241: Ethernet link configuration
Ticket #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.
2608 - ip3010 TEL port not working in NT mode
Ticket #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.
2583 - IP810: Config Wizard
Ticket #79284. Mechanism needed to add platform specific files (IP810 uses same firmware as IP6010, IP3010, ...)
2600 - ISDN: Fix for call completion interworking
Ticket #79498. CCBS/CCNR does not work in some cases
2500 - Media Relay: Don't send dummy RTP data on incoming calls
Ticket #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
2523 - Media Relay: Don't terminate T.38 protocol in media relay, forward transparently
Ticket #78610. This should add robustness and reduce CPU load
2568 - Merge of v10 changes: Dynamic Timer Management/oem serial no
Ticket #79127. Better power management on some platforms Display of oem serial no stanard behaviour of delete on NULL
2570 - Merge of v10 changes: Firmware File check when updating improved
Ticket #79130. for OEM needs
2572 - Merge of v10 changes: Internal interfaces enhanced
Ticket #79135. for OEM support
2574 - Merge of v10 changes: Library update
Ticket #79140. .
2569 - Merge of v10 changes: Memory Management
Ticket #79129. for OEM needs
2571 - Merge of v10 changes: More ciphers
Ticket #79134. for OEM SRTP and other Certificate formats
2575 - Merge of v10 changes: permit DHCP mode change between client/disabled without reset
Ticket #79152. for OEM support
2587 - Merge of v10 changes: Support for SIP specific phone features
Ticket #79306. for OEM support
2573 - Merge of v10 changes: ZipZip Tone definition updates
Ticket #79136. should overlay normal voice
2541 - myPBX: Pass language code to reporting
Ticket #78874. For localization of the call list.
2544 - PBX Admin UI: Editing profiles was not possible if the user had too many contacts
Ticket #78883. The form is now submitted using POST instead of GET.
2606 - PBX Waiting/Broadcast: Incomplete CDRs if CFNR configured on object
Ticket #79549. If the CFNR was executed, the CDR for the call to the Waiting Queue/Broadcast object ended without release.
2578 - PBX-SOAP: Conference Id was missing in call-info for parked calls
Ticket #79243. This caused problems with the innovaphone operator
2579 - PBX-SOAP: UserClear cause should not be used for remote party
Ticket #79249. This can result in strange call clearing without tones
2590 - PBX-Trunk: Handling for Incomplete, Invalid, Busy, No Anser destinations for Media calls only
Ticket #79319. For example this should not be done for presence/dialog subscriptions
2576 - PBX: A CFB at a PBX object was executed on call to busy user
Ticket #79199. It should only be executed, if the calls exceeded the Busy On number.
2593 - PBX: Conference & VM
Ticket #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.
2584 - PBX: Include error in resulting XML if submit-object with insufficient rights was attempted
Ticket #79289. Problem with innovaphone operator.
2589 - PBX: Local objects could not be called from Nodes with escapes as expected
Ticket #79317. Was not possible to call at all, or the number was wrong
2577 - PBX: Mapping of Message Center Number of MWI
Ticket #79232. Mapping of Message Center Number of MWI.
2607 - PBX: Web User interface problem with IE and non-ascii PBX names
Ticket #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.
2542 - PBX: When doing a show users, unnecessary data was sent to the browser
Ticket #78878. The data could be reduced and thus speeding up the display of big users lists.
2537 - phone: ip222, ip232: USB controller sometimes hangs
Ticket #78824. happened with certain headsets after fast connect/disconnect sequences
2585 - phone: ip222,ip232,ip241: wrong AM/PM time display on status line
Ticket #79302. 15:19 PM was displayed instead of 03:19 PM
2588 - phone: permit to select the directories to be searched on indirect dialing
Ticket #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.
2483 - Phone: Possibility to reject incoming SIP calls with customized reason phrase
Ticket #77928. E.g. "SIP/2.0 480 Do not disturb" instead of "SIP/2.0 480 Temporarily unavailable"
2546 - Phone: Show number of voice messages on label of MWI fkey
Ticket #78894. Use "%u Message(s)" as label text for active state of MWI fkey.
2565 - Qsig Interworking: Apply interface CGPN map to DGPN
Ticket #79060. Qsig Interworking: Apply interface CGPN map to DGPN
2592 - Refactored some ASN.1 BER Handling
Ticket #79357. Became necessary in the run of: #79260: ASN.1 BER: Support for indefinite-length encoding
2530 - RTP: No check for changed media destination because of received multicast
Ticket #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.
2549 - SIP: Failed to decode presence XML from CUCM
2603 - SIP: From-URI may got lost after call transfer
Ticket #79536. UPDATE may be send out missing From-URI.
2609 - SIP: Max forward value of 32 could be too small for some provider
Ticket #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.
2610 - SIP: Re-negotiation for T38 did not work in media-relay scenarios
Ticket #79583. Bad SDP answer was generated.
2553 - SIP: Re-transmission of ACK with SDP answer
Ticket #78975. Handling of re-transmissions of 200/OK with SDP offer. ACK must contain SDP answer.
2557 - SIP: REFER does not work as expected on Gateway interfaces without registration
2660 - AC-DSP3: Switch trace off if the DSP Host interface shows an error
Ticket #80130.
2696 - AD Replication: LDAP filter encoding failed, when Poll Timer was configured
Ticket #80658. AD Replication: LDAP filter encoding failed, when Poll Timer was configured
2710 - Build Number format changed
Ticket #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.
2621 - Config: Could not dynamically set or reset /trace on the LICENSE module
Ticket #79709. Missing handling of MODULE_UPDATE.
2651 - Dect: AD replication makes IPEI disappearing on all users
Ticket #80015. Side effect of a previous fix #72672 aiming at ad-replicated and deleted objects coming back to life.
2617 - DHCP Server Identifier was cleared after editing the DHCP-Server page
Ticket #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.
2659 - DHCP Survivability mode doesn't work after a DHCP restart
Ticket #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.
2631 - Fax: Channel memory leak
Ticket #79815. Memory leak is fixed in fax channel now.
2639 - Gateway: Allow setting of system time from ISDN time
Ticket #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
Ticket #79639. The default of 4s is not good for all applications. Now configurable between 500ms and 6000ms.
2682 - Gateway: Pass through ctSetup facility
Ticket #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.
2700 - Gateway: Routing of incoming SIP calls may not work
Ticket #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.
Ticket #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.
2666 - IP-DECT/Analog Features: Call Park should be done on the last active call
Ticket #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.
2680 - IP-DECT: Anonymous endpoint information
Ticket #80421. Information about anonymous endpoint is added to the event to the DECT system. In preparation for the new feature "easy subscription".
2681 - IP-DECT: DECT security
Ticket #80424. Some changes in preparation for the new feature "DECT security". This feature will be available later.
2653 - IP-DECT: Handover with IP1202
Ticket #80022. Only one handover is possible since the last fix with the IP1202. It is fixed now.
2676 - IP-DECT: Handset display
Ticket #80402. Names with special characters are not correctly shown in the handsets. This is fixed in the IP1200 now.
2694 - IP-DECT: Idle display update
Ticket #80654. Handling of idle display update message in the call release message is added. Used with an OEM PBX.
2711 - IP-DECT: Memory leak for SIP calls
Ticket #80937. buffer for received name-id was not freed. Could happen with other facilities as well.
2642 - IP-DECT: New Master/Radio behavior for license incompatibility
Ticket #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.
2618 - IP-DECT: OEM Configuration of Cisco Features changed
Ticket #79684. should depend on installed license
2643 - IP-DECT: Reset required if Radio password changed
Ticket #79929. Now reset required is shown if the Radio password for the Master registration is changed.
2678 - IP-DECT: Signal waiting calls to handset
Ticket #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.
2689 - IP-DECT: User log in, endpoint data added
Ticket #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.
Ticket #80111. increased to 640 Bytes to allow G711 80ms.
2677 - IP210 IP230 IP240: Handset receiver volume increased, especially at lower frequencies
Ticket #80403. For better performance with jinlida receiver.
2686 - IP232: Invalid text on second line of fkey label
Ticket #80484. Invalid text on second line of fkey label when changing registrations.
2627 - IP241,IP222,IP232: Show lengthy number information on Partner fkey
Ticket #79779. Toggle between 'show head' and 'show tail' presentation.
2648 - ISDN: Problem configuring negative volume levels
Ticket #79992. Collateral Damage from 79028: IP-DECT: Feature Code for calling predefined service URIs
2613 - LDAP Server caused crash on port-scanner attack
Ticket #79634. LDAP Server caused crash on port-scanner attack
2685 - Linux: IP address with external DHCP server
Ticket #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.
2635 - memory violation in http client when processing an URI > 8000 byte
Ticket #79860. happens when the digest authentication header is constructed
2619 - myPBX launcher: Automatically move main window into visible area of screen
Ticket #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.
2698 - myPBX launcher: Configurable hotkey action
Ticket #80684. Options are: - Copy selected phone number to myPBX - Show myPBX
2626 - myPBX: Problem with checking browser capabilities in Firefox
Ticket #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.
Ticket #79636. When the connected name of a call contained the ' character there was a script error.
2652 - myPBX: Setting CFx with boolean did not work in IE8
Ticket #80018. Using Internet Explorer 8 the selected Boolean object was not saved. Internet Explorer 8 requires option tags to have a value attribute.
2451 - myPBX: Syslog for sessions and application sharing
Ticket #77289. Syslog entries should be useful for debugging session timeouts and problems with the external application sharing solution.
2656 - Name and Number configured for a registration must be completely replaced by registration result
Ticket #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 ...)
2636 - PBX Broadcast: Forking/Mobility configured at Broadcast object did not work
Ticket #79863. Mobility can be used now for forking with delay
2630 - PBX Mobility: Pick mobile call on fixed phone
Ticket #79794. by dialing number of Mobility object on fixed phone
2612 - PBX Waiting: Filter did not work anymore for CFNR
Ticket #79610. Collateral damage from
fix: #75465: PBX: CFNR Loop check detected loops that weren't
Ticket #79734. Needed if the FindUser is used to look for users which have the 'Hide from LDAP property' set.
2672 - PBX-Waiting: Calls to a Waiting queue object in altering state could not be cleared with SOAP
Ticket #80291. A SOAP UserClear issued for the call monitored at the Waiting Queue object didn't do anything
2715 - PBX: Adjusting called number, when sending call to extern
Ticket #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.
2623 - PBX: Allow call completion only if recall is not prohibited by CFU
Ticket #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.
2650 - PBX: Better handling of presence information without activity
Ticket #80006. This type of presence information could contain a note which refers to the future
2611 - PBX: CFNR at Gateway object with incomplete destination
Ticket #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.
2673 - PBX: DECT attributes for DECT security
Ticket #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.
2632 - PBX: Export to CSV, utf-8 byte order mark was missing
Ticket #79844. Without the utf-8 BOM the file was not correctly interpreted as utf-8 by Microsoft Excel and other applications.
2633 - PBX: For registrations containing name and number, use name to identify the device
Ticket #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.
2697 - PBX: More consistent use of conferenceId after transfer
Ticket #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
2657 - PBX: Possible trap on calls from misconfigured nodes/PBXs (node parent loop)
Ticket #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
2688 - PBX: Routing problem from sub-slave to master to object in sub-slave node, but registered to master
Ticket #80591. Collateral damage from #77874: PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX
2708 - PBX: Routing problem with nodes/escapes/slaves with calls to extern numbers in other PBX defined nodes
Ticket #80853. Could happen that the number dialed when sending the call to the extern interface was not adjusted correctly
2616 - PBX: Status displayed on boolean function key could be wrong
Ticket #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.
2695 - PBX: Update of conference id did not always work after multiple transfer accross PBXs
Ticket #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.
2647 - PBX: When editing a Node object it was changed to a PBX object
Ticket #79982. This was a collateral damage of fix: #78878: PBX: When doing a show users, unnecessary data was sent to the browser
2683 - Phone: Accept MWI from Exchange Server
Ticket #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.
2675 - phone: ip222, ip232: Jabra USB Headset feature "Reject incoming call" supported now
Ticket #80401. A double tap on talk button rejects a ringing call with 'busy'
2668 - phone: ip222,ip232: more USB headsets supported
2703 - Phone: Message function key cannot be configured
Ticket #80766. Configuration is not saved
2646 - Phone: Out-Of-Memory-Trap
Ticket #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.
2644 - Phone: PBX (operator) initiated outbound call was connected but mute when another call was ringing on phone already
Ticket #79943. the phone was not switched from ring mode to handsfree/headset mode
2671 - Phones: Show 'tel' presence on configuration screen
Ticket #80274. Show latest presence on 'Presence' fkey, but show 'tel' presence on phone config menu.
2674 - Phones: Sometime fkey could not be deleted
Ticket #80389. Sometime fkey could not be deleted when using the fast edit mode (long fkey press).
2669 - Phones: Switch for phoneapp to disable auto-answer
Ticket #80233. Disable/enable auto-answer support on phoneapp level.
2637 - Potential trap in packet libraray
Ticket #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.
2662 - RPCAP trace: Indicate transmit/received packets by setting the remote mac address to 00-90-33-00-00-00
Ticket #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!
2690 - SIP: Calls may remain in clearing state
Ticket #80623. SIP calls may remains undeleted.
2634 - SIP: CLEARMODE does not work since Hotfix10
Ticket #80635. Reduce number object constructions/destructions during message encoding.
2664 - SIP: Content of Allow-Events header must be treated case-insensitive
Ticket #80173. Event names are case insensitive.
2706 - SIP: Diversion information was wrong on incoming SIP calls
Ticket #80812. Parsing of History-Info header was wrong, but only if multiple History-Info headers were present.
2629 - SIP: DNS resolving of STUN server failed
Ticket #79788. Wrong STUN server port used (5060 instead of real port).
2638 - SIP: Domain Name System (DNS) names compared case sensitive
Ticket #79887. Domain Name System (DNS) names are "case insensitive"
2670 - SIP: Handling of 488 for encrypted media
Ticket #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.
2624 - SIP: Handling of Call Pickup
Ticket #79741. Failed to handle INVITE with Replaces as Call Pickup. Must redirect Pickup call to alerting party.
2640 - SIP: Interworking with "AUDC-IPPhone"
Ticket #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.
2645 - SIP: Interworking with HD audio client
Ticket #79953. Support for unknown codecs. Passing transparentley.
2661 - SIP: Keep backup registration while calls are active
Ticket #80137. Terminating backup registration will active calls (through backup system) get disconnected.
2701 - SIP: Memory leak when closing signaling interface
Ticket #80752. Memory leak when closing signaling interface while DNS request is pending on a call.
2687 - SIP: New config option /take-zero-addr-for-hold
Ticket #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.
2691 - SIP: Pass display information to application
Ticket #80632. Pass display information received in Call-Info header in 200/OK for BYE to app.
2605 - SIP: Pass display names of <dialog-info> to phoneapp
Ticket #79543. Pass display names of <dialog-info> to phoneapp.
2641 - SIP: Problems with DNS resolving of proxy adresses
Ticket #79907. If resolving of the primary proxy failes, the backup proxy is never resolved.
2665 - SIP: Support for Alert-info:<Bellcore-dr1>
Ticket #80174. Proprietary tagging of internal calls
2714 - SIP: Transcation handling was wrong
Ticket #80989. Double delete on a INVITE client transaction object.
Ticket #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.
V9 Hotfix 13 (9061024)
2764 - Chrome compatibility issue on PRI statistics page
Ticket #81859. If clear was clicked an error message was displayed
2770 - Debug information on assertion
Ticket #81973. More debug information on default event handler.
2722 - H.323: Memory leak when sending special OEM H.235 key elements
Ticket #81160. Only happens if Avaya SRTP is enabled in OEM products
2723 - H.323: Potential Trap in case of high load and media renegotiation/call clearing collision
Ticket #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.
2750 - HTTP: Possible trap on many simultaneous sessions
Ticket #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
2730 - IP-DECT: Duplicate call setup
Ticket #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.
2745 - IP-DECT: Memory leak with special Cisco features
Ticket #81463. Only with special OEM features.
2749 - IP-DECT: Mobility Master registration
Ticket #81571. The Mobility Master does not accept clients with the OEM name IP1202. This is fixed now.
2731 - IP-DECT: OEM configuration option
Ticket #81214. The configuration format of an OEM configuration option was changed for the backward compatibility.
2755 - IP-DECT: Own priority for idle display set by messages
Ticket #81616. The idle display set by messages has got an own priority now.
2752 - IP-DECT: System GUI, disabled local coder options
Ticket #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.
2753 - IP-DECT: System settings not to dynamically connected radios
Ticket #81612. System settings should not be sent if the radio-master connection is a dynamic one (IP1202 only). This is fixed now.
2735 - IP0010 IP1060 IP3010 IP6010: During long reset the ready LED was blinking orange and the Ethernet LEDs were swaped
Ticket #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.
2727 - IP1060/3010/6010/22/24/28/302/305: sequence number to DSP changed from byte to word, caused jiter buffer warnings
Ticket #81189.
2649 - IP222 equalizer update
Ticket #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
2758 - IP222 IP232 IP241: Force same speed of the switch ports for 1000M/100M scenarios
Ticket #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.
2654 - IP222 IP232 IP241: repeated ethernet link status 1000M wrong
Ticket #80029. .
2739 - ip222,ip232: waiting calls mute after having been accepted/connected on USB headsets, mainly seen with Jabra LINK 14201-30
Ticket #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.
2728 - IP222/232/241/1060/3010/6010/22/24/28/302/305: min jitter buffer changed to 10ms
Ticket #81197. was 35ms, now 10ms as on ip6000/800
2725 - IP222: Handset receiver volume increased
Ticket #81175. By 6db, so the volume similar to IP241, but still has low THD+N
2761 - IP232: Show more of collapsed fkeys
Ticket #81807. 140px when expanded. 100px when collapsed (80px before).
2724 - IP241 handset equalizer
Ticket #81174. Enable IP241 handset micro and speaker equalizer. Same values as on IP222. Reduces noise in G711 mode.
2765 - PBX Admin UI: Not possible to change DECT parameters for users with cf/grp admin rights
Ticket #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
2763 - PBX Mobility: No response for call to busy mobile phone, if no fixed phone
Ticket #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.
2709 - PBX-SOAP: UserRedirect - original called number got lost
Ticket #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.
2734 - PBX: Handling of call limits at PBX objects improved
Ticket #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"
2741 - PBX: Objects list filter for numbers did not work correctly anymore
Ticket #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
2679 - phone: ip222, ip232: Pressing Talk button on USB Headset when there is an active and a held call does not transfer
Ticket #80413. Instead of transfer the active call is disconnected and the held call is retrieved
2769 - phone: ip222,ip232: more USB headsets supported
2771 - phone: ip222,ip232: support for advanced USB headset functions (redial, reject call, accept waiting call)
Ticket #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.
2768 - phone: ip222,ip232: when a call is released from remote while another call is waiting the waiting call cannot be accepted.
Ticket #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.
2732 - phone_orchid: micro mute when a waiting call was connected after an outbound call setup had been cancelled
Ticket #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
2762 - SIP: Double Replaces header in INVITE after REFER
Ticket #81847. Reject by "Swyx IpPbxSrv/8.1.0.246" with "400 Bad Request(Multiple values in single-value header Replaces)"
2719 - SIP: Handling of multiple 401/407 responses
Ticket #81045. Implemented handling of up to 3 401/407 responses per transaction.
2773 - SIP: Memory leak in SIP stack
Ticket #82058. Memory leak in SIP stack.
2760 - SIP: New config option "No Inband Disconnect" on GW interfaces
Ticket #81803. Introduced VOIP_OPTION_NO_INBAND_DISC. Option was missing on Gateway interfaces.
2751 - SIP: New config option /no-certificate-check
Ticket #81601. New config option to disable validation of remote certificate name when opening outbound TLS connection.
2767 - SIP: New config option /product-id-format
Ticket #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
2736 - SIP: New config option for endpoints not refreshing their registration during call
Ticket #81243. Interoperability: New config option /keep-active-endpoints for endpoints not refreshing their registration during call.
Ticket #81089. Timer fires during call release and causes trap.
2717 - SIP: Un-escape content of XML elements
Ticket #81019. XML element content requires some resevered characters to be escaped (<>). These escape sequences (> or <) must be un-escaped onthe receiving end.
2756 - SoftwarePhone: Support for Jabra PRO 930
Ticket #81618. Support for Jabra PRO 930 with product id 0x1016 added.
2757 - SoftwarePhone: Trap with trace and mute ringer event
Ticket #81619. A trap occurs if the trace is enabled and the ringer mute option is toggled.
2754 - Support for old versions of Jabra BIZ 2400 headset with 48 kHz output sampling frequency
Ticket #81613. the newer versions come with the usual 16 kHz sampling frequency but have the same signature (vendor/product) as the older ones.
2738 - UI hint where to update Linux AP
Ticket #81295. Changed a hint on the upload/update tab and added a link to the update/upload page on the firmware side.
2737 - X.509: Creating certificate containing IPv4 address did not work
Ticket #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.
2786 - DHCP: Name registration at WINS was not refreshed as requested by TTL in registration response
Ticket #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.
2845 - External-UC: Presence info assigned to wrong PBX object
Ticket #83020. When presence eventlist is received from External-UC all presence info was assigned to same PBX object.
2805 - failure of analog ports of ip28
Ticket #82488. ip28 analogue ports do not react to incoming calls and hook-off. Problem could only be solved by reset.
2790 - Gateway CDR with '0. 0' charge amount
Ticket #82359. Should be '0.00' instead
2801 - Gateway: 'Enable PCM' option added at the CONF interface of the IP800
Ticket #82481. The option 'Enable PCM' is available at the CONF interface of the IP800 now.
2861 - Gateway: No Media received event was generated for T.38 calls, which started without actual RTP Traffic
Ticket #83433. This happend for calls to/from Fax servers, which switch to T.38 without sending any RTP packets first.
2860 - Gateway: On IP24, IP302, IP305 it could happen that multiple routes disappeared if a route was deleted
Ticket #83278. Hard to predict when this happened.
2802 - Gateway: PCM mode of the CONF interface fixed
Ticket #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.
2804 - Gateway: Routing problem with blockdial route and following matching non-blockdial incomplete routes
Ticket #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"
2832 - H.323: Support for Avaya SRTP with AES128/80
Ticket #82829. Needed for OEM
2793 - H.323:No Media for calls with reverse media to a H.323/SIP exclusive Code Media Relay interface
Ticket #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.
2774 - HTTP-Server: Configuration of "Public compact flash access" did not work for all cases
Ticket #82064. E.g. /DRIVE/CF0/Neuer Ordner/ does not work, because HTTP request contains escaped sequences.
2729 - Incorrect disk usage calculation for more than 4GB
Ticket #81209. The disk usage calculation was wrong, if more than ~4 GB of the card were used.
2826 - IP-DECT: Call transfer compatibility with SIP
Ticket #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.
2777 - IP-DECT: DECT security (2)
Ticket #82191. Some changes in preparation for the new feature "DECT security". This feature will be available later (IP1202).
2794 - IP-DECT: DECT Users Administration Log-in
Ticket #82422. The GW-DECT module's users administration log-in is fixed.
2782 - IP-DECT: Display info with failed user log-in
Ticket #82200. If the user log-in fails, the cause is shown in the display.
2797 - IP-DECT: Display info with remote control call
Ticket #82444. The display info shown after accepting a remote control call is fixed now.
2784 - IP-DECT: Hanging aborted semi-attended call transfers with SIP
Ticket #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.
2781 - IP-DECT: Idle display update
Ticket #82199. Handling of idle display update message in the call release message is added. Used with an OEM PBX.
2858 - IP-DECT: LDAP replication alarm loop
Ticket #83249. If the LDAP replication is configured, but it can not be synchronized, there is an alarm loop. This is fixed now.
2785 - IP-DECT: LDAP server GUI description
Ticket #82258. The LDAP server GUI description is wrong and corrected now. (The LDAP server can not be a HTTP server.)
2779 - IP-DECT: No delay with semi-attended call transfer and SIP
Ticket #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.
2780 - IP-DECT: No fall-back after unattended call transfer
Ticket #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.
2783 - IP-DECT: Short tone info by PBX only one time
Ticket #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.
2854 - IP-DECT: Support for more than 32 LDAP attributes
Ticket #83191. The IP-DECT devices support up to 256 LDAP attributes now.
2776 - IP-DECT: Trap during subscribing handsets
Ticket #82190. A trap occurs during subscribing handsets on the IP1200. This is a fix for the previous fix #80424 and fixed again.
2778 - IP-DECT: Wrong release code
Ticket #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.
Ticket #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.
2799 - IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: reduce probability of false DTMF detection
Ticket #82470. Change DTMF signal to noise ratio from 12db to 18db
Ticket #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.
Ticket #82551. 660.27 is the released version on the audiocodes web. Fax relay and modem bypass improved.
2838 - IP0010 IP1060 IP3010 IP6010 IP810 IP302 IP305: ISDN: Enable fax detection only after connect - fixed
Ticket #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
2839 - IP1060: Memory size is not correctly shown
Ticket #82906. The memory size is not correctly shown in the IP1060 in diagnostics counter page.
2842 - IP222 IP232 IP241: New equalizer and volume setting
Ticket #82952. - use equalizer up to approx 6db - use digital volume to adjust volume
2823 - IP222 IP232 IP241: Reduce DTMF level
Ticket #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)
-->
2810 - IP222 IP232 IP241: update to DSP code 660.27
Ticket #82554. 660.27 is the released version on the audiocodes web. Improved acoustic echocanceller.
2824 - IP222,IP232: Going offhook in call list always dials first list entry (not touched list entry)
Ticket #82732. Going offhook in call list always dials first list entry (not touched list entry)
Ticket #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
2821 - IP3010 IP810: number of DSP channels and number of conference channels was wrong.
Ticket #82675. Now the IP810 shows 20 DSP channels and 30 conference channels. Now the IP3010 shows 42 DSP channels and 60 conference channels.
2800 - IP800 IP6000: reduce probability of false DTMF detection
Ticket #82471. Change DTMF detector signal to noise ratio from 12db to 18db.
2792 - ipv6: memory leak when sending fragmented packets
Ticket #82394.
2795 - Kerberos: Allow editing multiple fields in admin UI
Ticket #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.
2825 - Memory leak after Firmware or Bootcode download
Ticket #82740. After a Firmware or bootcode download via Maintenance/Download the associated command processor instance was not released.
2830 - myPBX: Show version of launcher in the list of sessions
Ticket #82821. The version of the used launcher is now shown on the page PBX/myPBX.
2788 - myPBX: Support contact names containing a single quote
Ticket #82323. Configuring contacts with a H.323 id containing a single quote (') caused script errors in the web application.
2836 - PBX Broadcast: No diverting name sent with broadcasted call
Ticket #82880. The diverting and original-called name info was missing from the diverting leg2 info generated by the broadcast object
2815 - PBX Mobility: Call to mobile phone was sent with invalid diverting information
Ticket #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.
2847 - PBX mobility: The forking destination put in for mobility at a user should be dialed from the node of the user
Ticket #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
2812 - PBX Mobility: Trap in case of Transfer of a call from a mobile endpoint to another mobile endpoint
Ticket #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
2844 - PBX Routing: A CFNR at a PBX object for WAN re-routing did not work if it contained escapes
Ticket #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.
2841 - PBX Routing: Node extern did not work for calls from a trunk marked as local object
Ticket #82948. The call was not routed back to the originating slave
2814 - PBX Trunk: Number to Name Feature did not work for calls to busy endpoints
Ticket #82619. Such a call was not rejected with cause 'User Busy', but was just hanging.
2740 - PBX Waiting: Call state shows "Disconnecting" after switch from announcement 1 to announcement 2
Ticket #81370. Announcement disconnect changed state
2834 - PBX Waiting: Evaluate Busy on ... Calls for calls to an operator
Ticket #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.
2746 - PBX-Mobility: Data Call Thru
Ticket #81513. To speed up dialing with the mobility client, the called number can be posted to the PBX via HTTP before the call
2835 - PBX: Avoid signaling loop after call transfer
Ticket #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.
2856 - PBX: Description was missing for DECT System object
Ticket #83198. A configurable description is useful for the DECT System object as for all other objects
2766 - PBX: Don't forward in-band info indicator if no media channel
Ticket #81879. Not good to indicate in-band info available if no media channel can be negotiated
2846 - PBX: Editing Config Templates impossible, if by some old firmware a strange config ended up in a User object
Ticket #83033. The problem was a empty hardware id (hw=""). This is now ignored.
2789 - PBX: RTP-DTMF was disabled by Voice Mail object during re-negotiation
Ticket #82332. RTP-DTMF acc. to RFC-2833 was disabled by Voice Mail object during re-negotiation.
2849 - PBX: Standyby PBX generated alarms for missing slave registrations, even if active PBX up
Ticket #83089. There should be only an alarm, if the standby PBX is active
2855 - PBX: Support for more than 64 LDAP attributes
Ticket #83192. The PBX supports up to 256 LDAP attributes now.
2743 - PBX: Switch from Music on Hold to inband ringback on Alert after unpark oder transfer
Ticket #81407. To give the caller feedback that soon somebody may answer the call
2857 - Permit logging to a second (shadow) log server
Ticket #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.
2840 - phone: a calling party name found by inverse directory lookup for an external call was sometimes not stored in call list
Ticket #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.
2791 - phone: call completion did not recover when DND(busy) was set at the phone requesting the call completion
Ticket #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.
2811 - phone: call forwarding options offered even when call forwarding was not possible
Ticket #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).
2807 - Phone: Display text received with BYE
Ticket #82525. Pass display text to phoneapp.
2775 - phone: ip222,232.241: no notification tone on a successfull redial attempt
Ticket #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.
2819 - phone: ip222,ip232: Plantronics APU70 - Savi 7xx - Radio Link not cleared on release from remote
Ticket #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.
2831 - phone: ip222,ip232: USB headset mute after activation of changes in codec parameters, headset icon cleared on status line
Ticket #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.
2817 - phone: ip222,ip232: the variable KEYS0/HID-MAP permits to map new USB headsets to builtin descriptors
Ticket #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.
2501 - phone: support PBX-directory access via TLS
Ticket #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.
2796 - phone_inca: when a call completion was set up with CLIR active the called party was not displayed on a possible recall
Ticket #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")
2787 - Phones: Presence info during ringing state may show garbage data
Ticket #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.
2828 - SIP: Bug in digest authorization
Ticket #82761. Sometimes wrong method is used in digest calculation.
2820 - SIP: Mobility did not work due to RTP-DTMF
Ticket #82674. DTMF must be passed through signaling channel to get mobility working. Suppress RTP-DTMF capability in SDP answer also.
2863 - SIP: Multiple subscriptions for 'message-summary'
Ticket #83469. After every RAS_START another subscription for 'message-summary' is established.
2850 - SIP: One-way audio after mutual hold on dect systems
Ticket #83126. Dect ep gets FTY_HOLD_NOTIFY but no FTY_RETRIEVE_NOTIFY.
2852 - SIP: Problems with presence signaling on External-UC link
Ticket #83177. Missing parameters on Contact-URI in 200/OK for SUBSCRIBE(presence): maddr transport
Ticket #82616. Bug in base64 decoding of SRTP key.
2808 - SIP: Trap during call handling
Ticket #82544. Trap during call handling
2827 - SIP: Trap if multiple calls arrive at same time
Ticket #82743. Trap if multiple calls arrive at same time.
2843 - SIP: Trap on out of memory
Ticket #83004. SIP-Client allocations not deleted. Outbound control calls without facility interworking.
2816 - SIP: Trap on subscription handling
Ticket #82623. Trap on RAS_REGISTRATION_VERIFY between sending SUBSCRIBE and receiving 200/OK.
2772 - SIP: Trap when configuring user presence
Ticket #81996. Trap may occur when configuring user presence.
2853 - SoftwarePhone: HID Support for new headsets
Ticket #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.
Ticket #82896. Misplaced reset-required indication on ISDN interface config (TEL1,TEL2,...).
2851 - X.509: Avoid alarms on missing system time after reboot
Ticket #83137. Allow 60 seconds for setting the system time before an alarm is set. Also certificates are now rejected silently, in that time.
V9 Hotfix 15 (9061078)
2864 - Alarm/Event handling: Authentication for received remote Alarms/Events
Ticket #83603. Allow by configuration to only accept authenticated alarms or events
2888 - CDR fixes for external call detection
Ticket #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.
2910 - H.323: Accidential fallback to slowstart if faststart response received in PROGRESS
Ticket #84601. Only happened in some H.323/SIP interop scenarios
2922 - H.323: Media Negotiation did not work for call with reverse media and media response in CALL-PROC
Ticket #84848. Caused SIP interop problems with CUCM
2915 - H.323: More information on "Unexpected Message" event
Ticket #84699. The message type and the state for which this message was unexpected is needed to find out what the problem is.
2905 - Hide LDAP Server Password For Viewer Accounts
Ticket #84557. Was accessible for viewers
2889 - http client : authentication was not retried after a failure when the offending request was repeated in the same session
Ticket #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.
2878 - HTTP: Chunked transfer fails if the last 2 bytes of the chunk header are in the next tcp packet
Ticket #83986. Seen with the application platform as broken pipe
2874 - HTTPCLIENT: Allow configuration of a http authenticated URL even if the server needs no authentication
Ticket #83900. If a password was configured and the server didnt need it, a HTTP put created a file of zero length.
2930 - IP-DECT: Call transfer timer
Ticket #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.
2881 - IP-DECT: Phone book 'Use TLS' option
Ticket #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.
2934 - IP-DECT: Phone book error events
Ticket #85001. The phone book module sends an error event now if the LDAP directory search fails.
2931 - IP-DECT: Reverse phone book
Ticket #84953. The reverse phone book does not work till hotfix 11. This is fixed again now.
2866 - IP-DECT: Trap during subscribing handsets
Ticket #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
2912 - IP0010 IP1060 IP3010 IP6010 IP22 IP24 IP28 IP302 IP305 IP800 IP6000: Minifirmware not shown on LED
Ticket #84616. Minifirmware should blink long green short red. This happened only after a firmware/bootcode update.
2886 - IP222 IP232 IP241: Force same speed of the switch ports for 1000M/100M scenarios (configuration option added)
Ticket #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.
Ticket #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
2893 - IP222 IP232 IP241: pressing speaker key when phone is in handset mode switches to handsfree mode instead to toggle monitor mode
Ticket #84297. handset/headset plus speaker is not supported
2911 - IP222 IP232 IP241: Updated gain and equalizer setting
Ticket #84605. Tuned for high MOS values according to ETSI ES202737 ES202738 ES202739 ES202740
2943 - IP232,IP222,IP241: Do not hide configured fkey if neither text nor icon is displayed
Ticket #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.
2919 - IP232,IP222,IP241: Rendering errors when trying to use backround image with indexed colors
Ticket #84814. PNG mode "indexed colors" is not supported. Trying to use an indexed color png leaves phone screen in bad shape.
2891 - IP6000: Prevent blinking error LED on old IP6000 with HW-Build <110
Ticket #84227. Conference DSP driver was started on old hardware that doesnt support the conference DSP
2925 - IPVA: V10 Code Merge Aiming To Stabilize FW Upload To DRAM
Ticket #84862. Addresses a problem in conjunction with innovaphone's automated test environment
2923 - Maximum LDAP PDU Size Too Small
Ticket #84851. Maximum LDAP PDU Size Too Small. Now internal maximum allocation unit plus a bit for encoding overhead.
2896 - myPBX: Inconsistent display of group monitoring rights
Ticket #84350. In the visibility settings groups were not displayed if the membership is "dynamic out". But members of that group still have monitoring rights.
2907 - myPBX: Only send one command at a time
Ticket #84568. Queue commands instead of sending overlapping commands. This limits the number of open AJAX connections to two.
2904 - myPBX: Pass selected device to launcher
Ticket #84556. Needed for version 10 remote video
2869 - myPBX: Remove unimportant notifications
Ticket #83707. The following events are not so important that the user has to be notified using a windows bubble notification. - Visibility requests - Missed calls
2921 - NAT: Don't forward DNS requests from public network
Ticket #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
2862 - New remote control codes to be used for phone tests by soap applications
Ticket #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)
2901 - Page Ldap/Replicator/Status didn't display in WebKit
Ticket #84400. Page Ldap/Replicator/Status didn't display in WebKit
2924 - PBX Broadcast: Call to group members was not cleared when CFNR was executed
Ticket #84857. This was a collateral damage from fix: #79549: PBX Waiting/Broadcast: Incomplete CDRs if CFNR configured on object
2935 - PBX Routing: A CFNR on a slave gateway object redirecting the call to the master failed
Ticket #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
2882 - PBX Waiting: Call to operator with Twin Phone Checkmark did not work anymore
Ticket #84084. Collateral damage of
fix: #82858: PBX Waiting: Evaluate Busy on ... Calls for calls to an operator
2876 - PBX Waiting: Trap on collision of operator connect and two-stage dialing
Ticket #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
2867 - PBX Waiting: Trap on leak-check if dtmf maps are configured
Ticket #83691. Only happened if debug.xml leak check was used
2877 - PBX: Call to a Trunk/Gateway was not marked correctly as external, if no connected number was received
Ticket #83940. Calls from a Trunk must be explicitly marked as internal with respective connected number, otherwise they should be treated as external
2926 - PBX: Filters should only be applied to calls with media channels
Ticket #84892. A join group operation could be prohibited with filters. This was unexpected.
2897 - PBX: Import did not work with some data
Ticket #84356. The data is processed in chunks. If the chunk border was right behind the closing </user> tag, the decoding of the next chunk failed.
2871 - PBX: New Feature to allow registration with password to devices regardless of address filter
Ticket #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.
2880 - PBX: Partial Rerouting was prohibited in Alerting State (CFNR)
Ticket #83993. This was done under the assumption the partial rerouting is not supported as CFNR by public networks.
2865 - PBX: Preparations for objects visible only if appropriate license installed
Ticket #83615. for v10
2879 - PBX: Registration with Name/Number did not work correctly if default device not first
Ticket #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
2944 - PBX: Slave with non-ASCII PBX name did not register at master
Ticket #85235. Error in utf-8 to unicode convertion in this case
2906 - phone: enable directory search function key also in connected state
Ticket #84559. it's sometimes useful to browse the directories for a number while talking.
2927 - phone: ip222, ip232: raise alarm if an USB Headset does not respond on USB bus anymore
Ticket #84893.
2913 - phone: ip222, ip232: status stage added to all USB control transfers
Ticket #84617. the status stage is mandatory as well for IN as for OUT control tranfers
2885 - phone: provide complete dialog info to a phoneapp
Ticket #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.
2899 - phone: status messages for outbound external calls were sometimes garbled on display
Ticket #84365. happened when for a preceeding outbound internal call in alerting state presence info was displayed
2898 - phone: when scrolling directory search results sometimes one of the numbers of a contact was not displayed
Ticket #84362. the tag characters assigned to the different numbers were not included in sort order.
2902 - phone_orchid: displaying both dialed and connected number may be misleading, it's better to omit dialed number
Ticket #84422. when for example 022222222 was dialed and the network reported a connected number 03022222222 the display info "022222222 -> 03022222222" looked like a transfer.
2887 - simple static logging interface
Ticket #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.
2917 - SIP: Announcing "a=T38MaxBitRate:14400" in T.38 offer
Ticket #84770. Announcing "a=T38MaxBitRate:14400" in SDP offer for T.38 since all current devices support that modem speed.
2928 - SIP: Fix for media negotiation on calls re-routed from TONE interface to outbound SIP
Ticket #84932. Helps on gateway interfaces configured for media-relay with exclusive codec.
2940 - SIP: Fix for STUN problems
Ticket #85118. Try alternative STUN server address if first fails and another was gathered by DNS.
2933 - SIP: Loosing remote IP addresses when DNS becomes temorarily unavailable
Ticket #84991. Loosing remote IP addresses when DNS becomes temorarily unavailable. Results in interfaces without remote ip addresses.
2938 - SIP: Memory leak
Ticket #85083. Memory leak.
2929 - SIP: Set numbering plan to "ISDN/telephony" in case of "Bellcore-dr2" as Alert-Info in INVITE
Ticket #84939. Set numbering plan to "ISDN/telephony" in case of "Bellcore-dr2" as Alert-Info in INVITE.
2936 - SIP: Switch to T.38 did not work when interworking with H.323 slowstart (XCAPI)
Ticket #85047. Switch to T.38 did not work when interworking with H.323 slowstart (XCAPI).
2918 - SIP: Trap during channel handling
Ticket #84800. Rare trap when re-assigning channels.
2883 - SoftwarePhone: Product string
Ticket #84119. The product string is changed, used e.g with the PBX registration.
2875 - SoftwarePhone: Signature added
Ticket #83915. The SoftwarePhone installer and the install package is signed now.
2872 - Voicemail: Prevent <prompt>, <record> on incoming control calls
Ticket #83826. Turned out to cause idle-reset requests never being processed.
Ticket #84171. The vertical resolutions 96 lpi, 196 lpi and 400 lpi are correctly saved in the SFF file with a proprietary definition known by the tool sfftobmp.
V9 Hotfix 16 (9.061101)
2999 - AD Replication: Configuration Buffer Increased
Ticket #86211. Was too small for many maps
2993 - Change order of links on debug page
Ticket #86121. Make "Tracing" the first item so that no leak check is triggered when opening the page.
2983 - DHCP: client IP connectivity lost when a renew/rebind request for the currently assigned address was refused by the server
Ticket #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.
2998 - Disable leak check if debug flag is not set
Ticket #86165. The leak check is only allowed if the config flag CPU /debug is set.
2984 - ENUM: Port in SIP-URI was not honored
Ticket #86037. Port in SIP-URI was not honored in regex of DNS result.
3026 - Gateway: New config option "No blind transfer"
Ticket #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.
3008 - Gateway: Routing on blind transfer call starts from wrong IF
Ticket #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.
2941 - H.323: Interop Problem with CUCM SIP for Transfer
Ticket #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.
2996 - HTTP-Client: Requests with specified credentials
Ticket #86133. Additional function for OEM httpclient::auth_request It uses the specified credentials and ignores the username and password from the configuration.
2969 - IP-DECT: Call transfer compatibility with SIP
Ticket #85795. For compatibility reasons with the SIP protocol no hold signal should be sent to the call transfer destination. This is fixed again.
2956 - IP-DECT: Configuration changes for OEM device
Ticket #85522. For an OEM device the configuration update behaviour is changed.
2962 - IP-DECT: DECT security (3)
Ticket #85554. Some changes in preparation for the new feature "DECT security". This feature will be available later (IP1202).
3011 - IP-DECT: GUI Master Configuration
Ticket #86442. Disabling LDAP Directory Search fixed for the IP1202.
2987 - IP-DECT: MAC-alias change of OEM device
Ticket #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.
3020 - IP-DECT: New hidden Radio option 'Max RTP streams'
Ticket #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.
2976 - IP-DECT: Radio reconnect handling with OEM PBX
Ticket #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.
2985 - IP-DECT: Transferred/rerouted call display
Ticket #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.
2989 - ip1202: Send Inband DTMF did not work
Ticket #86052. the "DECT/Config/Master/Send Inband DTMF" checkmark had no effect.
2988 - ip1202: support capture of raw ethernet packets exchanged between MSP (aka DSP) and the firmware
Ticket #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.
2945 - IP152: Call replacement (blind transfer) did not work
Ticket #85313. Call replacement (blind transfer) ends up in no audio.
2961 - IP222 IP232 IP241: DSP Update to Version 680.05
Ticket #85552. Sporadic problems with noise reduction fixed. Noise reduction can be enabled for Handset/Headset/Handsfree mode
2955 - IP232,IP222: Automatically enter input mode 'alpha' when entering directory search screen through fkey
Ticket #85493. Automatically enter input mode 'alpha' when entering directory search screen through fkey.
3002 - ip72 firmware did not boot anymore since V9hotfix11
Ticket #86246.
3013 - Linux: GUI in viewer mode
Ticket #86455. The Linux General page is fixed for the viewer only mode.
3021 - Linux: GUI removed from IP1060
Ticket #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.
3009 - Linux: Start-up failures
Ticket #86399. Linux start-up is improved. This fixes hanging Linux start-ups caused by compact-flash failures.
2960 - Media: RTP did not work on switch from slowstart T.38 to EFC G.711
Ticket #85548. T.38 retransmission packets were sent, which caused problems with the RTP
2959 - PBX-Exec: Call at secretary for executive had wrong destination name/number
Ticket #85535. Call should be displayed as call to the scretary, diverted by the executive
3016 - PBX: Conference trap
Ticket #86473. Potential trap in the PBX BC conference object is fixed.
2951 - PBX: Potential trap on registration of a slave PBX, with Master GK-ID configuration changes
Ticket #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.
2997 - phone: coder settings of a "Create Registration" function key were not applied to the created registration
Ticket #86164.
3006 - phone: headset function key mode 'control' could be configured via WEB interface only
Ticket #86327. must be possible locally at the phone too
3027 - phone: ip222, ip232: handset/handsfree speaker was not switched off when changing over to headset mode
Ticket #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.
2995 - phone: ip222, ip232: recover from USB port failure probably caused by electric sparks
Ticket #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.
2903 - phone: ip222, ip232: USB Headset could not be disabled via Menu or by Headset Function in Enable mode
Ticket #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.
3007 - phone: Partner state provided via dialog info was not reset when the subscription call was released because of network errors
Ticket #86330.
2972 - phone_orchid: configuration prameters to adjust microphone and speaker volume
Ticket #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.
3028 - phone_orchid: pressing the speaker key while in headset mode did not switch over to handsfree mode
Ticket #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.
2971 - Phones: New config file option /recording-without-remote-party-info
Ticket #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
3001 - Phones: New config option "Allow User Settings at Phone"
Ticket #86243. Allow changing of User Settings even if 'Protect Configuration at Phone' is activated.
2977 - Phones: Touching should start dialing of selected directory entry
Ticket #85959. Touching should start dialing of selected directory entry instead of opening directory entry for editing. At least in Context of Wahlvorbereitung.
2994 - Phones: Using config option "CGPN" as CGPN on outbound SIP calls
Ticket #86122. Using config option "CGPN" as CGPN on outbound SIP calls.
2949 - Presence note got lost
Ticket #85469. When configuring a presence on the phone (Main menu/User setup/Presence) the note is not save along with the selected activity
2953 - Presence: Note moves into 'tel' presence
Ticket #85482. Presence note moves into 'tel' presence when changing activity by use of 'presence' fkey.
3004 - SIP: Avoid re-configuration of DSP channel when processing re-INVITE
Ticket #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.
2968 - SIP: Do not send "504 Server Time-out" after "200 OK"
Ticket #85788. Do not send "504 Server Time-out" after "200 OK".
2973 - SIP: Handling of "sendonly" offer after "inactive" offer
Ticket #85857. Switching from "inactive" into "recvonly". Giving REMOTE_RETRIEVE and HOLD_NOTIFY to app.
2975 - SIP: Ignore From-URI in re-INVITE and UPDATE when "from-change" not supported
Ticket #85873. Ignore From-URI in re-INVITE and UPDATE when "from-change" not supported by remote side.
2957 - SIP: New config file option /fixed-contact-addr
Ticket #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)
2948 - SIP: New config file option /tls-unchecked
Ticket #85449. New config file option /tls-unchecked disables validation of remote server certificate.
2982 - SIP: Optimization when validating local media address
Ticket #86029. Optimization when validating local media address. Keep number of SOCKET_GET_LOCAL_ADDR low.
3015 - SIP: PRACK after CANCEL contains bad RAck header value
Ticket #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.
2966 - SIP: Providing private RTP address although public RTP address is available
Ticket #85751. During re-negotiation: Providing private RTP address to external endpoint although public RTP address is available.
2992 - SIP: Support for multiple audio media descriptions
Ticket #86083. One without encryption and one with encryption:
Ticket #86569. In case of unconnected calls that were subject of a prior <pbx-fwd>
3000 - whistling tone in all ip28 a/b ports on incoming call
Ticket #86212. This phenomena occured after few day uptime after sending CLIP
V9 Hotfix 17 (9061152)
3017 - AD Replication: Refactoring Poll-Timer-based Replication
Ticket #86476. -Ridded usnChanged filter term -Added two status messages
3159 - Admin IP: static routes configured at an ETH interface disappear when NAT or VLAN config of this interface is updated
Ticket #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".
3030 - Disable LDAP Server When Erasing Flash Directory
Ticket #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.
3034 - Gateway: #11 could not be dialed on analog interfaces with feature codes enabled
Ticket #86819. This is a featiure code used on DECT systems and it was not disabled on analog interfaces
3165 - Gateway: Hanging calls in state "Clearing"
Ticket #88527. Hanging calls in state "Clearing". Caused by failed call replacement.
3162 - Gateway: Make interop flag "Ack incoming call" configurable on UI
Ticket #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
3116 - Gateway: MOH from call on hold in CONF interface calls
Ticket #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.
3036 - Gateway: Potential trap when recording is configured and a transfer happens
Ticket #86837. The problem is with a transfer, which is executed in the Gateway, not the PBX
3101 - Gateway: Routing on blind transfer call starts from wrong IF (undo of #86386)
Ticket #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.
3097 - H.323: Better error handling in case of incompatible SRTP parameters
Ticket #87624. Was needed for DECT OEM.
3098 - H.323: Media Negotiation did not work for Hold/Retrieve after a DECT conference
Ticket #87635. No media after retrieve
3156 - H.323: Potential Max-Busy-Ticks restart
Ticket #88441. The H.323 state machine could enter a endless loop
3090 - H.323: Problem sending real big signaling messages
Ticket #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.
3151 - H.323: Problem with Media Re-Negotiation on a DECT handover call
Ticket #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
3106 - H.323: Q.931 User Info could not be sent in incoming call proceeding state
Ticket #87740. This is used by the Fax interface
3072 - HTTP: Chunked transfer fails if the chunk header is not in a single packet
Ticket #87292.
3135 - HTTP: Chunked transfer sporadic fails with webdav
Ticket #88078.
3050 - IP TOS value is now traced with Wireshark
Ticket #87025. This value is now correctly traced within IP4 UDP/TCP traces.
3059 - IP-DECT: DECT security (4)
Ticket #87157. Some changes in preparation for the new feature "DECT security". This feature will be available later (IP1202).
3086 - IP-DECT: Disturbances GUI info change
Ticket #87522. The text of the GUI page disturbances is changed.
3131 - IP-DECT: Fault logging for Master module
Ticket #88005. Call and channel fault logging for the IP-DECT Master module are added.
3167 - IP-DECT: Logging for handover calls
Ticket #88536. Logging events for IP-DECT handover calls added.
3070 - IP-DECT: Login feature with disabled local R-key handling
Ticket #87269. The login feature is fixed now if the local R-key handling is disabled.
3103 - IP-DECT: No idle display update with own message
Ticket #87715. A message with the same sender and receiver name overwrites the handset's idle display. This feature is removed now.
3157 - IP-DECT: No media after conference and toggled to held call
Ticket #88442. If the DECT user leaves the conference mode and toggles to an held call, there is no voice. This is fixed now.
3083 - IP-DECT: No voice with early handover
Ticket #87504. This fixes outgoing calls without voice if early handover is done before the media channel is initialized.
3166 - IP-DECT: Potential buffer overrun
Ticket #88535. Buffer overrun check added in dectusers module for command 'show'.
3080 - IP-DECT: Remote hold in conference mode
Ticket #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.
3150 - IP-DECT: SIP/Intop - rejected call transfer by target
Ticket #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.
3105 - IP-DECT: Trap with rejected handovers
Ticket #87730. A trap occurs in the rare situation if the handover is rejected by the radio.
3152 - IP-DECT: Unused OEM modules removed
Ticket #88413. Unused OEM modules has been removed now.
3088 - IP-DECT: Web UI info page, version and release state
Ticket #87538. Now the version and the release state are shown on the web UI info page of the IP1202.
3126 - IP-DECT: Wrong radio list after MAC-alias change
Ticket #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.
3068 - IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: DSP Update to Version 680.05
Ticket #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.
3031 - IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: fax bad signal quality events are sent on good fax connections
Ticket #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)
Ticket #87878. receiver volume adapted to changed equalizer settings from fix #80403
3160 - IP232,IP222,IP241: Reduce memory requirements of display rendering
Ticket #88485. Reduce memory requirements of display rendering.
3117 - IP241,IP222,IP232: Presence note may is not enough truncated on 'presence' fkey
Ticket #87825. Presence note may is not enough truncated on 'presence' fkey. Esp. when there's also an icon to be displayed.
3138 - IP6000 IP2000: Webinterface for ETH1 Link setting didnt work
Ticket #88137. ..
3044 - IPVA: Query Physical Link Status.
Ticket #86937. -The physical link status wasn't evaluated so far. Now it is. -Added statistics page 'Interfaces/EthX/Statistics'
3041 - Linux: IP0010 available again
Ticket #86903. With V9 hotfix 16 Linux can not be started on the IP0010. This is fixed now.
3110 - Linux: Trap of IP810 if Linux is started
Ticket #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.
2895 - make update script parsing more tolerant to suspicious line ends
Ticket #84349. Some WEBDAV tools garble line end when a text file is stored after editing. Last seen \r\r\n instead of \r\n. Any sequence consisting only of \r \n chars should be read as one line end because empty lines have no meaning in an update script.
3032 - Mis-typing on Maintenance/Diagnostics/Counters
Ticket #86812. Show "kbit/s" instead of "kBbit/s".
3133 - OS: Quota mechanism did not work for CPU time consumed by timer handling
Ticket #88034. This could result in a MAX_BUSY_TICK restart if many timer expired at the same time
3112 - PBX administration: Viewer could change phone configuration of users
Ticket #87774.
3144 - PBX Boolean: Unpredictable behaviour if more the 16 times are entered
Ticket #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.
3048 - PBX Conference: Calling Party was missing in the CDRs for calls to conference members
Ticket #86977. Only the conference object itself was present as forwarding party
3174 - PBX Conference: No Media in case media offer only received with alert/connect from called member
Ticket #88590. This could happen if a broadcast object was called by a conference
3033 - PBX Mobility: Take precaution against hanging calls because of misbehaving mobile phone or client
Ticket #86813. Use a 20s timer to terminate any call to the mobility object if no internal call is initiated
3123 - PBX Trunk: Diverting as Calling Feature should replace the name as well
Ticket #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
3118 - PBX Trunk: If a call was forked to a trunk with no-name option, name was removed from the original call as well
Ticket #87843. A global flag was set on the incoming call, which was wrong.
3055 - PBX Trunk: New destination for rejected calls
Ticket #87151. Calls which are rejected (busy after alert), can be redirected to a configurable destination.
3046 - PBX Waiting: CFNR with number filter did not work
Ticket #86959. The CFNR was not executed
3102 - PBX Waiting: DTMF overlap dialing or blind transfer to same Waiting object was rejected with busy
Ticket #87681. Even if this was caused by a CFB or CFU on the dialed destination
3100 - PBX Waiting: Missing ringback on call forward after announcement
Ticket #87674. This was a collateral damage of
fix: #81370: PBX Waiting: Call state shows "Disconnecting" after switch from announcement 1 to announcement 2
3154 - PBX Waiting: User Information Message from announcement interface accidentally forwarded to caller
Ticket #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.
3092 - PBX-SOAP: UserCall on WQ did not take "Send Number" into account and call was not marked as internal
Ticket #87559. Calls should be send with "Send Number" as source if configured
3132 - PBX-SOAP: UserCall with 'cn' as destination to a user in different node did not work
Ticket #88008. The call is done using the number, but the number needs to be adjusted according to the nodes
3149 - PBX: Allow configuration of default presence/dialog-info visibility for group members
Ticket #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.
3089 - PBX: Blind transfer to WQ by IP-DECT caused hanging calls
Ticket #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
3161 - PBX: Call Completion monitoring call should be terminated only after the CC ringout call is alerting
Ticket #88489. This is needed for QSIG interoperability
3054 - PBX: Call, which was rejected with busy, because of CF loop, could not be forwarded by Trunk Busy destination
Ticket #87150. Such a call was always disconnected
3095 - PBX: Called number was missing in CDRs for calls to busy endpoint
Ticket #87620. In case there was no other event then rel-to/from and the call was dialed with overlap sending
3093 - PBX: CFB configured at PBX object did not patch number correctly
Ticket #87581. In case the called endpoint was in different node then PBX object, the number did not contain all the needed prefixes
3111 - PBX: CFNR to same node but different PBX failed on Slave
Ticket #87772. The PBX routing did not work correctly in this case
3073 - PBX: Conference and hanging VM script
Ticket #87312. Wrong media initializing in the BC Conference object causes a hanging VM script if a file should be played. This is fixed now.
3099 - PBX: Conference trap
Ticket #87662. A trap in the BC conference PBX object is fixed.
3040 - PBX: Name beginning with '*' caused problems with SOAP applications
Ticket #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.
3119 - PBX: No CDR was generated for enblock call, which was rejected because of filter config
Ticket #87869. A CDR with cause 52 (outgoing call barred) is generated
3051 - PBX: Replication from a dyn PBX was not possible
Ticket #87065. Configuration of dyn PBX id now possible for replication
3038 - PBX: Trap if a Hold was attempted for a call without media
Ticket #86874. Could be caused by a misbehaving application or voip device
3037 - PBX: XML Export/Import did not work, if DTMF Feature objects present
Ticket #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.
3079 - phone: "Spare" function key to reserve key positions for administrative purposes
Ticket #87406. This new key permits to reserve key positions for later definition by administrator.
3078 - phone: Modification of function key on phone can be disabled for any single key
Ticket #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.
3045 - phone: call diversion override via indirect dialing could not be disabled
Ticket #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
3114 - phone: Function key options which need to be enabled in Phone/Userx/Preferences could be edited on phone although disabled
Ticket #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
Ticket #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.
3084 - phone: ip222, ip232: multiple reports indicating offhook were misinterpreted and could drop a call
Ticket #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).
3125 - Phones: Allow lcd_dump.bmp to be retrieved with viewer credentials
Ticket #87945. Allow lcd_dump.bmp to be retrieved with viewer credentials.
3139 - SIP: Decoding problem on application/simple-message-summary
Ticket #88188. application/simple-message-summary may get decoded wrongly.
3142 - SIP: DNS priority value not honored always
Ticket #88248. DNS priority value not honored, if SRV query returns names without address in additional records.
3029 - SIP: Do not interwork holdNotific and retrieveNotific while on hold
Ticket #86736. Interworking problem with VOXTRON/XCAPI Do not interwork holdNotific and retrieveNotific after remoteHold.
3143 - SIP: Do not take "9564+4631559300" as E.164 number
Ticket #88254. Do not take "9564+4631559300" as E.164 number.
3076 - SIP: Do registration refresh more ahead of expiration
Ticket #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
3113 - SIP: Don't try to dns-resolve IP adresses
Ticket #87781. Don't try to dns-resolve IP adresses.
3115 - SIP: Error handling huge INVITE requests
Ticket #87811. Failed to compose provisional and final response. Lots of error messages in trace and truncated responses.
3108 - SIP: Parsing of multiple Allow lines was not implemented
Ticket #87753. Now all Allow lines are parsed (not only first one).
3049 - SIP: Rare trap when cancelling call
Ticket #86994. When call abort interferes with re-routing in gateway application.
3094 - SIP: Trap when logging out dect user
Ticket #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
3109 - syslog packets were sent with the initially assigned source address even after a new address had been assigned
Ticket #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.
3052 - TLS: Ignore incoming HelloRequest messages
Ticket #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.
3163 - Viewer was able to download config with standard password
Ticket #88503. This was a security hole
3104 - Voicemail: <pbx-fwd>, Forward Timer Didn't Fire For Connected Calls
Ticket #87717. Voicemail: <pbx-fwd>, Forward Timer Didn't Fire For Connected Calls <!- pbx_vm.cpp -->
3071 - Webdav: DELETE may fail if file has been opened for reading before
Ticket #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.
3081 - Webdav: Write information into trace if DELETE fails because file is in open state
Ticket #87488. Write information about Webdav session holding open file handle.
V9 Hotfix 18 (9061158)
3177 - Alarm and Event forwarding to a SYSLOG server
Ticket #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".
3175 - Announcement Calls with DTMF feature code object
Ticket #88643. A new DTMF feature code allows to make announcement calls to a dialed number.
3213 - Gateway: Missed FAX pages received with the FAX interface
Ticket #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.
3186 - IP22 IP24 IP28 IP305: Sometimes the DSP stops after sending CLIP
Ticket #89130. ..
3140 - IPVA: Enhancement for innovaphone testbed: Indicate whether Upload to DRAM took place
Ticket #88223. Required for innovaphones' automated testbed
3181 - Kerberos: Admin UI trap when having too many Kerberos hosts
Ticket #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.
3187 - PBX: Configuration UI broken for checkmarks on PBX/Config/General and PBX/Config/myPBX
Ticket #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
3188 - phone: ip222,ip232: the destination of an acticve diversion was not saved when the diversion was deactivated at phone
Ticket #89177. The destination (name or number) disappered when the diversion was deactivated. It was also not visible via the WEB interface.
3183 - SIP: Provide display name in 200/OK for SUBSCRIBE
Ticket #88764. Add display name to To header in 200/OK for SUBSCRIBE.
3179 - TLS: Do not ignore early SOCKET_RECV
Ticket #88668. Allow SOCKET_RECV from application between SOCKET_CONNECT and SOCKET_CONNECT_COMPLETE. This event flow is used by the HTTP client.
V9 Hotfix 19 (9061180)
3234 - 'power-off loop' relay switching function failure
Ticket #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.
3199 - AD Replication: A Buffer for Processing The Paged Result Cookie Was Too Small
Ticket #89385. 1KB wasn't enough, now 2KB.
3222 - AD Replication: Merge v10 code into v9 code
Ticket #89671. Some recent refactoring works weren't in sync. A state machine didn't reach "Completed".
3198 - case independence for the characters of the Basic Russian Alphabet added
Ticket #89367. mappings added
3230 - Fix for MIPS counter
Ticket #89804. MIPS counter was incorrect on IP1201 and IP4001
3207 - H.323: Automatically connect signaling TCP if NAT router is detected
Ticket #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.
3196 - H.323: RTP-DTMF did not work on exclusive coder/media relay configurations
Ticket #89328. Problem for DTMF on SIP trunks
3229 - IP-DECT: Trap with call transfer
Ticket #89786. A trap occurs if a call transfer is received in the IP-DECT radio module. This is fixed now.
3236 - IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: DSP Update to Version 680.07
Ticket #89956. Fixed modem bypass with slow modems
3206 - IP22 IP24 IP28 IP305: DSP debug code added
Ticket #89493. ..
3227 - IP22 IP24 IP28 IP305: Sometimes the DSP stops after sending CLIP (2)
Ticket #89760. ..
3192 - IP232,IP222,IP241: Config option to adjust LCD brightness in idle state
Ticket #89261. -> Main Menu -> Phone Setup -> LCD light (idle state) Can be tuned down to zero.
3200 - LDAP Expert: "Next"-Browsing through DB failed
Ticket #89391. URI encoding error
3205 - log message forwarding to another innovaphone device did not work since V9hotfix15 (on the receiving device)
Ticket #89492. this problem is located on the reciving device, it does not depend on the the version of the forwarding device.
3221 - Media: Redirecting SRTP streams for NAT clients only after successful SRTP authentication
Ticket #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.
3235 - myPBX: Default group visibility was not displayed correctly
Ticket #89954. The default group visibility can be configured from v9hotfix17. Regardless of that configuration myPBX showed full visibility in the visibilty settings.
3215 - OS: Potential trap when starting a timer, which is already deleted
Ticket #89628. Could result in a strange trap with PBX transfer/recall
3217 - PBX Admin UI: Reconfiguring user visibility (Access) did not work correctly
Ticket #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.
3208 - PBX Mobility: Conference ID not set for outgoing calls initiated by myPBX
Ticket #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
3164 - PBX-SOAP: Present normalized number of peer also
Ticket #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"
3204 - phone: an intrusion call set up via Partner function key could not be cleared at the intruding phone via TAPI
Ticket #89443. The TAPI interpreted the recording state as a conference
3216 - phone: while recording was running for a call this call ould not be cleared via SOAP/TAPI
Ticket #89629. The TAPI assumed the intruding phone to be in a conference
3224 - phone: if a call ringing while the handset was offhook was accepted via SOAP/TAPI the call was not cleared when going onhook
Ticket #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.
3201 - phone: ip222, ip232,ip241: the notification tone indicating a new message when a call is active was sent to remote
Ticket #89392. instead to notify the receiver of the message the remot party did hear the tone
3203 - phone: ip222,ip232: humming noise in USB headset speaker in outbound call setup phase, disappears once connected
Ticket #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.
3226 - phone_orchid: pressing speaker key in handset/headset mode switches to handsfree mode, pressing again returns to previous mode
Ticket #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.
3219 - Possible Trap While Getting Post-Mortem Log
Ticket #89652. Possible Trap While Getting Post-Mortem Log
3210 - PRI-QSIG: Interop config for channel numbering
Ticket #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.
3158 - SIP/TLS: Rejecting server certificate
Ticket #88444. Validating server certificate against configured domain name. Must be validated against configured proxy domsina name.
3202 - SIP: "Supported: timer" missing in UPDATE message
Ticket #89429. "Supported: timer" missing in UPDATE message.
3228 - SIP: Bug in handling of INVITE with Replaces
Ticket #89777. Bug in handling of INVITE with Replaces. Results in hanging call.
3212 - SIP: Don't tell application that registration is down when handling redirect response
Ticket #89582. Don't tell application that registration is down when handling redirect response for REGISTER.
3231 - SIP: Media negotiation fails on calls into Waiting Queue
Ticket #89838. Media negotiation fails on calls into Waiting Queue if caller put call on hold before WQ agent accepts the call.
3214 - SIP: No T.38 parameter when indicating capabilitity only
Ticket #89617. No T.38 parameter when indicating capabilitity only.
3193 - SIP: Wrong branch value in Via header in ACK request
Ticket #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.
3197 - Voicemail: <pbx-getcallinfo out-confid="...">, pass conference guid into a script
Ticket #89332. In order to allow correlation of CDRs to voicemail-recorded files
3218 - Voicemail: Trap
Ticket #89648. If Trace-Checkmark activated, a diverted call carrying an mwi interrogate was processed
V9 Hotfix 20 (9061198)
3259 - 802.1x: Version 3 Frames Weren't Processed
Ticket #90485. Now they're read initially. Handshakes remains to be in version 1.
3252 - Gateway: Support of a header line for FAX documents
Ticket #90374. Support of a header line for FAX documents is added in the FAX interface.
3243 - H.323: Allow media offers with 0.0.0.0 as address
Ticket #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.
3313 - H.323: Allow update of Registration password
Ticket #91440. needed in case a password is changed, which is replicated to a DECT system
3291 - H.323: Incoming faststart call was sometimes not accepted as faststart
Ticket #91136. Only happens if non EFC is used on the incoming call, so this only happens in interop cases with other H.323 equipment
3287 - H.323: No event should be generated in State 11 and 25
Ticket #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.
3276 - H.323: Potential Trap in special case which could only happen in version 10
Ticket #90768. This fixed is merged to version 9 only of consistency reasons
3315 - H.323: PROGRESS in connected state was treated as 'unexpected'
Ticket #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.
3249 - H.323: Support for registration from a private network thru NAT
Ticket #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
3273 - H.323: Timer to monitor response to setup too short for some traffic cases
Ticket #90696. This created unnecessary event in IP-DECT systems when calling powered off or out of range handsets
3257 - IP-DECT: Busy state on maximum call count
Ticket #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.
3314 - IP-DECT: Configuration option 'Registration with system password'
Ticket #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.
3242 - IP-DECT: Don't show DTMF in radio call list
Ticket #90034. User dialled digits during calls which are sent as DTMF should not be shown in the radio call list. This is fixed now.
3264 - IP-DECT: RTP stream
Ticket #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.
3302 - IP-DECT: Trap in Radio module (IP1202)
Ticket #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.
3311 - IP0010 IP1060 IP3010 IP6010 IP22 IP24 IP28 IP302 IP305: switch from modem bypass to voice did not work
Ticket #91429. In some cases announcments with music trigger modembypass, in this case at least the fallback to voice should work.
3247 - IP2000: Prevent blinking error LED IP2000
Ticket #90102. The firmware tried to load the conference DSP, which is not available on the IP2000
3239 - IP232,IP222,IP241: Truncate directory entry information
Ticket #90027. Truncate directory entry information to keep from overlapping with number type indication.
3306 - Ldap Replication from NDS
Ticket #91347. Skip isDeleted attribute with content others than 'true'. Occurred when replicating from an NDS running in AD compatibility mode.
3010 - Linux: Memory allocation changed for IP810
Ticket #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.
3285 - PBX CSV Import: Corrupted objects at buffer boundaries
Ticket #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
3281 - PBX Trunk: Name to Number Feature did not work with calls to extern
Ticket #90858. If an endpoint cannot be found by name, the call should be forwarded to extern
3275 - PBX Waiting: Potential Trap if editing while a call is initiated with SOAP
Ticket #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
3263 - PBX-SOAP: Support for 'rc' and 'srce164' on UserCall on Waiting Queue
Ticket #90537. If a Waiting Queue is used for outgoing calls, these features can be usefull for some applications
3244 - PBX: Busy On ... Calls at PBX objects did not take into acccount that a call may be routed back to Slave
Ticket #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.
3254 - PBX: CC Requests were sent with wrong number if a SendNumber was configured at the user
Ticket #90432. Usually this is no problem only when interworking with some QSIG PBX's this causes the call-completion to fail.
3260 - PBX: Forking a call to a Trunk with "Outgoing call restricted", causes the original call to be restricted as well
Ticket #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.
3272 - PBX: New configuration option 'Hide connected Number' at object
Ticket #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.
3309 - PBX: Not possible to login as user with non-full admin rights if 'Password protect all Pages'
Ticket #91414. Some pages needed for the UI, which are normally not password protected, could not accessed with the reduced rights
3258 - PBX: Support for Opticaller Data Callthru did not work
Ticket #90480. Support for Opticaller data callthru was added, but did not work
3240 - PBX: Support of long user-user-informations by SOAP
Ticket #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.
3237 - phone: new "Do Not Disturb" action "ring once"
Ticket #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.
3286 - Phone: Trap when selecting registration for a directory entry
Ticket #90999. Trap when selecting registration for a directory entry, but only if the registration has either no name or no number.
3255 - phone_orchid: dialtone missing when recording is active and the active call is held to open a consultation call
Ticket #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.
3256 - Possible trap when doing a leak check
Ticket #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.
3241 - Relay: Support of long user-user-informations by FAX
Ticket #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.
3209 - SIP: Auto answer with SDP in ACK
Ticket #89539. Auto answer results into no-audio when INVITE comes w/o SDP offer.
3268 - SIP: Disabled IP-DECT interface tries to register
Ticket #90597. Disabled IP-DECT interface tries to register.
3211 - SIP: Fix for overlap dialing with KPML
Ticket #89581. Dialing digits entered before KPML subscription is established need to be queued until KPML subscription is established.
3310 - SIP: Follow offers ptime proposal
Ticket #91421. Better follow offers ptime proposal. Otherwise SAMwin operator does not stop sending re-INVITE.
3289 - SIP: Heavy TLS retry load when server certificate was rejected
Ticket #91033. Collateral damage from earlier fixed for DNS refreshing.
3261 - SIP: Locally configured DNS entries were not used if no DNS server configured
Ticket #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.
3298 - SIP: Media negotiation for video fails if called through waiting queue or multi reg
Ticket #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.
3248 - SIP: Memory leak when receiving NOTIFY(message-summary)
Ticket #90106. Memory leak when receiving NOTIFY(message-summary)
3278 - SIP: Possible buffer overrun
Ticket #90780. Fix for possible buffer overrun.
3238 - SIP: Re-try INVITE after 407 even if no password configured
Ticket #90024. Calculate Digest with zero-length password and re-try INVITE.
3277 - SIP: Respect changes in PAI/PPI header when receiving UPDATE with SDP offer
Ticket #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.
3308 - SIP: SDP answer for T.38 switch-over must contain multiple media descriptions
Ticket #91377. ... if SDP offer contains multiple media descriptions.
3299 - SIP: Secondary target (hostname) is not resolved
Ticket #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.
3250 - SIP: SUBSCRIBE using old IP address in Contact field
Ticket #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.
3253 - SIP: Support for MESSAGE inside voice call
Ticket #90408. Support for text messages inside voice call.
3265 - SIP: Trap on IP-DECT
Ticket #90569. Trap in GK-CHANNEL when Dectmaster application sends DTMF before call if created.
3288 - SIP: Trap when cancelling call
Ticket #91023. Double delete of a call entity.
3307 - SIP: Wrong call was disconnected after successful transfer
Ticket #91349. Wrong call was disconnected after successful transfer.
3280 - SIP: Wrong error log "Timeout during media negotiation for call"
Ticket #90821. Error log "Timeout during media negotiation for call" may occur after re-negotioation. Re-negotioation occurs during hold/retrieve/transfer.
3295 - SNMP Get-Next Requests Carrying an Octet-String Value Caused Memory Leak
Ticket #91215.
3251 - Voicemail: Memory Load High With Repeated Calls To <store-getnext>
Ticket #90362. The use-case does not happen within the standard voicemail, but within e.g. innovaphone conferencing.
3246 - Voicemail: Send silence RTP during recording
Ticket #90095. Send silence RTP during recording for some SIP carriers that do not send RTP without receiving RTP.
3318 - Voicemail: URL-En-/Decoding
Ticket #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"/>
V9 Hotfix 21 (9061222)
3422 - 802.1x (EAPOL) did not work on interfaces configured for VLAN
Ticket #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
3411 - Click sounds at caller side when calling another port of same gateway
Ticket #95436. occured since V9hotfix5
3377 - Gateway: CGPN-Maps executed even if the Route did not match in case of enbloc calls
Ticket #93400. For enbloc calls the CGPN of routes were executed even if the dialed number was incomplete.
Ticket #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.
Ticket #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.
3385 - Gateway: Header line of FAX documents with big endian devices
Ticket #94824. The header line of FAX documents with big endian devices was not correctly printed. This is fixed now.
3356 - Gateway: Transmitting FAX documents to receiver with polling mode
Ticket #92388. It is not possible to send fax documents to a receiver with polling capability. This is fixed now.
3390 - H.323: No media after blind transfer to Waiting Queue on other PBX, when operator connects
Ticket #94961. Happened when different coders where used on caller and called side.
3334 - H.323: Unnecessary re-initializing of rtp-channel on incoming calls to phone
Ticket #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.
3374 - IP-DECT/Relay: Blocked calls by hidden feature code *5/*7
Ticket #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.
3369 - IP-DECT: Call transfer with enbloc dailing
Ticket #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.
3348 - IP-DECT: Cipher key index request for security test devices
Ticket #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).
3357 - IP-DECT: Cipher key index table update function
Ticket #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.
3366 - IP-DECT: CSV user export
Ticket #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.
3345 - IP-DECT: Hold and Retrieve with SIP and SRTP could result in no media
Ticket #92189. Problem with forwarding changed SRTP Keys from master to radio
3342 - IP-DECT: Hold/Retrieve could result in no media for incoming SIP calls with SRTP
Ticket #92014. This was in fact a H.323 media negotiation problem between master and slave
3350 - IP-DECT: Potential trap when switching to an from 3pty conference
Ticket #92262. In case of a very unlikely collision of events, a message was sent to an already deleted object, which cause a restart
3327 - IP-DECT: Static ports between Master and Radio
Ticket #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.
3326 - IP-DECT: Trap with data calls
Ticket #91807. The IP-DECT Radio traps if a data call is released and the release includes a facility.
3337 - IP-DECT: Wrong XML data in Radio endpoint
Ticket #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.
3317 - IP0010,3010,6010,1060,810: ethernet link down not detected when a cable was unplugged after boot
Ticket #91600. The link state interrupt was triggered only once after boot, further link state changes were not indicated.
3368 - IP0010,3010,6010,1060,810: limit number of ethernet packets processed per receive interrupt
Ticket #93130. this prevents that too much time is spent in ethernet driver in case of broadcast storms or DOS attacks.
3335 - IP22,IP24,IP28,IP302,IP305: RTP-DTMF not offered when using a/b interface
Ticket #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.
3413 - IPv6: Disable checkmark on ethernet interfaces added
Ticket #95452. Allows to disable sending/receiving of IPv6 packets on this interface. This may be desireable for security reasons
3330 - Kerberos administration: Increase maximum number of Kerberos users from 20 to 50
Ticket #91847. Increase limit on page General/Kerberos. Use POST for submitting form, instead of GET.
3384 - LDAP Client: SearchRequest.derefAliases Changed To neverDerefAliases(0)
Ticket #94812. Was derefInSearching(1)
3321 - myPBX: Rejecting an incoming broadcast call disconnected the call for all alerting endpoints
Ticket #91719. The call sould continue alerting on the other endpoints.
3412 - PBX Broadcast: Memory leak when calling busy broadcast object with round-robin config
Ticket #95439. If all destinations of a broadcast object are busy a name-id facility generates a memory leak.
3400 - PBX Exec: Secretary availability monitoring did not work with multiple scretaries, with names starting identically
Ticket #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.
3340 - PBX Mobility: Call from mobile endpoint to a user with mobility, but without PBX phone was disconnected
Ticket #91976. This was a collateral damage of
fix: #86813: PBX Mobility: Take precaution against hanging calls because of misbehaving mobile phone or client
3274 - PBX Mobility: Connected number from mobile phone was forwarded to caller
Ticket #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.
3361 - PBX Trunk: Flag to block presence/dialog-info subscriptions
Ticket #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.
3394 - PBX Waiting: Presence set for operator was not cleared, on delete or editing of Waiting Queue object
Ticket #95016. When configuration of the Waitinng object is now changed, any presence set by the Waiting object is cleared.
3332 - PBX-SOAP: Potential Trap with UserClear
Ticket #91886. A SOAP applicatoin (e.g. TAPI) uses the method UserClear to clear a call. This could cause a trap on some platforms when doing this for a mobile endpoint.
3319 - PBX: Allow Name (instead of Long Name) to identify user for mobility data call thru
Ticket #91660. This should simplify Opticaller configuration
3353 - PBX: Call Completion to a user without registration, but forking, caused immediate Callback
Ticket #92341. Instead to call completion request should be rejected in this case
3381 - PBX: Hide connected Endpoint did nor work for forward to other PBX or Gateway object
Ticket #93513. Diverting Leg Information facilities or Name Identification facilities were still forwarded in this case.
3376 - PBX: Master Slave license update period 10s instead of 10min
Ticket #93330. For test purposes the period was reduced to 10s, but by accident this change ended up in version 9 as well
3336 - PBX: MOH URL parameters for parked calls should refer to the parked-to and not parking user
Ticket #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.
3379 - PBX: Port License counting wrong, when moving users
Ticket #93477. When moving a user from a master to a slave, the license count on the master was reduced only after reboot.
3365 - PBX: Trap if user object is deleted, which is used by other applications (e.g. myPBX)
Ticket #92985. The applications need to cleanup in this case
3380 - PBX: XML error on User Interface if slave name with non-ascii characters used and registration was redirected from such a slave
Ticket #93483. Conversion of UCS2 as received from the registration to UTF-8 on Web UI was wrong
3322 - phone: ip222, ip232: reset/restart USB headset when a hang condition or a port disconnect condition is detected
Ticket #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
3323 - phone: ip222, ip232: Some USB headsets were not detected after a soft reset
Ticket #91734. This was observed with Jabra BIZ 2400 USB at IP222/232 with hardware build 800 and newer.
3373 - phone: ip222,ip232: sporadic boot time trap when a Jabra LINK 280 adapter is plugged
Ticket #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.
3346 - phone: ip241: do headset volume control in digital domain
Ticket #92203.
3355 - PHONE_SIG_MODE_KEEP_NUMBER_TYPE can be set at phonesig startup to keep type of number in all q931 numbers
Ticket #92385.
3284 - Potential Trap when rapidly switching local Media connections (Conferencing)
Ticket #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
3341 - SIP/DNS: Wrong port used on secondary SIP server
Ticket #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.
3396 - SIP/UDP: Sending response to wrong address and port
Ticket #95065. Sending response to wrong address and port. But only if Via header of incoming request contains domain name.
3344 - SIP/WLAN: Keep local Contact-URI up-to-date on subscriptions
Ticket #92134. Keep local Contact-URI up-to-date on subscriptions (e.g. message summary) when local ip address changes at runtime.
3293 - SIP: Better handling of incoming calls
Ticket #91153. Msg sequence INVITE,CANCEL,INVITE may result into second call rejected.
3372 - SIP: Debug information for problems with STUN
Ticket #93233. Added debug output to trace problems with STUN.
3391 - SIP: Display names need escaping of " and \ according to RFC-3261
Ticket #94966. Acc. to RFC-3261 characters " and \ (%x22 and %x5C) are to be escaped as "quoted-pair".
Ticket #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.
3420 - SIP: Interworking issue with "LifeSize Passport/LS_PP1_4.11.9 (8)"
Ticket #95785. Problems decoding large and complex SDP offer from LifeSize.
3389 - SIP: New interop tweak "No Remote Hold Signaling"
Ticket #94954. New config option on gateway interfaces "No Remote Hold Signaling". Disables interworking of "inactive" into remoteHold.
3386 - SIP: Port mapping with STUN failed since DNS resolvin of STUN server failed
Ticket #94876. Port mapping with STUN failed since DNS resolvin of STUN server failed. Wrong STUN server port was used (5060 instead of 3478).
3371 - SNMP Walk udpTable, tcpConnTable Could fail
Ticket #93225. The table index returned wasn't always lexically ascending, causing a walk to stop.
3333 - SRTP: Avoid one-way media with high start sequence numbers
Ticket #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.
3370 - SRTP: Remove traces when packet authentication failed
Ticket #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.
Ticket #91874. Passing name info of divertingLegInformation2 facility into the script.
3359 - Voicemail: Emailing file names with '#'-characters failed
Ticket #92757. An Escape mechanism wasn't applied
3367 - Voicemail: Url Parameter To Control Number Processing
Ticket #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
3362 - Webmedia: New URL parameter "fallback=true"
Ticket #92957. New URL parameter "fallback=true". If given and the specified media file cannot be retrieved, webmedia starts playing builtin MOH.
V9 Hotfix 22 (9061240)
3450 - Gateway: CGPN-Maps executed even if the Routing was already completed
Ticket #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
3448 - Gateway: FAX interface on IPVA
Ticket #96660. The FAX interface on IPVA can not connect to a remote device because of wrong protocol events. This is fixed now.
3452 - H.323: One-way-voice if SRTP call to a Waiting queue is forwarded via Waiting Queue Maps to a phone
Ticket #96721. This is a problem with SRTP key exchange, which could happen in other traffic scenarios as well.
3427 - Httpclient: Problems with HTTPS URLs
Ticket #96099. Shortcut to local file I/O did not work fot HTTPS URLs.
3453 - IP-DECT: OEM PBX type info in GUI data
Ticket #96723. Now the GUI data includes the type info of an OEM PBX.
3440 - IP-DECT: Physical location was wrong after logout/login on handset
Ticket #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.
3429 - IP-DECT: Possible no media in case of media renegotiation after handover
Ticket #96124. The problem happened in about 50% of the cases of a media renegotiation which results in a different coder after handover.
3430 - IP-DECT: Show release state in IP1202
Ticket #96157. The release state is not shown in the IP1202. This is fixed now.
3454 - IP-DECT: Web UI administrator user list removed on IP1202
Ticket #96726. The Web UI administrator user list is removed on the IP1202 now.
3436 - IP-DECT: Wrong GK id of standby Master to Mobility Master
Ticket #96302. The standby Master uses a wrong gatekeeper id to register to the Mobility Master. This is fixed now.
Ticket #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.
3455 - IP22 IP24 Ip28 IP302: Don't complete media negotiation for ab-interfaces if no media can be sent
Ticket #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.
3421 - IP222 IP232 IP241: LCD display is after softreset sometimes out of sync
Ticket #95866. Displaycontroller needs to be stopped before restart.
3456 - IP28: Click sounds at caller side when calling ip28 gateway
Ticket #96782.
3458 - IP28: Ring voltage failure (RING_FAIL)
Ticket #96826.
3457 - IP28: Sometime Ringing stopped working on an anlog port. Worked again only after reset.
Ticket #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.
3446 - License download not working
Ticket #96622. License download stops. Licenses are not downloaded.
3414 - PBX Executive: Allow monitoring of availability of secondary secretary, don't treat Exec as secretary
Ticket #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.
3432 - PBX Trunk: "Outgoing Calls restricted" did not work correctly, Presentation restricted was set, but number could be wrong
Ticket #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.
3441 - PBX-CDRs: Conference ID missing in CDRs created by Mobility data callback/callthru
Ticket #96399. This caused these calls not to show up in the reporting
3447 - PBX: Changing of PBX Object Name did not change the name to be used for registration
Ticket #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.
3155 - PBX: New option for RTP Proxy - proxy only if different registration address
Ticket #88439. To avaoid RTP Proxy for two endpoints located within same private network behind NAT
3439 - PBX: pbx_makecall.txt not only for mobility, but for Waiting Queue also
Ticket #96384. Allows to initiate call with simple web request from a Waiting Queue
3438 - PBX: Send forking calls as diverted calls
Ticket #96370. So that on the called side, it will be displayed who forked the call the same way as a diverted call is displayed.
3443 - PBX: Unexpected behaviour if too many filter were configured
Ticket #96416. No new filters were accepted without error message
3194 - PBX: URI dialing, should not be case sensitive and numbers should be possible
Ticket #89326. needed for federation
3428 - phone: do not report "No Media Data received" errors for connections to a recording device
Ticket #96102. some recoding devices, for example ASC never send data on a recording connection
3442 - phone: if a number to be dialled contains a comma, the digits following the comma are sent as DTMF tones after connect
Ticket #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.
3460 - phone: ip222,ip232,ip241: Local Network Coder default for User-2..6 was G711 instead of G722 as for User-1
Ticket #96890. WEB-Interface "Phone/User-2..6/General/Options/Local Network Coder" was preset to G711 instead to G722 as for User-1.
3434 - phone: ip222,ip232: inbound calls automatically connected to Plantronics Savi W440/740/745 headsets with new firmware Versions
Ticket #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.
3431 - SIP: Must not answer "refresher=uac" if request contains "refresher=uas"
Ticket #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
3488 - SIP: New interop tweak /register-interval
Ticket #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.
3437 - SIP: Workaround for buggy registrar
Ticket #96313. Workaround for buggy registrar. Different expirtes values in Contact header and Expires header.
3449 - SRTP: One way audio after some minutes on IP6000 IP2000 IP6010 IP0010
Ticket #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.
V9 Hotfix 23 (9061252)
3466 - ASN.1 BER: Decoding of Sequence Member with indefinite length failed
Ticket #97072. An Avaya supplementary service couldn't be decoded
3471 - Gateway: FAX interface on IP800/IP305/IP302
Ticket #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.
3467 - H.323: Channel Close sometimes not sent on hold
Ticket #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)
3475 - H.323: Fast Unregister/Register operations could lead to failed registrations, in case of fixed signaling ports
Ticket #97637. A listening socket could still be in use. Only happened with IP-DECT and multi-master.
3463 - H.323: Offered packetization should be honored for SIP interoperability
Ticket #96983. Problem happend with calls from Samwin CBC
3487 - H.323: Potential Trap when reconfiguring an H.323 registration
Ticket #97820. This is a collateral damage from
91815: IP-DECT: Static ports between Master and Radio
3476 - H.323: Trap when Name-Id of more the 128 characters is to be forwarded
3468 - IP4 did not work anymore when IP6 was disabled via WEB interface
Ticket #97111. When the "IP6/ETNx/IP6/Options/Disabled" was checked the IP4 operations did stop after a while.
3473 - ISDN: Send legacy Redirecting Number for 'old' Fax Servers
Ticket #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.
3489 - PBX Executive: Calls with calling id restriction and without calling id, were sent to executive, even if secretary available
Ticket #97897. This happend for calls coming in from public ISDN with calling id presentation restriction thru a trunk object without number
3465 - PBX Waiting: timeout argument for pbx_makecall.txt URL
Ticket #97010. A timeout argument was added to the PBX0/ADMIN/pbx_makecall.txt URL to cancel an outgoing call initiated with this URL.
3481 - PBX-CDRs: Better CDR for pickup
Ticket #97681. A pickup was not indicated in the CDRs
3485 - PBX-SOAP: Status of Boolean object indicated as local number
Ticket #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)
3486 - PBX-SOAP: UserPark allows to park to another object
Ticket #97741. The argument 'cn' was ignored in v9 and earlier, now it can be used to identify a destination for the park.
3477 - PBX: Name-Id of busy destination was not forwarded to other PBX
Ticket #97646. Display was different when calling a busy phone on local PBX or on another PBX
3494 - PBX: Potential trap when receiving unknown presence activity
Ticket #98043. In the respective version unknown activities are mapped to "busy"
3479 - phone: "Prepare Override" function key did not work since V9hotfix21, the overriding source address was ignored
Ticket #97665.
3470 - phone: DTMF digits following a comma in a number to be dialed were not handled correctly in some cases
Ticket #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
3469 - phone: ip222,ip232: Plantronics Savi W440 dosn't report Talk-Key events in a call established at phone or by a CTI application
Ticket #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.
3462 - phone: ip222,ip232: USB headset echo effects in call setup phase when a G722 call is started by a CTI application
Ticket #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.
3484 - phone: ip230,ip240,ip241 : unefined codes received from a DHSG Headset basestation were misinterpreted as Hookswitch indication
Ticket #97732. sometimes DHSG Headset basestations send codes not defined for DHSG which are silently discarded now.
3464 - RTP-DTMF: Must increase duration field when sending RTP-Event with END marker
Ticket #97001. Must increase duration field when sending RTP-Event with END marker to comply with RFC.
3491 - SIP: Dialog-Info was encoded with wrong state attribute
Ticket #97926. Dialog-Info was encoded with wrong state attribute "full". Must be "partial".
3493 - SIP: Logging was wrong
Ticket #98037. Logging was wrong.
3480 - SIP: Trap when terminating a call while re-negotiation is ongoing
Ticket #97675. Trap when terminating a call while re-INVITE is pending.
3490 - SIP: Trap when using STUN
Ticket #97898. Trap when using STUN.
3461 - SUBSCRIBE for MWI not correctly handled after change of IP address
Ticket #96898. SUBSCRIBE for MWI not correctly handled after change of IP address
V9 Hotfix 24 (9061271)
3506 - Corrupt trap buffer when tracing UART messages
Ticket #98533.
3519 - enabled state of an external directory configured via a PBX config template was lost in some cases,
Ticket #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)
3535 - Gateway: Only transparent (clearmode) coder in offer if data call
Ticket #99234. This is a SIP interop issue. Some equipment cannot ignore clearmode coder offers, if not supported.
3515 - Gatway: Configuration of Blockdial Timeout at Routes did not work
Ticket #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.
3534 - H.323: Alternate Registration to IP address did not work, if primary used Discovery
Ticket #99228. If the registration to the primary destination, which used discovery failed, the alternate registration to IP address was sent to the discovery port (1718) instead of the registration port (1719).
3563 - H.323: Fallback to Slowstart after CFNR did not work
Ticket #99600. If a CFNR to a Slowstart endpoint (e.g. XCAPI) was performed after a call to an EFC endpoint, the fallback to slowstart did not work in a szenario with multiple PBXs.
3569 - IP-DECT: Default boot code file name of the IP1202
Ticket #99688. Now the default boot code file name used by the update script is correct.
Ticket #98972. Codec register debugs are enabled with dsp trace. Codec register are dumped after changes to analyse Manits 97903
3523 - IP222 IP232: Noise in the microphone at 6400Hz
Ticket #98941. .
3552 - IP232,IP222,IP241: Rendering error on fkeys during hotdesking
Ticket #99446. Bad pixels right of fkey during hotdesking. But only at fkeys with icon.
3507 - IP232,IP222: Confirming blind transfer with redial key did not work
Ticket #98591. Short user guide:
Press 'redial' before accepting the call, enter telephone number for the call diversion and confirm by pressing 'redial' again.
3510 - IP241: DHSG Headset messages are sometimes wrong
Ticket #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 -->
3497 - IPVA: ETHx Transmit Queue Size Limited To 1MB
Ticket #98187. Was unlimited
3496 - IPVA: Unused ETH1 Could Cause Out-Of-Memory Situation
Ticket #98179. Outgoing packets could queue up without ever getting purged.
3505 - LDAP Replication: ASN.1 Decoding Failed For More Than 100 Attributes Per Object
Ticket #98468. Happend with 100 attributes in a PBX stored phone configuration.
3503 - Linux: Shutdown warning message
Ticket #98438. If Linux is running, a shutdown warning message is shown at the Linux General page now.
3504 - Logging: "Alarm and Event Forward Server" address could not be changed anymore once configured
Ticket #98995. A cause code was missing in CDRs generated by mobility
3560 - PBX Mobility: Call Waiting Facility was missing for waiting calls
Ticket #99572. A caller could not see that his call was waiting
3572 - PBX Mobility: Dial thru
Ticket #99780. When calling the Mobility object from the mobile phone, additional dialed digits are used to call the destination. This is an alternative to using DTMF for dialing. How many digits may be dialed depends on what the network of the mobile phone supports
3528 - PBX Mobility: No DTMF R-Key Features possible after data callback
Ticket #99020. For a mobility call, which was established with data callback no DTMF R-Key Features (e.g. put the call on hold) were possible.
3547 - PBX Mobility: Trap if calling from a slave to an object at the master, with mobility destination on the same slave
Ticket #99400. Null pointer access happend
3567 - PBX: CFNR/CFB on PBX object did not work in some cases
Ticket #99674. The number appended to the call when the forwarding was executed was not correct sometimes
3525 - PBX: No config updates were sent to the phones, when selection of templates was changed
Ticket #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.
3562 - PBX: No Registration redirect back to original PBX
Ticket #99579. This could happen with a config error or during startup when not all objects are read.
3559 - PBX: Trap with operator searches and objects with 'Hide from LDAP' and multi-level nodes
Ticket #99557. An endless loop happened, which caused the PBX to restart.
3545 - phone: a park function key with both 'Number' and 'Name' left empty implies to use 'Number' or 'Name' of the Registration
Ticket #99387. The key works the same way as a key with an explicitely configured 'Number' or 'Name'.
3526 - Phone: Could not configure fkey labels containing single quotation mark
Ticket #98986. Could not configure fkey labels containing single quotation mark.
3518 - phone: directory search highlights first matching entry if the search expression contains any non numeric digit
Ticket #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.
3558 - phone: flag to disable speaker key to prevent conversations in handsfree mode
Ticket #99555. set via
config add PHONE APP /no-speaker-key
3495 - phone: ip222, ip232: support Jabra UC Voice 750 and Jabra Speak 510
Ticket #98092.
3568 - phone: ip222,ip232,ip241: send RTP data to network after hold/retrieve even if remote party does not send
Ticket #99682.
3549 - phone: ip222,ip232: "Phone/Preferences/Use Handset like a Headset" mode did not work
Ticket #99414. on an ip2x2 only the headset function key (Mode:Control) can be used to to start an outbound call or to accept an inbound call in this mode but the headset function key was ignored when no headset was plugged/enabled.
3564 - phone: ip222,ip232: delay ringing to USB headset when a previous call was released immediately before
Ticket #99616. Some USB headsets (even wired ones) need a surprisingly long time to disconnect (up to 500 millisecons). To play the ring tone the headset must be connected again and this may fail before the disconnect is completed. The default delay is one second from start of last disconnect, it can be set by config add PHONE APP /usb-calm <ticks> where <ticks> means 20 ms timer ticks.
3531 - phone: ip222,ip232: ignore HID function of USB headset charging cables
Ticket #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.
3570 - phone: ip222,ip232: Plantronics DA45 with new firmware version (0090) did not work
Ticket #99724. The new firmware rejects commands with STALL which were accepted by the older firmware (the commands were sent with trailing zeros which were silently ignored). This may also apply to Blackwire C420 / C435 / C620 which use the same firmware.
3554 - phone: ip222,ip232: prevent unintentional autoconnect of an inbound call arriving while the headset radio link runs down
Ticket #99497. This happens when another call was just released and the new call arrives while the phone is in idle state but the headset base is running down the radio link.
3542 - phone: ip222,ip232: support USB headsets with two audio input channels from microphone
Ticket #99290. First seen with a Plantronics Blackwire C320 (one of the cheapest wired models)
3502 - phone: ip222,ip232: USB connection sometimes lost until reboot
Ticket #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.
3509 - phone: ip222,ip232: USB headset media connection lost after a release received from a remote conference peer
Ticket #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.
3556 - Phone: NOTIFY(sipfrag) was missing after transfer complete
Ticket #99535. When call replacement (REFER) is completed a NOTIFY(sipfrag:200/OK) must be sent to the sender of REFER (transfering party).
3550 - phone: volume of pickup notification tone is set according to the volume configured for internal ring tone
3557 - Phones: Allow changes to language, ringtones and fkeys if "Allow User Settings at Phone" is activated
Ticket #99548. Allow changes to language, ringtones and fkeys if "Allow User Settings at Phone" is activated.
3511 - SIP/TLS: Using domain name as fall back to proxy name when comparing to the certificat "subject"
Ticket #98643. Using domain name as fall back to proxy name when comparing to the certificat "subject".
3571 - SIP: Display name of "original called party" was missing
Ticket #99739. Display name of "original called party" (first diverting party) was missing in case there were multiple diversions. When processing INVITE with (multiple) History-Info header.
3551 - SIP: Handling of 180 with SDP answer is required after 180 without SDP
Ticket #99428. A 180 with SDP answer must be processed after 180 without SDP has already been received.
3553 - SIP: Must re-create message-summary subscription after re-connecting to server after local address change
Ticket #99488. Must re-create message-summary subscription after re-connecting to server after local address change. Must not re-use call-id and tags for re-subscription.
3520 - SIP: New interop tweak /accept_any_reg_interval
Ticket #98887. For endpoints that do not read the expires value from 200/OK response to REGISTER. E.g. "User-Agent: TRBOnet.Enterprise"
3536 - SIP: New interop tweak /c-line-at-session-level
Ticket #99237. New interop tweak /c-line-at-session-level for clients that do not read RTP address from media description. For clients not compliant to RFC-4566.
3514 - SIP: SDP version not increased when answering an offer where only media-mode has changed
Ticket #98739. If remote side changes from 'sendrecv' to 'inactive' the SDP answer follows this change of media-mode, but SDP version was not increased.
3499 - SIP: STUN not working
Ticket #98223. STUN not working if STUN server IP address is configured. No problem if if STUN domain name is configured.
3530 - SIP: Switch-over to t38 did not work in one configuration scenario
Ticket #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.
3498 - SIP: Trap - not checking array boiundaries
Ticket #98219. Trap - not checking array boiundaries when processing User-Agent header.
3521 - SIP: Trap when sending <dialog-info>
Ticket #98902. Trap when interworking group-indications into dialog-info.
3513 - Voicemail: Allow SMTP Email Port Different Than tcp/25
Ticket #98683. eg in email.xml: ..smtp.foo.bar:888..
V9 Hotfix 25 (9061282)
3581 - AD Replication: Mapping yielding empty CN caused 100% cpu load
Ticket #100209. A detection for data differences was by-passed.
3608 - Correctly distinct Quick Dial and Directory Search object
Ticket #101020. Sometimes die Quick Dial object became a Directory Search object, if the submit has been rejected.
3561 - Escape Mechanism for Flash Directory Objects
Ticket #100128. DSP code 680.12.pf.01. Assert if DSP is not responding added
3603 - LDAP/Expert: Search Page-Size Now 50
Ticket #100847. was 100
3586 - Media: NAT workaround was activated during transfer in some cases
Ticket #100338. NAT workaround was activated during transfer in some cases. Late packets from transferring endpoint may trigger the NAT workaround at the 2 transferred endpoints, making the transferred endpoints redirecting to the obsolete RTP addr/port of the transferring endpoint.
3595 - myPBX: Add node escape prefix to number from PBX LDAP directory
Ticket #100698. Internal numbers might differ depending on the caller, due to node membership. So the internal number has to be adapted for the individual users.
3604 - NTP-Server: use destination address from client request as source address in response to client
Ticket #100940. a response to a client request received via ETH1 was sent with the ETH0 address as source address when routed through default gateway on ETH0. a response with a source address not matching the adressed server is discarded on client side.
3575 - PBX: Admin rights restriction was not displayed after Apply anymore
Ticket #99987. If an admin had restrictions for PBX oder Node, this retrcition was not shown after pressing Apply when editing an object.
3579 - PBX: Case insensitive search for PBX objects did not work for non-ascii characters
Ticket #100195. No result was found when searching with lower case and Name contained upper case
3580 - PBX: When counting calls to slave PBXs, subscriptions were counted as well
Ticket #100203. Only calls with media should be counted
3592 - Phone: Distinctive ringing of emergency calls
Ticket #100490. Distinctive ringing of emergency calls.
3577 - phone: support transparent recording of calls to/from other registrations than the active registration
Ticket #100120. If a registration is configured for 'transparent' recording a call via this registration is always recorded now even if the registration is not the 'active' registration and no recording is configured for the 'active' registration.
3589 - Replication: Distribution Of Trace Flag To Existing Connections
Ticket #100383. Prior to this fix a reboot was necessary to see all trace-output.
3582 - SIP: Don't use uri scheme "sips"
Ticket #100210. Don't use uri scheme "sips" even when using TLS as transport. "sips" is not accepted by MS LYNC.
3596 - SIP: No INVITE is sent if client has registered with a local domain name
Ticket #100728. No INVITE is sent if client has registered with a local domain name instead of ip address.
Ticket #100107. Problems on media negotiation when CFNR is executed by PBX on an incoming SIP call without SDP offer.
3600 - SIP: Workaround for SIP client giving wrong ip address in Contact-URI
Ticket #100822. Workaround for SIP client giving wrong ip address in Contact-URI. 3CXPhoneSystem gives ip address of PBX in Contact-URI. PBX sends upcoming SIP requests to its own ip address.
V9 Hotfix 26 (9061288)
3615 - Gateway: Disabled routes were barely distinguishable from normal routes
Ticket #101422. Light gray background helps
3619 - IP-DECT: Radio call list with OEM PBX
Ticket #101606. The radio call list can't be shown with Mozilla if an OEM PBX is used and a 'R'-key has been pressed. This is fixed now.
Ticket #102240. Simulatenous ec and pcm trace cause packet loss at the DSP host interface.
3628 - IP6000: Prevent blinking error LED on old IP6000 with HW-Build 200
Ticket #102151. Conference DSP driver was started on old hardware that doesnt support the conference DSP
3617 - Linux: Network configuration reset message
Ticket #101492. The reset needed message is sometimes missed if the network configuration for Linux is changed. This is fixed now.
3612 - myPBX: Possible trap on configuring visibility and call diversions
Ticket #101325.
3618 - PBX: Blind Transfer to a different Slave could fail under special circumstances
Ticket #101509. The blind transfer failed when - e.164 style nodes are used - transfer destination is thru a trunk on a different PBX - transfering endpoint is configured to a node, which is another PBX - transfered call is from trunk - ...
3622 - PBX: Boot-Loop when replicating objects with wrong password
Ticket #101879. When a replication was configured, replicating all objects from a different PBX, with different PBX password then configured already on the box, the box entered a boot loop, which only could be stopped with a long reset.
3629 - PBX: ISDN Partial Reroute did not work with e164 configurations
Ticket #102226. The destination of the reroute was wrong (digits missing) if the reroute was done on a trunk object not in the root node with escapes
3616 - PBX: Routing from one slave to another with multiple nodes and overlap dialing could fail
Ticket #101491. Only under more special conditions
3614 - PBX: The checkmark "RTP Proxy - Except Addresses are identical or private" could not be cleared
Ticket #101421. once set, this checkmark could not be unset
3610 - phone: ip110/150/230:- start completion of a prepared dial string in 'numeric' keyboard mode if last char of string is numeric
Ticket #101082. Pressing a 'Dial' function key whith 'Prepare' checked permits further editing of the string configured under 'Number' or 'Name'. The keyboard mode was set to 'alpha' if the string contained any character not permitted in an e164 number. Now the mode is set to 'numeric' when the last character of the string is a decimal digit.
3624 - phone: ip222,ip232: Unintended autoconnect to handset when headset USB port connection is lost while ringing
Ticket #102028. If a Jabra PRO 930 has "Auto sleep Mode" enabled it enters sleep (low power) state when it was not used for 8 hours. When a call arrives this is signaled to the headset. When the Jabra PRO 930 is in sleep state this works for about one second but then the USB port connection gets lost. The handling of such presumably temporary failures was not correct in case of a ringing call.
3584 - phone: Permit Call Intrusion in Silent Monitoring Mode via Recall-Menu if configured at Phone
Ticket #100266. Preconditions: - "Phone/User-x/Preferences/Enable Call Intrusion" checked - "config add PHONE APP /recall-ci-monitor" performed In the Recall-Menu opened when the Menu-key is pressed after dialling a busy user the "Monitor" option is offered in addition to the "Intrude" option.
3620 - SIP: New option "Filter incoming calls" on gateway interfaces
Ticket #101622. New option "Filter incoming calls" on gateway interfaces.
3613 - SIP: STUN server could not be removed from Gateway interface config without reboot
Ticket #101384. STUN server could not be removed from Gateway interface config without reboot. It disappeard from config, but was still used by SIP interface.
V9 Hotfix 27 (9061294)
3726 - DHCP: A 'Coder' manufacturer option longer than 31 characters could not be configured at server and not evaluated by client
Ticket #105071. A coder config longer than 31 characters could not be entered in the field "IP4/ETXn/DHCP Server/Offer Parameters/Coder" and the DHCP Client silently discarded a longer coder config possibly provided by a non innovaphone DHCP server.
3642 - Gateway: Trap on collision of call-complation termination collision
Ticket #103125. A restart could happen if a call-completion monitoring was terminated at the same time from the network and from the user/pbx.
3702 - IP-DECT: Trap with login feature
Ticket #104609. There is a trap in DECT radio if the user login feature is used. This is fixed now.
3632 - IP232,IP222: Wrong keyboard input
Ticket #102369. Wrong keyboard input. Key toggle runs even after key was released.
3653 - ip24, phone_orchid, ip6010...: too much padding of short ethernet frames
Ticket #103410. some firewalls complain if short ehternet frames are padded to more than 60 bytes which was mistakenly done in some ethernet drivers.
3714 - IP241 IP222 IP232: Change back to previous DSP code
Ticket #104862. Previous DSP has a better echocanceller. Also the IP241 Handset micrphone parameters are updated. The IP241 handset receiver equalizer is unchanged.
3651 - myPBX: Add node escape prefix to number from PBX LDAP directory did not work
Ticket #103344. Previous fix #100698 did not work.
3643 - NAT: Do keepalive on TCP sessions
Ticket #103133. Otherwise no cleanup of TCP sessions would happen if the remote endpoints are restarted.
3662 - PBX-SOAP: Memory leak when terminating a SOAP session in an unusual way
Ticket #103668. For example if just the network connection is lost while a Poll command was pending a cmd_exec object was leaking.
3659 - PBX: Call Completion was executed on termination of multicast call
Ticket #103628. The fact that the call was accepted was treated as a user action, but the accept was automatic, so no user has touched the phone
3654 - PBX: Master/Slave Max Calls take subscriptions into account
Ticket #103415. Functionkeys with dialog or presence subscription accross PBXs could block calls.
3633 - PBX: Restart when submitting an object (under very unlikely timing conditions)
Ticket #102482. This only happend when during the submitting of the object the HTTP session was terminated extremly quickly.
3664 - phone: ip222,ip232: call waiting not signaled in USB headset when "Call Waiting: beep once" was configured
Ticket #103792.
3657 - phone: ip222,ip232: handle additional product id for 'Jabra BIZ 2400 Mono USB'
Ticket #103560. the versions tested so far had product id 0x2401, newer ones come with 0x2401
3637 - SIP: Config option /take-zero-addr-for-hold did not work anymore
Ticket #102984. Config option /take-zero-addr-for-hold did not work anymore. Since v9hf16 (#85534: SIP: Interop with Genband SBC).
3672 - SIP: Handling of reject for UPDATE for session refresh was not correct
Ticket #103996. Receiving a reject for UPDATE used to refresh a call (session refresh) must be handled like receiving BYE.
3638 - SIP: Memory leak when handling SUBSCRIBE(dialog-info)
Ticket #102997. Memory leak when handling SUBSCRIBE(dialog-info), but only if a endpoint sends overlapping SUBSCRIBE requests to same destination with different Call-ID's.
V9 Hotfix 28 (9061309)
3703 - EDSS1 Interworking: Interworking Of Incoming Partial Rerouting Failed
Ticket #104610. A number field wasn't initialised, leading to an interworking fault at the boundary between EDSS1 and H.450.
3710 - HTTP client: Update of nonce is ignored in digest authentication
Ticket #104733. Once digest authentication is chosen the HTTP client does not accept any more changes to the digest parameters in the same session.
3692 - Increasing memory usage when viewing PBX pages with Kerberos login
Ticket #104506. When the the PBX pages are displayed using a Kerberos login, some command_exec objects are never deleted. This causes increasing memory usage.
3715 - IP232,IP222, IP241: Fkeys may overlap call control
Ticket #104886. Fkeys overlap call control in case of two inbound ringing calls.
3730 - LDAP Replication: Increased Buffer for Computation of Object Differences
Ticket #105189. Was to small
3687 - Media: Do not write error log if RTP is received before media negotiation is complete
Ticket #104390. Do not write error log if RTP packets are received before media negotiation is complete. Error 0x00050003 (Wrong Payload Type received) was generated before.
3693 - PBX E.164 Configuration: Call forward to remote Trunk, should call internal loopback destination
Ticket #104520. If a call forward is configured to the switchboard of a remote location (typically -0, same as trunk prefix) the call should not be sent out to the trunk, but the internal loopback destination should be called.
3733 - PBX Trunk: List of Facilities Could Get Corrupted
Ticket #105255.
3655 - PBX: In e.164 configuration a CFNR on slave PBX for call from local trunk back to local trunk did not work
Ticket #103430. The object initiating the CFNR was lost when the call was sent to the master. Collateral damage of fix: #99674: PBX: CFNR/CFB on PBX object did not work in some cases
3734 - phone: pickup notification tone too loud and tone blurred on ip110,150,200a,,230,240
Ticket #105176. Check Request-URI checking for loop error.
3704 - SIP: Cannot change a password on DECT systems without restart
Ticket #104614. Cannot change a password on DECT systems without restart. Event RAS_UPDATE_KEY was not handled by SIP stack.
3723 - SIP: Memory leak when receiving 403 after 401 for REGISTER
Ticket #105022. Memory leak when receiving 403 after 401 for REGISTER:
REGISTER 401 Unauthorized REGISTER with Authentication 403 Forbidden
3705 - SIP: Memory leak when receiving BYE for a dialog in early state
Ticket #104628. Memory leak when receiving BYE for a dialog in early state. On a call which is not connected yet. On a call where a INVITE server transaction is pending.
3732 - SIP: New config file option /no-cng-tone-detection
Ticket #105219. New config file option /no-cng-tone-detection To keep calling side from initiating switch-over to T.38.
V9 Hotfix 29 (9061320)
3769 - "DELETE" Assertion traces caller
Ticket #106293. For debugging purposes
3809 - Gateway: Fix for call-replacement
Ticket #107318. When handling a call leg replacement the Gateway releases the replaced call before accepting the replacement call. May confuse the replacing endpoint. In case of SIP this regards handling of INVITE with Replaces header. In case of H.323 this regards handling of SETUP with ctSetup facility.
3741 - IP-DECT: Handover not possible for accepted waiting calls
Ticket #105747. Handovers are not possible for accepted waiting calls. This is fixed now.
3858 - IP-DECT: Trap in Radio
Ticket #108556. A rare trap can occur in the IP-DECT Radio with IP1202 and a multi-master solution.
Ticket #106791. Cannot move cursor rightwards in 'indirect dialing' screen. Moving cursor leftwards works, but rightwards doesn't.
3787 - IP6000 IP2000: Crypto driver stopped working after receiving bad SRTP packets
Ticket #106681. Better protection against receiving non-SRTP packets.
3811 - PBX Map: Overlap dial thru a Map Object on Slave with a call via the Master did not work
Ticket #107682. If a phone registers from a different location, any call from this phone has to be routed via the master to check for 'local' objects. In this case overlap dialing thru a Map object on the slave did not work.
3747 - PBX: For pickup a wrong picked from number was displayed in case of nodes with escapes
Ticket #106051. Number adjustment did not work correctly in this case
3738 - phone: fine grained function locking - PHONE_LOCK_USER_INFO bit supresses display of local user info
Ticket #105697. For phones installed in rooms open to the public it's sometimes required to prevent this phones from beeing called by non authorized persons. Adding this bit to the mask defined under "Phone/Protect/Fine grained Function Locking" supresses any info about the local user (number/name/display name).
3817 - phone: importing a phonebook may result in memory leaks
Ticket #107760. happens when phonebook entries containing non UTF8 characters are deleted
3744 - Phones: SIP-Call was rejected if first offered codec was CLEARMODE
Ticket #105932. SIP-Call was rejected if first offered codec was CLEARMODE
3779 - SIP/SDP: Workaround for illegal codec signaling from Ricoh FAX
Ticket #106513. Workaround for illegal codec signaling from Ricoh FAX:
3797 - SIP: Call was dropped after successful session refresh
Ticket #106886. Call was dropped after successful session refresh. Handling of 200/OK for UPDATE was wrong. Was wrong since bug fix #103996 (v9hotfix27)
3802 - SIP: Do not send SDP answer twice (PRACK and ACK)
Ticket #107107. Do not send SDN answer in ACK if it already been sent in PRACK. Regards early media scenarios that starts with INVITE without offer.
INVITE(no sdp)
183(sdp offer) PRACK(sdp answer) 200(PRACK)
180(no sdp) PRACK(no sdp) 200(PRACK)
200(no sdp) ACK(no sdp)
3800 - SIP: Memory leak when receiving BYE while re-INVITE server transaction is pending
Ticket #107066. Memory leak when receiving BYE right after re-INVITE. re-INVITE server transaction is not deleted.
3768 - Voicemail Objekt: Trap During Reconfiguration
Ticket #106274. Wasn't reproducable. Added a counter-measure against a suspected scenario.
V9 Hotfix 30 (9061325)
3848 - IP-DECT: Traces added for login feature
Ticket #108300. New traces added for login feature.
3844 - IP-DECT: Trap with login feature
Ticket #108236. In a rare case a trap can occur if the login feature is used and the master is changed. This is fixed now.
3835 - IP232,IP222,IP241: Display information of pickup fkey truncated too much
Ticket #107962. Display information of pickup fkey truncated too much
3881 - PBX: Leak when sending group indications to an not responding endpoint
Ticket #109270. Each call only a single group indication was removed from the queue, if the rate of group indications was higher then the rate of failed calls, the memory for group indications accumulated.
3853 - PBX: Make Node/PBX at Config Template configurable
Ticket #108506. For management of administration rights
3585 - PBX: Number mapping for calls sent to 'Route Master calls if no Master to' sometimes wrong
Ticket #100299. Esspecially in the case that the master was available but the call could not be forwarded to the final slave.
3856 - SIP: Bug in media negotiation
Ticket #108538. Bug in media negotiation when processing CFNR on an incoming SIP call received without offer.
3841 - SIP: Different registrations for the same AOR from same ip address and same port were handled as one
Ticket #108199. SBC forwards different registrations for the same AOR to the PBX from same SBC ip address and SBC same port. PBX must take this as individual registrations as long as Contact-URI differs. Even is REGISTERs are sent from same ip address and port and for same AOR.
3857 - SIP: Trap when outgoing SIP subscription is canceled while DNS is pending
Ticket #108550. Trap when outgoing SIP subscription is canceled while DNS is pending.
3839 - wrong activation of non-existent spread-spectrum clock
Ticket #108014. happens for all non-ip28 (ip22/24/302/305) gateways if hardware build >= 402, causes the gateways to stall due lack of clocking
V9 Hotfix 31 (9061333)
3944 - DHCP: A server with "Reserved and same Vendor Clients only" checked did not provide leases to IP62 phones
Ticket #111276.
3940 - IP-DECT: OEM Configuration read failure
Ticket #111110. There is a read failure for an OEM configuration. This is fixed now.
3907 - IP-DECT: Transferred remote initiated calls without voice
Ticket #109998. Some transferred remote initiated calls have no voice connection. This are calls which are initiated with myPBX to an external endpoint. This is fixed now.
3937 - IP-DECT: Wrong trace warning
Ticket #111061. A wrong trace warning of the last fix is removed.
3911 - Linux: DNS configuration changes device DNS
Ticket #110073. The Linux DNS configuration changes the local device DNS configuration. This shouldn't be and is fixed now.
3913 - Linux: Empty server identifier and no NTP server if only ETH1 is used
Ticket #110695. If only ETH1 is used and Linux gets a fixed IP address, the DHCP message doesn't include a valid server identifier and NTP server address. This is fixed now.
3861 - Media channel diagnostics
Ticket #108639. Added some traces to support debugging of media channel issues.
3749 - phone: ip222,ip232: support Jabra UC VOICE 550 / 750 Version A headset models
Ticket #106061. Headsets with Version A printed on the package have IDs different to the non-A versions even if the part numbers do not differ. The USB firmware of the Version A headsets differs from the predecessor firmware and requires a special timing.
3946 - phone: ip241,ip222,ip232: sometimes display and USB hadrware did not recover from a reset
Ticket #111309. sometimes the display and USB hardware was not working after a reset (firmware update or configuration change) and a power cycle was required to bring them up again.
3897 - SIP: Insufficient buffer space for response construction
Ticket #109624. CANCEL response was not sent if received CANCEL request was bigger than expected. E.g.
3938 - Trap in webdav client when processing XML directory listing
Ticket #111063. Trap in webdav client when processing XML directory listing.
V9 Hotfix 32 (9061342)
3993 - Gateway/H.323: Trap when canceling an call with Media Relay because out of Resources
Ticket #112690. In this case the cleaup of the outgoing call was incorrect and caused a trap. Only happened when the outgoing call was H.323.
3954 - Linux: Disable feature
Ticket #111515. It can occur that the Linux cannot be disabled. This is fixed now.
3872 - PBX: Calls to 'No Master' were sent with wrong Number under special Conditions
Ticket #109000. This happened if there was a call-forward, which resulted in a call to the master and the master could not send the call to the destination slave either because this Slave was not registered or there was a busy-out setting preventing it.
The problem only happend with a E.164 config.
3996 - PBX: Connected Number missing on calls to some PBX objects
Ticket #112731. The connected number is needed to determin if the destination of the call is internal, which is needed for features like not automatic recording of external calls
3997 - PBX: Handling of enblock (sending-complete) calls improved
Ticket #112746. Respond with CallProceeding, so that if the call is rejected, there is some ack before the reject. Otherwise this would look like an error.
3990 - phone ip222,ip232: Plantronics Savi W440 Headset sometimes mute when controlled by a SOAP-Application or myPBX
Ticket #112578. When an outbound call was started by a SOAP-Application after a call started using the headset Talk-button the headset was mute because the radio link was not established.
3972 - phone: PBX directory config page extended by Address, Gatekeeper ID and Attribute field to permit for non default values
Ticket #111980. By default address and gatekeeper ID of the PBX where the user is registered are used and the 'Long Name' is searched. Now for example this can be changed to use the master PBX and to search the 'Display Name'.
3984 - phone: ip222,ip232: support for Jabra BIZ 2300, Sennheiser Presence UC
Ticket #112335.
3958 - phone: ip222,ip232: USB Bluetooth dongle of some "Plantronics Voyager" Headsets not detected anymore since V9hotfix24/V10rc1
Ticket #111590. The Plantronics bluetooth headsets Voyager PRO UC, Voyager Legend and Calisto 620 come with an USB bluetooth dongle with one of the product codes 0415, 0416, 0417. Dongles with the product code 0416 were not detected.
3976 - phone: set up call with "Sending complete" when the number has been provided before the call is initiated
Ticket #112103. This applies to calls initiated while browsing a directory or a call list, by pressing a dial function key or via indirect dialing, i.e. when a number is entered before going off-hook. To permit for incomplete numbers in a phone directory "Sending Complete" is not set when a number is terminated by a '+' character. Then the '+' is stripped off and the number can be completed by typing more digits. In this case and in case the user goes off-hook before typing any digit the number is assumed to be complete when a '#' character is entered or the "Enblock Dialing Timeout" is reached before the next digit was entered.
The old overlap sending behaviour can be restored by config add PHONE SIG /overlap-sending
3968 - phone: under soap control no audio data was sent when a call was retrieved after another call has been transferred
Ticket #111662.
3969 - Phones: Pickup list sometimes contains doublets
Ticket #111725. Same call could be is listed more than once.
3983 - Since fix #97150 a series of comma didn't extend the wait time before DTMF dialing any more.
Ticket #112334. Since fix: "#97150: phone: DTMF digits following a comma in a number to be dialed were not handled correctly in some cases." from 21.3.2013 a series of comma didn't extend the wait time before DTMF dialing any more. The wait time was always 1 second because only the last comma was seen.
3978 - SIP: REGISTER refresh was rejected "503 Service Unavailable"
Ticket #112186. REGISTER refresh was rejected "503 Service Unavailable" if Contact header contains a not-quoted display-name. Eg:
3974 - H.323: The efc-features were not forwarded accross PBXs from an endpoint, which was called with slowstart
Ticket #112037. If a slowstart endpoint performed a transfer, connecting two efc endpoints on other PBXs, it could happen, that the media negotiation between the new endpoints was slowstart, because the PBX on which the transfer was performed did not receive the efc-featurse
4011 - H.323: unexpected Restart on a very unlikly Hold/Disconnect collision
Ticket #113079. If the two events happened during the same couple of microseconds an assertion in the code caused a restart.
4041 - ip1202: config flag to force reboot when receive interrupts are missing for a certain time
Ticket #113948. By default the the MAC is is reset in case of missing receive interrupt.
config add ETH0 /rx-miss-reboot forces a reboot instead of a MAC reset.
config add ETH0 /rx-wait-max <seconds> defines the maximum time to wait after the last receive interrupt before MAC reset or reboot (default is 30 seconds).
config add ETH0 /itrace activates an interupt backlog which is written to trace buffer before MAC reset or reboot.
4008 - IP222 IP232: Option to disable Energy Efficient Ethernet (EEE) added
Ticket #112979. Needed for some PC´s that loose the link with EEE. EEE status display added to V9 and V10.
4087 - NT ISDN Point to Multipoint Interfaces: Rejecting of a call had delay of 4.5s
Ticket #115118. A call was not rejected right away, but SETUP was resent in case another endpoint would respond. This should be done only if the call was rejected because of incompatible destination.
4097 - PBX: If an endpoint performs a pickup-req, the resulting call should be sent to the requesting endpoint only
Ticket #115569. If on a user two phones were registered and one phone performed a pickup, both phones were ringing for the call to be picked up.
4013 - PBX: The top level Tag of a CDR should contain the normalized number of the endpoint it was created for
Ticket #113112. This was sometimes not the case, but only the extension number without node prefixes was included.
4026 - phone ip222,ip232: added config flag to prevent ringing via speaker when a headset is plugged and enabled
Ticket #113263. config add AC-DSP0 /headset-only
unconditionally disables ringing via speaker when a headset is plugged and enabled. this is done independent of the "Do not Disturb" setting.
4010 - phone ip222,ip232: phone keypad locked when digits are entered too fast (can be unlocked by ESC key)
Ticket #113068.
4044 - TLS: Problem with negotiation of protocol version on server side
Ticket #114046. When the client offered TLS 1.2 or higher, the connection was refused instead of downgrading to the highest supported protocol version.
4032 - Voicemail: <exec> Without "url" Causes Trap
Ticket #113428. A check was missing
V9 Hotfix 34 (9061355)
4118 - (clone of #114012) SIP: Trap in federation scenario
Ticket #116005. Trap in federation scenario when processing INVITE.
4102 - DHCP: Increase maximum length of "Local Networks" and "IP Routing" option strings from 127 to 252 characters
Ticket #115709.
4083 - H.323: No media if a reverse Media call is sent to a slowstart endpoint and tranfered to a EFC endpoint
Ticket #115018. Media negotiation problem which could happen under special conditions when an XCAPI application is performing a call transfer
4133 - ISDN: Missing Ringback on calls sent out to an NT Mode interface
Ticket #116390. If there is no progress indicator indicating inband tones, channels should not switched on for calls sent out to an NT Mode ISDN interface. Otherwise RTP containing silence could switch off any locally generated ringback.
4021 - PBX: Pickup call was not indicated as internal in Connected Number
Ticket #113197. This could cause problems were it is important to know if the call is internal or external, for example if only external calls are to be recorded.
4069 - phone: In Recording Mode 'transparent' or'optional' a 2nd call started by a dialing application could terminate the 1st call
Ticket #114789. This happened when a 2nd call was started by a dialing application and then terminated again while the call was in alerting state.
4061 - phone: In Recording Mode 'transparent' or'optional' internal calls were recorded although 'External Calls Only' was checked
Ticket #114516. happened only to outbound calls initiated by some dialing application. outbound calls initiated directly at the phone and inbound calls were recorded correctly.
4104 - Primary Address for "Alarm and Event Forward Server" of type SYSLOG could not be configured
Ticket #115745.
V9 Hotfix 35 (9061367)
4068 - Gateway trap with 'Out of Memory' when CF-card stucks
Ticket #114781. A CF-card that stucks leads to huge memory allocations of type cf_command containing non-processed CF-requests.
4166 - H.323: Don' generate "Unexpected Message" event for messages received after sending call clearing
Ticket #117248. These messages are not unexpected, but results of a normal collision
4205 - H.323: Internal/External information got lost on Endpoint after Transfer
Ticket #118350. The information if the endpoint to which a phone is connected after a transfer is internal or external was not available on the phone. The recording of internal or external calls only did not work in this case.
4175 - IP-DECT: Resent disconnected calls to handsets
Ticket #117376. Calls disconnected by the gatekeeper can be wrongly resent to the handsets, if the calls are disconnected with the release code Non-selected-user-clearing (26). This is fixed now.
4167 - ip1202: DTMF tones to be sent to the local DECT phone were sent to the voip channel
Ticket #117254. thus DTMF tones sent from a remote peer were not heard by the local peer
4178 - PBX Gateway: Internal Destination flag did not work for outgoing calls
Ticket #117418. If external transfers are not allowed, this flag should allow a transfer to a gateway object for a call coming in from an external source.
4157 - phone: ip222,ip232: support Jabra Pro 935 USB-Bluetooth Headset
Ticket #117060. The Pro 935 looks like a Pro 930 but has a bluetooth- instead of a DECT-headset. The bluetooth-headset can be paired with a mobile phone.
3671 - phone: two way media on a recording connection did not work anymore
Ticket #103956. Since v9hotfix22/v10beta6 recording connections are established in sendonly mode because usually recorders do not send any media data and thus bulks of No Media Data received events may be reported.
Now two way media can be explicitely enabled by checking "Phone/User x/Recording/Two Way Media" if required (for example for the Innovaphone Operator "Greeting Function").
4163 - SIP: Interop to "Thomson TB30 hw4 fw2.77.0.2 00-26-44-30-5D-A2"
4148 - SIP: The 2xx response to the REGISTER request MUST contain, in a Contact header field, a complete list of bindings
Ticket #116835. The 2xx response to the REGISTER request MUST contain, in a Contact header field, a complete list of bindings that have been registered for this address-of-record. Not only the one that has just been added.
V9 Hotfix 36 (9061368)
4218 - PBX Exec: Partner Keys at exec did not work correctly if secretary names matched in the first half
Ticket #118869. If two secretaries were configured with names, being identical in the first half and identical length (e.g. 'Hans' and 'Harz'), for some functions like presence status not the correct secretary was found.
4229 - phone: CLIR couldn't be overridden at phone by "Number Presentation: On" when "Hide own Number" was checked in a config template
Ticket #119270. Overriding via WEB interface works
4222 - phone: ip110/150/200a/230/240: false "Excessive loss of Data" reports when playing Music on Hold (MOH)
Ticket #119055.
4208 - phone: keep remote party name after connect when dialled and connected number differ in first digits only
Ticket #118537.
4239 - SIP: Bug in media re-negotiation
Ticket #119635. Bug in media re-negotiation on media-relay interfaces.
4236 - Waiting Queue: Switching to next announcement by DTMF "0" did not work
Ticket #119525. Switching to next announcement by DTMF "0" did not work.
V9 Hotfix 37 (9061372)
4263 - DHCP-Server: strip leading and trailing spaces from values entered in "IP4/ETHx/DHCP-Server/Offer Parameters"
Ticket #120514.
4259 - FXS with Feature Codes, possible trap on call-completion
Ticket #120384. When call completion was executed, there was a chance of a trap under special conditions
4261 - IP-DECT: Rare trap on IP1202
Ticket #120442. There is a rare trap in DECT-Master if a new call is sent to the radios and there still exists an old call for the endpoint and this call is assigned to an unregistered radio. The trap only occurs on the IP1202, not the IP1200. This is fixed now.
Ticket #120007. - workaround for 10/100Mb/s gemac Rx lockup: the interface is run in promiscuous mode and the driver filters the packets - workaround for Rx Queue Overrun problem: on a Rx Queue Overrun interrupt gemac and phy are rest completely
- for test purposes promiscuous mode can be disabled|enabled by !config add ETH0 /rx-promiscuous 0|1 or temporaryly by !mod cmd ETH0 rx-promiscuous 0|1
4255 - PBX-SOAP: Clearing of call to waiting queue took 30s
Ticket #120231. Call was hanging as if there was in-band information
4266 - PBX: Pickup accross locations from different nodes did not work
Ticket #120638. Adjustment of number was missing
4250 - PBX: Pickup with partner key did not work if visibility was configured by name
Ticket #120070. The call was displayed on the partner key, but the pickup did not work. It did work if visibility was configured with a group.
4265 - PBX: Trap on Park/Pickup
Ticket #120579. If a Park function key is used to park a call and pickup it again, a restart happend. This is a collateral damage from
fix 115569: PBX: If an endpoint performs a pickup-req, the resulting call should be sent to the requesting endpoint only
from v10sr8 and v9hf33
4273 - phone: ip222,ip232: Jabra UC VOICE 550/750 Version A - Microphone occasionally mute
Ticket #120815.
4268 - SIP: Offered wrong local IP address as RTP address
Ticket #120739. Offered wrong local IP address as RTP address. Collateral damage of #119269: SIP: Offered wrong local IP address as RTP address
V9 Hotfix 38 (9061386)
4336 - Adjusting the Volume level of Local Playback of DTMF Tones - Marcus Mülbüsch <muelbuesch@as-infodienste.de>
Ticket #122706. The customer complained that DTMF feedback tones to the user were too loud. There was one obvious reason in the sources: the table of VoiceOutputGain steps was not in sync with the table of SignalLevel steps, i.e. the relation between speech and DTMF level differed depending on the volume setting. The other point is that DTMF tones are perceived louder than the lower frequency call progress tones. Therefore adjusted the SignalLevel steps to correlate with the VoiceOutputGain steps and introduced 6 dB extra attenuation for DTMF tones compared to call progress tones.
4317 - Admin; The input field for the device name showed the url-decoded name
Ticket #122076. If a name with '+' or '%' was configured as device name, these charecters were nocz displayed correctly in the input field.
4300 - IP-DECT: Trace and variable check added
Ticket #121496. Trace and variable check in IP-DECT master added to give some hints of bugs.
4298 - IP-DECT: Watchdog trap on IP1202 with feature codes and handovers
Ticket #121456. There is a watchdog trap on IP1202 with enabled feature codes with firmware V9 hotfix37, if more than one master is used and the handset makes a handover. This is fixed now.
4316 - PBX CDRs: CDR was missing for calls rejected because of busy_on ...
Ticket #122014. This was fixed in version 10 already and now merged back to version 9
4315 - PBX SOAP: Trap if trying to initiate a call for a User with Mobulity configured, without specifiying the device
Ticket #122005. With the PBX SOAP API a call can be initiated for a user, without specifying for which device the call should be initiated. In this case a default device is picked. If an application does this for a user with mobility, a restart happens because of a null pointer access.
4313 - pbx: memory leak when trace is active
Ticket #121897.
4285 - PBX: Transfer with consultation in ringback - no ringback after transfer if performed by analog phone on IP22/.../IP28
Ticket #121197. It is not a problem of the analog interface, but the PBX, which does not play ringback if a retrieve is done before the transfer, which is done by the FXS.
4297 - PBX: When a dyn PBX was deleted, with an id identical to the start of the id of another dyn PBX, this other dyn PBX was broken
Ticket #121433. Some VARS of the wrong dyn PBX were deleted
4322 - phone recording - supress calling tones and call status display for calls to recording device
Ticket #122221.
4320 - phone: do not mute microphone in alerting state
Ticket #122205. For some some analogue endpoints it is not possible to detect when the media connection is really established, it may hapen before connect is signaled to the phone. To prevent confusion when voice is received from remote but the answer is supressed the microphone is unmuted now already in alerting state by default. The former behaviour can be restored by config add PHONE APP /mute-while-dialing 1
4282 - phone: ip222,ip232: audio parameter configuration via command line did not work in some cases
Ticket #121065. Happened with command lines containing options without a value, for example a config change AC-DSP0 HEADSET /spk-volume /mic-volume 5 did not affect the microphone volume. Further input was not validated so big negative or positive values gave confusing results.
4339 - SIP: Bug in media negotiation when processing reINVITE without SDP offer
Ticket #122780. Bug in media negotiation when processing reINVITE without SDP offer. Exclusive codec config got lost during call.
4342 - SIP: Mobility did not work with SIP
Ticket #122942. Mobility did not work with SIP since RTP-DTMF was ot suppressed.
4305 - SIP: Wrong expires value in Contact header of 2xx response for REGISTER
Ticket #121641. Wrong since v10sr9, v9hotfix35. Wrong expires value in Contact header during registration refresh. Correct value in Expires header.
V9 Hotfix 39 (9061388)
4345 - FXS: Trap on very rare race collision of retrieve with call release
Ticket #122980. If a retrieve happens at the same time as a call release of the held call, a trap could happen. The propabilty of this to happen was very low.
4369 - IP222 IP232: Propietary SmartEEE disabled
Ticket #124415. Needed for some PC´s that loose the link with EEE.
This is caused by the "propietary smartEEE " feature of the ethernet phy. When disabled the link is stable
Ticket #123369. This way numbers may be misinterpreted as names.
V9 Hotfix 40 (9061390)
4381 - QSIG: Progress Indicator was missing in PROGRESS on call to Busy User
Ticket #124957. This did not do any harm, but an error log was generated in a Unify PBX.
V9 Hotfix 41 (9061392)
4451 - CONF Interface: Noise on IP800/IP305
Ticket #127695. There are noise and peaks in a conference call on the IP800 and IP305 caused by the CONF interface. This is fixed now.
4466 - IP-DECT: Default config change
Ticket #128180. There is a change in the default configuration which prevents a wrong configuration if a software factory reset is done with the command config clear. IP-DECT handover fails with the wrong configuration. This is fixed now. This fix was lost with V10 SR22 and added again with V10 SR35.
4456 - PBX Map: Hide Connected Endpoint, should be evaluated for calls thru Map
Ticket #127834. This is a feature introduced with v10, which is now merged back to v9
4432 - PBX: Node prefixes missing in CDRs from Broadcast object
Ticket #127131. The number of the object which generated the CDRs shall contain the normalized number including all prefixes.
4461 - Phone sometimes returns to speaker-mode instead to release the call when going onhook
Ticket #127990. This could happen when handset was lifted while holding speaker key pressed or when the speaker was relased at the same time when the handset was lifted.
4450 - phone: ip110,200,230,240: sometimes no DTMF tone was sent for a digit entered in an active call
Ticket #127693.
4414 - phone: ip222,232: support Jabra UC VOICE 750 MS Duo Drk PN 7599-829-409 29/12/14 Version:A
Ticket #126419. Just another model of the Jabra UC VOICE 550/750 Series with a new product ID
4416 - SIP: No reINVITE with updated identity was sent
Ticket #126722. No reINVITE with updated identity was sent on interfaces with media-relay and exclusive codec.
V9 Hotfix 42 (9061397)
4500 - Media Relay: Support for unknown audio codecs
Ticket #129907. Support for unknown audio codecs (e.g. iLBC) when forwarding RTP audio (RTP-Proxy, Media-Relay).
4481 - PBX: Avoid hanging calls after unsuccessful blind transfer to busy endpoint
Ticket #128908. Happen with a Multicast object: Call to multicast, then hold and another call to the same multicast object, which returns busy, then hangup, which initiates a blind transfer to busy endpoint.
4512 - phone: with optional recording a 3-party conference could not be established although recording was stopped
Ticket #130286. Call flow: - A calls B - B answers -> recording is started - A stops recording with redial-key - A opens a consultation call to C - C answers -> recording is restarted - A stops recording with redial-key When A presses the Menu-key a 3-party conference should be started. This did not work anymore since V9hotfix25 and not at all in V10.
4504 - Trap in PBX during boot after downgrade from v11
Ticket #130015. Trap in PBX during boot after downgrade from v11.
V9 Hotfix 43 (9061401)
4548 - GUID generation fixed. Could result in duplicate GUIDs
Ticket #132117. This could create problems in different places for example in Reporting when two CDRs with same GUID were sent.
4536 - IP-DECT: Debug for rare trap
Ticket #131677. A debug message in case of a rare bug is added.
4538 - IP-DECT: Feature codes trap
Ticket #131806. There can be a trap with feature codes caused by an uninitialized variable. This is fixed now.
4528 - ISDN: Call was rejected without cause on channel collision
Ticket #131351. This is a protocol violation and created unwanted log entries on the other side
4535 - PBX Routing: A objcet shadowing a node escape should be used as node extern as default
Ticket #131645. So that for calls to a local trunk no node extern needs to be configured
4549 - phone: ip222/232 - added support for Jabra BIZ 2300 USB Duo headset
Ticket #132131.
4517 - phone: ip222/232 - Config: Reject Automatically Connected Inbound Call routed to Headset if Headset is not plugged or disabled
Ticket #130575. Using the configuration given below an inbound call is automatically connected to the headset if a headset is plugged and enabled, otherwise the call is rejected with cause busy. "Phone/User-x/Preferences/Announcement Calls/Micro On" "Phone/User-x/Preferences/Announcement Calls/Treat any Call as Announcement" "Phone/Preferences/Route Automatically Connected Inbound Calls to Headset (if enabled)" "Phone/Preferences/Reject Automatically Connected Inbound Call routed to Headset if Headset is not plugged or disabled" The last checkmark affects only normal inbound calls. Announcement calls via the PBX MCAST-Announce object or via the "Dial/Announce" Function key will be routed to the speakerphone if no headset is plugged or if the headset is disabled.
4516 - phone: ip222/232 - support for new Jabra EVOLVE headset series and for additional Plantronics Blackwire headsets
Ticket #130552.
4072 - Voicemail failed in Chief+Secretary Scenario
Ticket #114863. Audio prompting didn't start
V9 Hotfix 44(9061405)
4571 - PBX CDRs: Records for calls to objects without registrations were missing
Ticket #132656. Old fix from v10 merged to v9
4562 - SIP: Cannot call from SRTP endpoint to non-SRTP endpoint
Ticket #132448. If called non-SRTP endpoint rejects, the call is re-tried as RTP call without encryption. But no if the caller is a SIP endpoint.
4558 - SIP: Fix for trap
Ticket #132389. Fix for trap due to failed assertion.
V9 Hotfix 45 (9061408)
4608 - AD Replication: Oversized AD Objects Deleted Replicated Objects
Ticket #135034. An internal error code wasn't set by a handling for the resulting decoding failure. The internal error code is now set to error=86, "LDAP Decoding Error". The replication will stop consequentially. Actual cause were the AD objects being member in too many AD groups.
4588 - IP305, include interface licenses in firmware
Ticket #133913. IP305 is sold with full interface licenses only
4605 - myPBX 9: Script error when using popup windows
Ticket #134965. After an update, Internet Explorer 11 showed a script error "Access Denied" when using popup windows in myPBX 9. This problem only occured when using the myPBX launcher. <-- client.js -->
4587 - PBX Waiting: Cause got lost, when disconnecting a waiting calls with SOAP
Ticket #133887. The call was disconnected without cause, which typically resulted in a display "call aborted" on the calling endpoint instead of "user busy" which could be desired by the application.
4610 - phone: ip222,232: audio connection to remote conference peer sometimes lost after a coder renegotiation on one connection
Ticket #135080. This problem occured in the folllowing situation: - a local call (audio+video) was established via an USB headset and then put on hold - a consultation call (audio) to an external peer was established - a 3-pty conference was established but the local connection remained mute
Ticket #139602. If a call setup to the FAX interface includes a user-user-information element, the call hangs. This is fixed now.
4701 - H.323: Call to a Call Broadcast Destination failed under special conditions
Ticket #140491. DTMF dial from a Waiting Queue to a Call Broadcast object with many destinations. This caused special timing in H.323, which created the problem.
4644 - Incomplete HTTP responses from HTTP server in certain circumstances
Ticket #138895. It might have happened, that the HTTP server closed the underlying TCP connection before all data could be sent.
4684 - IP-DECT: Reverse phone book search configuration
Ticket #140086. The IP-DECT reverse phone book search accepts a configuration with phone number types like e164:H,mobile:M now.
3955 - PBX/IP6000: Potential restart if there are groups or boolean objects with non-Ascii characters
Ticket #111560. This is a general problem that the strcmp from the standard lib does not work correctly under very special conditions.
4681 - SIP: Remove all bindings did not work
Ticket #140071. A REGISTER with "Contact: *" was not handled as it should.
V9 Hotfix 47 (9061418)
4801 - IP-DECT: Master trap
Ticket #143206. There is a Master trap because of an uninitialized variable within a facility call. This is fixed now.
4781 - PBX Waiting: Cause got lost, when disconnecting a waiting calls with SOAP (again)
Ticket #142458. Previous fix did not work
4778 - PBX: OEM Registration licenses did not work anymore
Ticket #142371. New handling of license versions broke the OEM licenses
4649 - SIP: Switch from Media-Relay to No-Media-Relay when handling INVITE with Replaces
Ticket #139046. Switch from Media-Relay to No-Media-Relay when handling INVITE with Replaces. May result into no media after INVITE with Replaces.
V9 Hotfix 48 (9061420)
4819 - Fixed possible trap on CF card error
Ticket #144019. The box might have trapped on CF card errors (card full, invalid data read etc.)
4809 - HTTPCLIENT: trap when an application cancels a request inmidst DNS-name resolution
Ticket #143738.
4927 - IP6000: Prevent blinking error LED on old IP6000 with HW-Build 201
Ticket #147092. Conference DSP driver was started on old hardware that doesnt support the conference DSP
4816 - Memory leak in the hardware encryption driver of the IP6000
Ticket #143945. Under excessive load some packets allocated in memory were sometimes not freed in the hardware encryption driver of the IP6000.
4807 - PBX: Execute CFB on Trunk/Gateway objects, if the far endpoint rejects call with busy
Ticket #143675. This is useful to do re-routing in case of a called service is busy
4840 - Phones: Partner fkeys with subscriptions or favourites may not work
Ticket #144707. Partner fkeys with subscriptions or favourites may not work in some cases. But only if partner's name is used as destination and if namesmatch partly. E.g. "name" and "name.x"
V9 Hotfix 49 (9061421)
4876 - IP-DECT: Phone book search filter
Ticket #145556. The configured phone book search filter isn't considered in the search string. This is fixed now.
4872 - IP-DECT: Wrong name with reverse phone book search
Ticket #145482. If there is a similar number in the LDAP directory, the number can be resolved in a wrong name. This is fixed now.
4869 - LDAP: Trap in Flash Directory UI
Ticket #145405. A deleted memory region was re-accessed.
4901 - Licenses containing digits (e.g. G729channel) did not work
Ticket #146486. Problem parsing the license string
V9 Hotfix 50 (9061427)
5004 - OEM Registration licenses did not work anymore
Ticket #150069. Collateral damage of fix: #146486: Licenses containing digits (e.g. G729channel) did not work
4023 - SHA-2 hash algorithms
Ticket #113239. Port the hash algorithm to our platform.
4940 - SIP: Changed handling of History-Info header and stop sending Diversion header
Ticket #147429. Trying to comply to RFC-7044 and RFC-7131.
Decoding: Skip top-most entry "History-Info" (highest index value) if this entry reflects the called party itself.
Encoding: Add top-most entry "History-Info" (highest index value) that reflects the called party itself.
SIP header "Diversion" is removed since it is declared as deprecated (RFC-5806 Category Historic now).
4029 - Support for SHA2 certificates
Ticket #113352. * encoding and decoding * verification * create such certificates on boxes (except sha224)
Ticket #151280. The error correction doesn't work if it is necessary. It results in missed document parts or failed connections. This is fixed now.
5068 - PBX: CFB on Trunk or Gateway did not work if the call was cleared with DISC
Ticket #151934. This happend for example on ISDN interfaces with in-band busy tones
5080 - PBX: Twin Phone algorythm did not work for transfer/recall
Ticket #152169. A recall after a transfer should also use the twin phone algorythm. For example if one of the phones is busy, the call should be sent to the busy phones only.
5083 - SIP: New config option /send-deprecated-diversion-header
Ticket #152337. Diversion header is not sent anymore since v11r1sr5 / v11r2sr1 / v10sr24 / v9hotfix50. For interop reasons this config option is added. If set the old and deprecated Diversion header is sent.
V9 Hotfix 52 (9061432)
5140 - IP-DECT: Fix for "Wrong name with reverse phone book search"
Ticket #154071. Since the fix "Wrong name with reverse phone book search" it doesn't work. This is fixed again.
5098 - IP2x P30x IPxx10: Tone is sometimes not switched off
Ticket #153129.
5128 - Web-UI: Font-family of input, select, textarea, button did not inherit body style
Ticket #153879. Font-family of input, select, textarea, button did not inherit body style. Using now "font-family:inherit" to have same font-familiy all over.
V9 Hotfix 53 (9061433)
V9 Hotfix 54 (9061435)
5406 - CF/SATA driver: Disturbs Linux SATA driver at start-up
Ticket #167567. The innovaphone CF/SATA driver can disturb the Linux SATA driver at Linux start-up, Linux recognizes a spurious interrupt and disables wrongly the SATA interrupt. The SATA device doesn't work or works slowly. This is fixed now.
5393 - IP-DECT: Forced logout does not store CKI
Ticket #167108. If an users logs in a handset and a previously used handset is logged out, the cipher key index for early encryption isn't saved for this handset. This is fixed now.
5383 - PBX SOAP: Remote number update missing on blind transfer on another PBX
Ticket #166764. The CT-COMPLETE facility used to transmit the new number, was not used to update SOAP call
V9 Hotfix 55 (9061440)
5549 - phone: ip222,ip232,ip112: USB headset support - Jabra EVOLVE 20 MS
The T.38 error recovery IFP packet is wrongly decoded. It can occur if a T.38 UDP frame is lost or swapped and can result in aborted fax connections. This is fixed now.
8542 - phone: ip112/222/232 - support for more Sennheiser USB headsets
wireless: DW Office ML, DW Pro 2, DW Pro2 ML, MB Pro2, MB Pro2 ML, Presence UC ML wired: SC70 USB Control, SC70 USB MS, SC 260 USB Ctrl II, SC 260 USB MS II, SC 660 Ctrl, SC660 USB ML SP 20 Speakerphone, SP 20 ML Speakerphone
V9 Hotfix 57 (9061442)
8861 - phone ip222/232/112: support Jabra SPEAK 510 MS
One more product ID for the USB Bluetooth dongle Jabra LINK 360 coming with this Headset version.