For keep alive pruposes on interfaces without registration. Required for Lync interoperability. (config change TSIP /options-interval 30)
Status:
sip.cpp/h siptrans.cpp/h
SIP: Fast re-routing on gateway interface w/o registration
This feature is required for a special application and is supported only for ip200a/230/240 phones with a directly connected headset (non DHSG). It is enabled via config add INCA_DSP /handset-spy <volume> whith <volume> in the range from 1..8
PBX: Translation of Cause "Call Rejected" to Cause "User Busy" for endpoint objects only
A CFB is triggered by a User Busy. If a CFB is used for example at a Trunk, the CFB is executed when the called remote user returns busy. Because this may be unexpected the CFB was not executed at a Gateway Type Object.
It is now enabled again, because it is useful when connecting external systems which return busy to indicate an out of channels situation
PBX Trunk/Gateway: Round robin within registrations to same device, different devices sequentially
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
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.
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.
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 config add KEYS0 /light-off The lcd backlight can be configured the usual menu way on the phone.
HTTP-Client: Allow user names longer than 16 characters
Phone: Call forwarding (always, busy, no reply) destination now choosable from dial-menu. Usage: enter number or search for phonebook entry, press menu-key, scroll down to choose call-forwarding (always, busy or no-reply) and acknowledge choice in CF-screen.
IP-DECT: Allow setting empty text for idle display
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).
Description: Phone: long function key titles hide idle screen information. Fixed: important idle screen information now shortens the amount of displayed function key name. Following information is now displayed over the function key text: a) crossed bell icon on do-not-disturb (lines 2+3) b) CFU + CFU-destination (lines 2+3) c) missed calls, unread messages and waiting callbacks (line 4)
PBX Waiting: Call forwarded with DTMF mapping was shown in myPBX for each registration
When the master is available the slave stores the licenses from the master including the usage. This stored usage included the licenses used by the slave itself, so if after a reset the master was not available the local usage just added to this.
Now from the stored usage the local usage is subtracted.
phone_orchid: microphone is not mute on a call intrusion in silent monitoring mode / microphone cannot be muted in a conference
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.
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.
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.
In H.323 this only happens for the CallIdentity used for H.450 call transfer message in case of blind transfer without consultation.
This fix breaks compatibility with earlier versions, for this reason this fix is available for version 9,8,7 and 6.
If phones and PBX with versions containing and not containing this fix are mixed the following problems will occur: - A blind transfer without consultation (initiated with the redial key) is not possible - A call which was transfered without consultation is not displayed at the transfered-to phone as transfered
SIP: Unwanted media-relay sessions when using forking/broadcast/multi-reg
Brother fax problem, the first fax is transfered, the next fax transfers fail. Switch to fax from remote is now done without reopening the channel. Closing the channel waits until t38 is switched off. Status:
ac_dsp3.cpp ac_dsp3.h
IP30x, IP1060, IP2010, IP6010: Fax did not work if rerouted from ISDN interface to a Voip destination
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.
In version 9 this non-ascii location information was not correct unicode at all.
The problem happened only if non-ascii characters were used when naming a PBX.
SIP: Generate new SRTP key on every incoming re-negotiation
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. (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.
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. Background image can be selected on phone menu. Now also external background image can be selected.
myPBX: Hide passwords for application sharing and reporting in config
Incoming unsolicited NOTIFY(message-summary) may cause pending control call on Gateway. Control calls are calls (signaling connections) without media channel. These calls are now released.
WEB GUI page cannot be scrolled completely when height of left hand logo is too big
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.
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.
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
most DHSG headsets generate a hookswich signal and enter voice mode when taken out of basestation. This hookswitch signal was simply ignored. Now the voice mode is cleared after one second if there is no other DHSG event before.
SIP/TCP: Transport error when connection is closed by client
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.
Sometimes voice from ISDN/Analog to the IP is muted after some time. Seems to be a problem in the latetest echocanceller. Change to old DSP code until fixed DSP code is available. Status:
ip24.mak
Denial of Service filter in ethernet library did not work
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
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. It is intended to be used as follows:
1. the box is started with DHCP enabled (no initial configuration) 2. the box contacts the DHCP server and gets the ip-address and also the Vendor Specific Information in option 43. Suboption 249 of the Vendor Specific Information specifies the URL of the boot config file. 3. the box polls the TFTP/HTTP server for the config file. 4. the box reads the config file and executes the commands provided in the file
The URL may contain the same meta-character strings an Update Server URL, for example #m (mac-address) The length of the URL in the DCHCP suboption is restricted to 127 characters.
The URL is polled in 5 second intervals. The config file is read and executed by the update process in the usual way. A 'creset' commmand as last command of the file will restart the box with the new configuration without writing any 'config' command options to the flash. After a restart by the 'creset' commmand the boot-cfg URL is ignored. After a restart by any other of the 'reset' commands or by a power cycle the boot-cfg URL is processed again.
On an Innovaphone DHCP-Server configuration of a boot-cfg URL and providing it to clients via suboption 249 must be explicitely enabled by config add DHCP0 /boot-cfg config write config activate Once enabled the URL may be entered under "IP4/ETH0/DHCP-Server/Boot Config URL" and is provided to all clients then.
If an Innovaphone DHCP-Client receivess a boot-cfg URL it is displayed under "IP4/ETH0/DHCP/Boot Config URL".
auto complete dtmf feature codes with '#' after 2 seconds
Optional feature for phones, which are not able to send a '#', e.g. the iPhone. They dial a feature like a cfu with a destination number and after two seconds, the feature code is automatically completed with a '#'.
PBX: HTTP request to initiate call for mobile phone
Do not answer with an error message to unexpected or malformed messages.
This protects against the "Kerberos Server Spoofed Packet Amplification DoS" attack. The attack causes two Kerberos servers to send each other error messages in a ping pong style.
RTP-DTMF: Start handling of RTP-DTMF on reception of END event
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".
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
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.
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.
In fact the call was rejected.
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.
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
In this case as only response to the incoming SETUP a PROGRESS was sent. This meant, that the caller was still in overlap dialing state, so a phone does not send DTMF, but translates input keys to INFO dialing messages.
A CALL-PROC is now sent before PROGRESS, which terminates the dialing.
SIP: Send BYE with Reason header with "Q.850 Recovery on timer expiry"
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.
Under "Menu/Administration/IP Settings/VLAN" there was only a "VLAN Priority" menu item. This menu item did override the 'Priority RTP Data' value but not the 'Priority Signaling' value as entered via WEB configuration. Now the items "Prio. RTP Data" and "Prio. Signaling" replace the "VLAN Priority" item.
IPxx10-sata: trap after config /trace /track activation
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. 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
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.
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.
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.
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.
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"
It seem to be problematic to reset all orchid modules, e.g. the DMA module during software reset. Now only USB and ENET modules are reset, the display gets also reset. The display reset is released in the firmware.
Status:
start_orchid.S platform_orchid.c phone_orchid.cpp boot222.y boot232.y boot241.y
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.
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
some load-balancing implementations send unicast IP packets (specially TCP-SYN) as ethernet broadcast packets. Such packets must be silently discarded if the IP destination address is not the address of one of the local interfaces.
IP222: Cannot leave menu screen with ESC when IP address ctrl was active
-Clear IRQ in handle-interrupt after os_interrupt is too late, since IRQ´s a enabled again and e.g. the timer irq is called again if a lower level IRQ like the enet occurs. -The IRQ needs to be cleared in the serial-irq handler, in all case. After the serial-irq other interrupts are enabled. Status:
ip6010.cpp ip6010.h
On "Unrestricted Digital Information" only CLEARMODE is offered (no audio codecs). On other bearer capabilities no CLEARMOE is offered (only audio codecs).
phone: dialog and presence subscriptions sometimes got lost after PBX restart when phone config was stored on PBX
This happened specially when both "Store Phone Config" and "Discard Config on Phone" was checked in the user object because of a unsubscribe/subscribe race condition.\t
when editing a phone config template the dialing location inherited from a predecessor template was stored in the edited templat
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. Going onhock while another call is waiting starts ringer. After going offhook again the waiting call is accepted, but no media in both directions.
CX0 Wave-Encoding Not Working If Fact-Chunk Present In Header
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)
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: 1) Set presence A with IP phone (fkey shows A) 2) Set presence B with myPBX (fkey shows B) 3) Delete presence with IP phone (fkey shows no presence)
Now Fkey shows presence B.
PBX: Tooltip on "PBX/Config/Log Calls" checkmark wrong
Packets arriving at RTP port must be discarded if the source if not the expected one. To be save against DOS attack and for interop with Lync. In some scenarios Lync starts sending RTP packets while having the call set to 'inactive'.
PBX: Blind transfer with consultation to BC-Conference failed
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. 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.
If "KIRK Wireless Server 300 PCS10__ r3327" calls into PBX and is connected with Voicemail, Voicemail may send re-INVITE with SRTP key. Instead of accepting or ignoring the SRTP option, KIRK Wireless Server 300 rejects the whole SDP offer.
Now we retry the re-INVITE w/o offering SRTP key.
PBX: Forward original received ISDN display element to picking up or forwarded call
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
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.
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.
Debugs added * enable with http://addr/debug.xml DSP trace and DSP control message trace to printout all packets to the DSP with a descriptive string. That allows to analyse the message flow to the DSP after a trap. * for further testing old fax disconnect procedure can be enabled with http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl with "t38 skip fax close".
Status:
ac_dsp3.cpp ac_dsp3.h ac_491.h dsp.xsl
latest touch was treated like the very old version that needed other parameters. Old touch is not supported any more ( only 3 were build)
Status:
edt_touch.cpp
IP241,IP222,IP232: Make password configuration more convenient
Center key should enter selected menu item. Not leaving the current menu screen. ESC key can be used to leave current menu screen. Phone app will ask whether to save changes.
myPBX: Name and Number Display not correct on IM sessions across PBXs
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. Content of multi-line-edit-control was mis-placed when touch-keyboard was activated. Display of first matching directory entriy on indirect dialing screen.
ip241 - monitor mode (handset + speaker) did not work in V9hotfix5
When ä,ö,ü are encoded as a,o,u followed by diaeresis from unicode block 'Combining Diacritical Marks' only a,o,u where displayed. Now ä,ö,ü are displayed.
IP241,IP222,IP232: Support for hebrew and arabic presence notes
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
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.
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
In some application sharing solutions a fixed link can be used to create and join meetings (GoMeetNow, BeamYourScreen).
There are two URLs configured, one for the presenter and one for attendees. When the user clicks the aplication sharing button the links are sent using chat messages.
PBX: Tracing flag turns on tracing in all dyn PBX's as well
a headset key mode can be configured to use the key either to enable/disable the headset (Mode: Enable) or to start/accept/clear calls via headset (Mode: Control)
Bug Fixes
Ip6010 DSP Allow coder change from T38 to voice and back to T38 with local DSP
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.
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.
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 (Lauthören): Level too low
Use independent analog codec channels for speaker and headset receiver. Speaker volume in monitor mode is configured as in handsfree mode.
To change to speaker level in monitor mode the gain of the speaker can be configured with config change AC-DSP0 RINGER /DualOutputModeGain level config activate
level is from 0..63 0 0 -> -32dm 32 -> 0db 63 -> 31db
Status:
ac_codec3.cpp ac_codec3.h
IP241,IP222,IP232: Could not activate first builtin background image from phone menu
If a call was diverted more than once, the phone shows first diverting party (original called number) and last diverting party. (Not only the last diverting party)
myPBX launcher: Unhandled exception when accessing browser object
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. If menu was left with DISC key (instead of ESC key) the changes have been discarded.
Also affects other phones: IP241,IP240,IP230,IP110 Changes are saved immediately when leaving the current screen. Not when leaving "User Settings" or "Phone Setting" screen.
PBX Waiting: Diverting leg1 info not correct when diverting to a Waiting Queue
Statement <pbx-upd-obj type="cfu"..> failed to work properly after being used for diversion manipulation multiple times within a single script session.
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. 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 to avoid TCP timeout when INVITE is to be sent. Signaling interface is marked as down and not used anymore.
IP232: Hiding touch keyboard by touching a control
CFU information in header bar is now displayed even if there's not is enough space between name and number. Either name or number is omitted is required.
IP241,IP222,IP232: Replace triangle by arrow to display diversion/transfer information
Replace quite heavy 'BLACK RIGHT-POINTING POINTER' by much lighter 'RIGHTWARDS ARROW' to display diversion/transfer information on call control, fkeys and call lists.
H.323: A forwarded HopCount>32 could result in a very small HopCount
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
The 'Resource-Priority' Header Field The 'Accept-Resource-Priority' Header Field The 'resource-priority' Option Tag 417 Unknown Resource-Priority response
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.
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
1)Phone->PBX LDAP Search returns normalised number to be dialled by phone. 2)Phone receives info about escape digits when registering at its PBX.
With 1) the PBX includes an object's normalized number into the LDAP search result. With 1) the phone is able to dial that normalized number. With 2) the phone is able to prefix required escape digits to the received normalized number.
Status:
checked in to 10.00, 9.00, 90600
If an USB headset with a known signature (vendor/product id) is plugged it is automatically enabled. This is indicated by the headset symbol in the status line. "Phone/Preferences/Start Outbound Call on Electronic Hook Switch (EHS) Signal" is implied in this case because some headsets will loose state if a hoook signal is ignored. Status:
checked in to 10.00, 9.00, 90600
phone: ip222, ip232: USB headset support - Plantronics C420, GN2000 USB - MS OC Version
Some SIP provider do not provide an Alerting signal when a mobile phone is called. This could result in no ringback signal to the caller or the min/max-alert feature not working.
This new checkmark provides a fake Alerting in case Progress is received
show linux shutdown warning on firmware reset page
Now the Linux menu is always shown and a link is provided to enable or disable the Linux support (RAM reservation). The support state is also saved in the downloaded configuration file and restored with the upload. Update: The support state is only saved in the downloaded configuration with password. The state is not saved in the configuration file with standard password or if downloaded by the update server. Please use the next or a later hotfix instead, see also fix #78836.
Bug Fixes
H.323: Media Negotiation problem with conferences on IP-DECT
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. 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
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.
The call which was the active call at start of a conference had to be cleared manually but the call which was on hold at start of conference was cleared automatically. Now any call will be automatically cleared when relesed from remote.
Status:
checked in to 10.00, 9.00, 90600
phone_orchid: remaining call mute after remote relase for the call which was the active call at start of a conference
In pre-V9 firmware hostnames were stored latin1-encoded. Names contaning non-ascii latin1 charaters must be converted to UTF8 before display.
Status:
checked in to 10.00, 9.00, 90600
Lifetime of an INVITE trasnaction is not limited by any timeout after provisional response has been send/received. Sudden death of a caller make calls hang forever. Now overall lifetime of an INVITE server transaction is limited to 3 minutes. After expiration fimnal reject response is sent and call is released.
IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax failure after transfer
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.
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.
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.
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.
Dummy RTP data is sent just in case a NAT router is within the media path to set a UDP mapping in case both legs of the call contain a NAT router.
An example for such a situation is a call coming from a SIP provider thru a NAT router to the PBX, which forwards the call back out to the SIP provider. The NAT router won't get RTP data from inside to set the mappings.
The dummy RTP was sent to all legs of the call, but it is better to send it to outgoing call legs only, because endpoints calling in may turn off a local ringback tone when receiving dummy RTP
H.323/SIP: Avoid delayed SDP within outgoing calls as far as possible
If media renegotiation is needed, to one side of the call an request for a media proposal (in SIP terms, this is an INVITE without SDP) is sent. The media proposal (in SIP terms SDP offer) is then forwarded to the other side.
The request for an offer should if possible not sent with the initial call, because there is equipment which does not handle 'delayed SDP'
Media Relay: Don't terminate T.38 protocol in media relay, forward transparently
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.
If first media answer was received with a PROGRESS message, the call leg from the radio to the master was switched to progress (no EFC features are defined for the Progress message).
This could cause media problems later in the call with hold/retrieve/transfer.
SIP: REFER does not work as expected on Gateway interfaces without registration
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.
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.
TEL port of ip3010 gateways configured in NT mode do not get Physical Link up indication. This problem applies to V9hotfix7 up to V9hotfix10.
Status:
ip6010.cpp
SIP: Max forward value of 32 could be too small for some provider
For a starting value of max-forwards a value of 32 was used, because this is the maximum value in H.323. This was too small for some sip providers. Starting value now increased to 64 and on H.323 the half value is transmitted.
SIP: Re-negotiation for T38 did not work in media-relay scenarios
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.
In preparation for the new DECT feature DECT security there will be new attributes for the endpoint data which must be taken over. With this fix the innovaphone PBX supports the new attributes if the user is edited.
phone: ip222, ip232: Jabra USB Headset feature "Reject incoming call" supported now
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.
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.
In case of severe network problems, it could happen that the status displayed on a Boolean function key was wrong and was only corrected when the boolean status changed.
DHCP Server Identifier was cleared after editing the DHCP-Server page
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.
Contact-URI should match the Request-URI of the SUBSCRIBE. Also the Message-Account URI in "simple-message-summary" was wrong as result of the wrong Contact-URI.
Config: Could not dynamically set or reset /trace on the LICENSE module
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.
When checking the supported browser features, Firefox thows an uncaught exception if cookies are deactivatd by the user. Therefore myPBX is stuck in the "Loading" screen instead of displaying a configuration hint.
IP241,IP222,IP232: Show lengthy number information on Partner fkey
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.
Ther was a not obvious size limit for packet::put_head and packet::put_tail, which caused a trap if the size was exceeded. This could happen if a CDR exceeded a certain size.
SIP: Domain Name System (DNS) names compared case sensitive
RFC-3551 4.5.2 Even though the actual sampling rate for G.722 audio is 16,000 Hz, the RTP clock rate for the G722 payload format is 8,000 Hz because that value was erroneously assigned in RFC 1890 and must remain unchanged for backward compatibility. The octet rate or sample-pair rate is 8,000 Hz.
Codec description must be: a=rtpmap:9 g722/8000 but "AUDC-IPPhone" sends: a=rtpmap:9 G722/16000 in SDP offer.
SIP: Problems with DNS resolving of proxy adresses
If LOG server is configured but not reachable the device will buffer arising LOG entries until a limit of 300kB. This limit was to high for old black/white telephones. Now not more then 1% of DRAM size is used for LOG buffer.
PBX: When editing a Node object it was changed to a PBX object
If for example Number and Name is configured but the registration result provides the Number only, the configured Name must not be used in further signaling operations (diversion queries ...)
Status:
checked in to 10.00, 9.00
PBX: Possible trap on calls from misconfigured nodes/PBXs (node parent loop)
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
``Survivability´´ mode is used by WLAN phones. In this mode the settings of the 'saved lease' (IP addr, mask, ...) are used until a fresh lease is received. The 'saved lease' is the last lease received from a server, it is kept over a reboot.
When WLAN coverage is lost for a while and then regained a DHCP restart is requested to get a fresh lease from a server in a possibly different network. If this happened while using the 'saved lease' the phone lost it's (saved) IP address.
Status:
checked in to 10.00, 9.00
AC-DSP3: Switch trace off if the DSP Host interface shows an error
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.
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.
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.
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.
A scenario which did not work was A calls B, B does consultation to C, B Transfers, C does consultation to D, C transfers with B on different PBX then A. After this the conference id on the call on A should be identical to the conference id on D. This was not the case.
SOAP/TAPI applications which are keeping track of transfered calls could have a problem with this.
AD Replication: LDAP filter encoding failed, when Poll Timer was configured
The conference ID is used (SOAP/TAPU, CDRs) to associate different call legs to the same call. After a transfer two calls, which have been seperate are connected, so one of the call legs has to change its conference ID, so that the resulting call has a single conference ID again.
There was a complicated logic implemented in the PBX to decide which conference ID should be used, this is now changed to a simple logic: The conference ID of the call on which the transfer is performed, is used.
Example:
A calls B, B does a consuktation to C, and B transfers A to C - This means the transfer is performed on call leg A, so the conference ID of the original call A-B is used for A-C
Gateway: Routing of incoming SIP calls may not work
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.
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.
A call from a PBX, which is sent back to the same PBX is not counted anymore. This can happen because of node-extern. Incoming calls at master, which are above the limit are rejected now. They can be rerouted on the slave with "Route Master calls if no Master to"
SIP: New config option for endpoints not refreshing their registration during call
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
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.
- handsfree speaker equalizer enabled - handset mic and receiver equalizer smoothed - ADC gain reduced, input gain increased ( after ec ) to avoid clipping - halfduplex mode disabled
IP222 IP232 IP241: repeated ethernet link status 1000M wrong
For example if an application used an Waiting Queue object to monitor for incoming calls and redirected these calls to agents. The agent receiving the call could not see if the call was diverted to the waiting queue already.
SIP: Send 200/OK for MESSAGE(text/plain) when accepted by application
XML element content requires some resevered characters to be escaped (<>). These escape sequences (> or <) must be un-escaped onthe receiving end.
IP241,IP222,IP232: Two status symbols may overlay each other
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.
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.
- 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
Now the ready LED shows the green blinking during long reset. The Ethernet LEDs are initialized directly after reset to overwrite the default setting that swaps link and speed.
X.509: Creating certificate containing IPv4 address did not work
In some cases only the accepted call was mute and the next call was OK again but the Jabra LINK 14201-30 lost the USB connection in most cases. Delaying the HID-commands sent to the headset solves this problem.
PBX: Objects list filter for numbers did not work correctly anymore
If the PARI function (only IP1202) of the IP-DECT Master is disable, configuration changes on the System GUI do not effect anything. The settings for the local coder are disabled on this GUI page now.
IP-DECT: System settings not to dynamically connected radios
In scenarios that operate one Ethernet port with 1000M and the other with 100M the switch througput was low. Now the 1000M port is reconfigured to 100M, and the throughput is high.
SIP: Support for multiple audio media descriptions
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.
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.
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.
Sometimes USB headsets come with a signature different from the signature of similar headsets which are already supported. A "vars create KEYS0/HID-MAP p <map>" maps the new signature to an existing one. <map> format is manufacturer:product=manufacturer:product the second manufacturer:product tuple is the signature of an already supported headset, 'manufacturer' and 'product' are plain 4 digit hex numbers without a "0x" prefix.
IP2x2, IP241: Coder Preferences for prefered coder G.722 suboptimal
In case G.7222 was selected as prefered coder and the called endpoint did not support G.722, as next best coder G.729 was selected. This is typically not what is desired in such a case, G.711 is the better alternative in this case
myPBX: Show version of launcher in the list of sessions
With this new version the following headsets are supported for call control: - Jabra GO 6430 (Jabra LINK 350 USB with firmware 5.4.17 or later) with product id 0xa342. Please select the first device. - Jabra SUPREME UC (Jabra LINK 360 USB) with product id 0xa346. Please select the first device. - Jabra PRO 9470 with product id 0x1042. - Sennheiser VoIP USB headset (SH 350 IP) with product id 0x0008. - Sennheiser DW Office with product id 0x740a. Please select the first device. - Sennheiser CEHS-CI 02 (USB adapter cable) with product id 0x0030. Please select the second device.
PBX: Description was missing for DECT System object
"Services/Logging/Log Server/Log Server Shadow/Address" defines the adress of a second server. "Services/Logging/Log Server/Log Server Shadow/Enable" starts/stops logging to the second server. Except the address the configuration for the second server is copied from the first server.
Bug Fixes
Incorrect disk usage calculation for more than 4GB
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.
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.
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.
If the DHCP-client gets a lease containing a WINS-server address and a NETBIOS node type P or M (1 or 2) the client tries to register it's NETBIOS-name (ipxxx-xx-xx-xx) with the WINS-server. The TTL returned by the server in the registration response determines when a name refresh has to be sent.
Phones: Presence info during ringing state may show garbage data
In case a presence update arrives at the phone while phone is in ringback state. Have been observed in conjunction with call forking with mobility only.
myPBX: Support contact names containing a single quote
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.
The execlusive coder/media relay config is used to avoid media negotiation problems with carrier which do not support media renegotiations. In case of a call with reverse media to such an interface, this did not work. This happens for example if a CFNR is configured at a Waiting Queue which redirects a call, which received an announcement from the Queue to such interface.
Kerberos: Allow editing multiple fields in admin UI
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")
Modembypass is enabled on all calls with disabled T.38 and coders G711A or G711U. Switch to modem bypass is indicated in the trace by "switch to modembypass". The feature can be disabled with http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl
Modembypass works best if T38 is disabled on both sides. If T38 is enabled on the called side the CED may trigger a T38 session, this changed back to voice and modem bypass is enabled (if G711 is active). The first modem tone is interupted, but we still have modembypass on both sides. If T38 is enabled on the calling side the calling side stays on regular G711.
IP800 IP6000: reduce probability of false DTMF detection
Allows to trace T38 connection on the PCM port and on the DSP host interface. Use this if fax modem problems are suspected. Enable at http://addr/debug.xml at trace->T38 trace.
Gateway: Routing problem with blockdial route and following matching non-blockdial incomplete routes
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"
Call forwarding is not supported when running SIP. But when the menu key was pressed after entering a number call forwarding options were offered (happened with the primary registration only).
PBX Mobility: Trap in case of Transfer of a call from a mobile endpoint to another mobile endpoint
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
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.
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.
During DTMF receive and transmit levels similar as on IP240 are used
-10db level 0xc0=208--> 22db attenuation also insgesamt ein level von -32db ( bei Vollauschlag ) oder -29dbm. Der alte Wert beim ac_phone3.cpp war -9db
Weitere Diskussion: Es gibt den Fall das inband DTMF zum IP Netz geschickt wird, da gabs in Fall 59846 die Änderung zum IP mit LEV=0x28 -->-10db und Attenuation 0xff-->18db, also mit -28db zu senden. Da das gut funktioniert und die beiden Pegel nicht so unterschiedlich sind unde der ac_dsp3 nicht unterschiedliche Pegel zum IP und zum Codec kann nehmen wir die -28db=-25dbm.
Laut www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-Q.24-198811-I!!PDF-E&type=item Tabelle A-1 sind die -25dbm am unteren Ende, da ist vom Wert A -22..-30 je nach Administration die Rede. Wahrscheinlich ist das kein Problem, bei Audiocodes (ac_dsp2) kann man die sensitivity von -28 bis -38db einstellen (DTMF_DETECTION_ENERGY_THRESHOLD__28dBm)
-->
IP222,IP232: Going offhook in call list always dials first list entry (not touched list entry)
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.
After for example config add AC-DSP0 HEADSET /InputGain 32 config activate the headset icon was cleared on status line and the headset was mute although the headset Talk key was handled.
PBX Waiting: Evaluate Busy on ... Calls for calls to an operator
A signaling loop could be created by calling from a phone registered at one PBX to a phone at another PBX, then putting the call on hold and do the same call again, accept on the other side the waiting call. If both parties do then a transfer there is the signaling loop.
Such loop ist now detected and the call is cleared.
PBX Broadcast: No diverting name sent with broadcasted call
When the external call setup came in with a name identification provided by the external source and there was another name found by inverse directory lookup the name from directory was displayed on the call screen but the name identificication was stored in the call list. Now the name found by inverse directory lookup will be stored.
PBX Routing: Node extern did not work for calls from a trunk marked as local object
The number configured at the PBX object is interpreted in the context of the node of this PBX object. If escapes were needed to dial the WAN trunk, it did not work.
External-UC: Presence info assigned to wrong PBX object
This number was dialed from the node of the Mobility object. This was confusing, because this number was configured at the user and it was also different behaviour as with forking without mobility
PBX: Standyby PBX generated alarms for missing slave registrations, even if active PBX up
The new UserRc codes are executed only when the addressed phone is either in handset, headset or handsfree mode, i.e when calling, connected or disconnected but not when alerting: 16 - change to handset mode 17 - change to headset mode 18 - change to handsfree mode 19 - monitor mode on (add speaker to handset or headset mode) 20 - monitor mode off (back to plain handset or headset mode)
Alarm/Event handling: Authentication for received remote Alarms/Events
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.
'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.
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.
If two-stage dialing (Maps) is used to call a Trunk or Gateway object, the call is sent after a blockdial timeout. If an operator connected the call before this timeout happened, a trap occured.
For this to happen DTMF maps and operators have to be configured on the same Waiting Queue object, with is kind of unusual
PBX: Call to a Trunk/Gateway was not marked correctly as external, if no connected number was received
For a registration with name or number, the information if the PBX password shall be used was always taken from the first device regardless if this was the default device (hw-id identical to name) or not
PBX: Partial Rerouting was prohibited in Alerting State (CFNR)
Both group indications and dialog infos are signaled via a group indication facility. For dialog infos the parked_to_alerting member was overloaded to provide the info as expected by the existing phoneapp. Now the parked_to_alerting member is passed to a phoneapp as received.
IP222 IP232 IP241: Force same speed of the switch ports for 1000M/100M scenarios (configuration option added)
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.
The type="ext" attribute was not set reliably. Additionaly an attribute pseudo was added to the <user/> tag to indicate the type of object the CDRis created for.
http client : authentication was not retried after a failure when the offending request was repeated in the same session
When a httpclient user repeated a failing request in the same session the authentication was not tried again. Thus a change of the client side URL password or a change of the server side password had no effect until a new session was started.
IP6000: Prevent blinking error LED on old IP6000 with HW-Build <110
when for example 022222222 was dialed and the network reported a connected number 03022222222 the display info "022222222 -> 03022222222" looked like a transfer.
As kind of denial of service attack, bursts of incoming DNS requests were seen. The nat process was forwarding these requests to the public DNS. This is a useful function for DNS requests from the private network, but not for requests from the public network.
These DNS requests are now discarded
H.323: Media Negotiation did not work for call with reverse media and media response in CALL-PROC
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)
These parameters set a volume correction factor which is applied at any volume level. Parameter changes are applied immediately even in an active call.
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.
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.
New config option "No blind transfer" to keep Gateway from handling blind transfer requests. If set blind transfer requests are passed through. Handling is performed at the next signaling hop.
Bug Fixes
phone: ip222, ip232: USB Headset could not be disabled via Menu or by Headset Function in Enable mode
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.
For the transfer the CUCM first sets the call on hold and then requests a new media proposal from this call, which we cannot deliver. The request is just ignored, there should be an answer.
IP152: Call replacement (blind transfer) did not work
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.
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.
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.
An USB port failure is indicated when a Plantronics DA45 headset adapter is plugged and a certain kind of table lamp (halogen) is switched on or off. It happens independent of current state of the headset (idle or in call) but only with the abovementioned adapter. The exact reason is not known yet, may be it's an electric spark from the switch of the lamp or some pulse. The fix is to reset the port and to restart the plugin process, a possibly active call is terminated.
phone: coder settings of a "Create Registration" function key were not applied to the created registration
Fix is required for interop with SIP devices sending re-INVITE for session-refresh, but incrementing version field in SDP body, altough there is no change in SDP.
phone: headset function key mode 'control' could be configured via WEB interface only
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.
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.
When the headset talk-key or the phone headset-control-key is pressed while the phone is in handset or handsfree mode the phone changes to headset mode, i.e. headset micro and speaker are activated. The handset or handsfree speaker should be switched off then.
phone_orchid: pressing the speaker key while in headset mode did not switch over to handsfree mode
A "Disable Modification on Phone" checkmark will be provided in the edit menu for each key. If checked the key cannot be edited on the phone anymore. This mechanism works in addition to the phone local key type mask set via "Phone/Protect/Function keys not modifiable on the phone" A key of a type NOT marked as ``not editable´´ in this mask can be made ``not editable´´ by setting the above mentioned checkmark A key of a type marked as ``not editable´´ remmains not editable, independent of the checkmark setting.
phone: "Spare" function key to reserve key positions for administrative purposes
If a hold notify message is received from a remote party and the conference mode is active, now the message is forwarded to the conference unit. This prevents the music on hold in conference calls. The state is also shown in the radio call list.
Webdav: Write information into trace if DELETE fails because file is in open state
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.
Some WEBDAV tools garble line end when a text file is stored after editing. Last seen \\r\\r instead of \\r . Any sequence consisting only of \\r chars should be read as one line end because empty lines have no meaning in an update script.
SIP: Do not interwork holdNotific and retrieveNotific while on hold
During an upload of a complete configuration the command "mod cmd FLASHDIR0 erase-all" will erase all flash directory content. Replication clients are going to receive nil-responses making them assume a certain entry does no longer exist.
Replication clients are now barred from accessing the LDAP server as long as the box didn't process the post-upload reset.
IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: fax bad signal quality events are sent on good fax connections
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)
The automatically generated user objects caused a problem. This could result in a config that caused the PBX to restart in a loop. The export/import was fixed and the PBX does not restart because of the corrupt config any more.
PBX: Trap if a Hold was attempted for a call without media
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
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.
If reading (GET) stops but HTTP session remains open, the file remains in state 'open' and subsequent DELETE request fails with "500 Internal Server Error". Close file and re-try to delete.
HTTP: Chunked transfer fails if the chunk header is not in a single packet
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).
The call-leg to the transfering phone was not cleared by the PBX, so if the phone did not clear this call, it was hanging for ever. Other phones clear such a call after a timeout, but this is only a workaround, the call must be cleared by the PBX
H.323: Problem sending real big signaling messages
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.
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
If a new call joins the conference or a call on hold retrieves the conference, and there are calls on hold in the conference, the music on hold can be heard in the new or retrieving calls. Now this is fixed. This affects all devices with a CONF interface, but not the IP800 and the IP305.
IP241,IP222,IP232: Presence note may is not enough truncated on 'presence' fkey
Sometimes USB headsets get disconnected from USB port because of certain electric pulses. To overcome this problem the headset port is reset and the media stream routed to the handset. If the headset comes up (logical plugged) again in a reasonable time the media stream is routed to the headset again. Otherwise the media stream remains on the handset and the call can be continued by taking off the handset.
PBX Trunk: Diverting as Calling Feature should replace the name as well
With this feature the Trunk object uses a Diverting Number as calling party number. But not only the number, but also the Name and Name Id should be replaced
Phones: Allow lcd_dump.bmp to be retrieved with viewer credentials
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.
If the call transfer target rejects the call in ringing state, no fall-back to the initiator call is done and it is not released. This is fixed now. It is only important for a third party PBX.
H.323: Problem with Media Re-Negotiation on a DECT handover call
The DECT handover call works a little special concerning media renegotiation in a way that local preferences are never honored (the real media negotiation takes place between the original radio and the remote endpoint, the handover radio is just told the result. This special mode did not work correctly
PBX Waiting: User Information Message from announcement interface accidentally forwarded to caller
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.
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".
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.
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.
The SOAP API presents the adjusted number of the peer (called/calling), which is the shortest possible number which can be dialed to call this. It is the same number as displayed on the phone. Sometimes an application needs to know the normalized number of the peer, which is the number in the context of the root node. This number is sent as additional number with the identifier "norm"
IP232,IP222,IP241: Config option to adjust LCD brightness in idle state
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.
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.
Branch value in Via header in ACK request must be new after 200 response. Branch value in Via header in ACK request must be same after non-200 response.
H.323: RTP-DTMF did not work on exclusive coder/media relay configurations
sometimes a humming noise was heard in the USB headset speaker in the setup phase of an outbound call. it disappeared as soon as the call was connected.
phone: an intrusion call set up via Partner function key could not be cleared at the intruding phone via TAPI
This could cause several problems: - When the call was sent to a local user with multiple registrations, the call to each registration had a different conferenceID, so myPBX could not match these calls to actually being only a single call, so multiple calls were dissplayed - The CDRs created for this call could not be matched
SIP: Don't tell application that registration is down when handling redirect response
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.
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.
This could happen when the handset was lifted and kept lifted after the disconnect key was prressed. an inbound call arriving in this state could be accepted via SOAP/TAPI and was connected to the handset but the call could not be cleared by going onhook. Only the disconnect key did clear the call. Now the call is cleared as expected when going onhook.
phone_orchid: pressing speaker key in handset/headset mode switches to handsfree mode, pressing again returns to previous mode
handset/headset plus speaker is not supported on phone_orchid, the previous solution where the connection was dropped when the speaker key was pressed again (see #84297) was perceived as irritating.
IP22 IP24 IP28 IP305: Sometimes the DSP stops after sending CLIP (2)
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.
IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: DSP Update to Version 680.07
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.
If "Phone/User-x/Preferences/Do Not Disturb/Action: ring once" is selected a new inbound call is indicated with a short tone only. Both the the tone and the duration of the tone can be configured under ""Phone/User-x/Preferences/Ring Tones/Do Not Disturb". If not configured the default ring tone is played for one and a half second.
PBX: Support of long user-user-informations by SOAP
Support of long user-user-informations (UUI) for SOAP sessions added. A long UUI is split into multiple short UUIs supported by Q.931. It is required by the FAX interface.
Relay: Support of long user-user-informations by FAX
In some cases it is desireable not to reveal the final destination of a call to a caller. For example a call center agent should not be called directly by the customer.
IP-DECT: Configuration option 'Registration with system password'
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.
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"/>
Needed for SIP interoperability. Some third party SIP PBXs use addresses of 0.0.0.0 to indicate that they don't receive media. This may happens if an endpoint is put on hold. We did not forward such an offer and thus no Music on Hold was heard.
PBX: Busy On ... Calls at PBX objects did not take into acccount that a call may be routed back to Slave
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.
In case an endpoint registers to a PBX from within a private network thru a NAT router, the signaling TCP connection must be maintained in order to be able to receive calls. When the registration is up a dummy call is sent to the PBX to establish the signaling TCP. This TCP connection is maintained after the dummy call is cleared. If this TCP connection is lost (e.g. NAT Router reset), the Registration is cleared and restarted, so that after the re-registration another dummy call is sent.
This is a fix for the previous fix
fix: #89497: H.323: Automatically connect signaling TCP if NAT router is detected
which did not work well
SIP: SUBSCRIBE using old IP address in Contact field
If the IP address is changed at DHCP renew (or network change) the endpoint will immediately do a re-register to update the SIP Proxy with the new IP address. All SIP messages but SUBSCRIBE uses the new IP address in the Contact field.
Voicemail: Memory Load High With Repeated Calls To <store-getnext>
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.
The "Outgoing call restricted" flag on the trunk object to which the call was forked caused the call as a whole to be marked as Calling Line Presentation Restricted.
SIP: Locally configured DNS entries were not used if no DNS server configured
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.
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.
The Waiting object can be used as outgoing dialing object with SOAP. If this is done and the configuration is changed while an outgoing call was pending, a trap could happen
H.323: Potential Trap in special case which could only happen in version 10
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.
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.
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.
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.
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.
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.
Now the VOIP connections between the Master and the Radio use static ports instead of dynamic ones. This is useful if only a few ports should be opened through a firewall. For calls from the Radio to the Master the ports 1716 and 1717 (TLS) are used. For the default Master connection for calls from the Master to the Radio the ports 1718 and 1719 (TLS) are used. For dynamic Radio-Master connections the ports from 1722 are used. Every connection needs two ports.
IP-DECT: Cipher key index request for security test devices
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).
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.
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.
If a call is answered on the mobile phone, it should look identical to the caller to the case that the call was answered locally. This means a connected number from the mobile phone must not be forwarded.
Potential Trap when rapidly switching local Media connections (Conferencing)
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
For unknown reasons some types of wireless headsets stop working after some hours or days. Either the port state changes to disabled or the device rejects control commands with a stall response. In both cases the device is reset and restarted now. If even this fails the complete USB host controller is reset and in most cases the device returns to operational state thereafter
phone: ip222, ip232: Some USB headsets were not detected after a soft reset
In some cases SRTP calls had one-way media because the RTP sequence number wrapped from 65535 to 0 at be beginning of the call before the receiver started receiving and processing packets.
The scope of start sequence numbers for RTP streams is changed from [0;65535] to [0;32767] to make sure that the receiver can always receive packets before the overflow happens.
The calculation of the roll-over counter (ROC) is also improved to be more reliable.
H.323: Unnecessary re-initializing of rtp-channel on incoming calls to phone
This did not create any problems except CPU load and together with another problem in RTP it caused no media on incoming SRTP calls approximately every 1000th call.
IP22,IP24,IP28,IP302,IP305: RTP-DTMF not offered when using a/b interface
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.
If SRV query returns 2 hosts with different port, but no IP address in additional records, SIP starts two A queries for the two host names. Both resolved IP addresses are combined with the port of the most preferred host of the SRV answer.
IP-DECT: Hold/Retrieve could result in no media for incoming SIP calls with SRTP
Reading SIP messages from TCP stream gets confused by huge SIP messages. Presence exchange with external UC was disordered. Increased size limit from 100KByte to 200KByte.
Gateway: Transmitting FAX documents to receiver with polling mode
The cipher key index table is wrongly updated in the Crypto Master if a entry line yet exists. This is fixed now. The Crypto Master is needed for DECT Security Early Encryption.
IP222 IP232: Handset gains changed to avoid low microphone volume
When tandeming VOIP links for trancoding or other purposes DTFM digits were sometimes duplicated. The RTP carried up to 25ms DTMF remaining DTMF, now its only 16ms.
PBX: Trap if user object is deleted, which is used by other applications (e.g. myPBX)
$_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
Call transfer with enbloc dailing fails. This is fixed now. This changes also the R-key handling: after dialling a digit for a consultation call the call must disconnect with R-1 like in ring-back state.
SRTP: Remove traces when packet authentication failed
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.
the initialisation fails with CC=5 on first device descriptor read. after restart of host controller serial_irq() traps in reading the done list. Happens mostly with upload DRAM.
IP-DECT/Relay: Blocked calls by hidden feature code *5/*7
Outgoing calls with beginning number *5 or *7 are blocked by the feature codes module because of hidden new service codes for an OEM device (#79028). This is fixed now.
PBX: Master Slave license update period 10s instead of 10min
There are Fax devices sending wrong (too long) message initially after being called. In this case it is best handled by ignoring these message and wait for the retry instead of disconnecting the call.
SIP/UDP: Sending response to wrong address and port
The availability state (secretary booked into the exec primary group) was not associated with the correct secretary. A compare of the names only covered the first half of the name.
Click sounds at caller side when calling another port of same gateway
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.
The physical location information is based on the redirection of the registration from the PBX at the physical location to the registration PBX. Some information was not cleared with the logout, so re-registration startet with the registration PBX right away.
phone: if a number to be dialled contains a comma, the digits following the comma are sent as DTMF tones after connect
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.
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.
New config file option /register-interval 60 Problem is too weired to explain. This option can be used to set the REGISTER interval to a fixed value regardless of the negotiation.
Bug Fixes
PBX: URI dialing, should not be case sensitive and numbers should be possible
For example if an analog Gateway was registered to a PBX user, and this Gateway did not send a Calling Party Number with the call, the call was sent with Presentation restricted, but without digits. This could affect Billing Applications which are based on CDRs from the Gateway.
phone: ip222,ip232: inbound calls automatically connected to Plantronics Savi W440/740/745 headsets with new firmware Versions
reported for: - Savi W440 with firmware 0118 on USB/DECT Dongle D100 - Savi W740/745 with firmware 0115
reason: the newer firmware versions reject truncated output reports (no trailing 0 bytes) with STALL. The error handling for this case was wrong and caused an autoconnect.
IP-DECT: Wrong GK id of standby Master to Mobility Master
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.
On the small gateways the DSP hangs in some conditions. Now the trace-stop is replaced with an Assert to recover from this situation.
To get a trace of this condition a new trace option is added at dsp.xsl, called txt-trace. This traces the DSP message as text, so that they can be read out after a trap. Typical usage is to enable DSP-trace, DSP control messages DSP data messages and DSP txt trace. DSP pcm trace and DSP T38 trace should be off to avoid excessive debug load.
Also, the CLIP messages are disabled since they caused problems in the past.
H.323: One-way-voice if SRTP call to a Waiting queue is forwarded via Waiting Queue Maps to a phone
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.
When monitoring a Boolean object with SOAP a call is indicated. The local number of this call is set based on the status of the boolean object (00 automatic-off, 01, automatic-on, 10 - manual-off, 11 - manual-on)
PBX-SOAP: UserPark allows to park to another object
Esspecially on the second hold within a call the Channel Close was not sent to the party, which put the other on hold. This caused the channel not beeing turned off on this side (the other side receives music on hold in this case)
IP4 did not work anymore when IP6 was disabled via WEB interface
When a call via this headset was initiated/accepted by the Redial-Key, the Headset(Mode:Control) function key or a CTI application, the call could not be disconnected by pressing the Talk-Key at the headset because the Headset did not report this action.
phone: DTMF digits following a comma in a number to be dialed were not handled correctly in some cases
- in the "Destination Number" configured under "Phone/Direct Dialing" in conjunction with a nonzero "Autodial Timeout": the DTMF digits were sent as dial digits - with a nonzereo "Enblock Dialing Timeout" configured under "Phone/User x/General/Options": sending of DTMF digits was delayed by the configured timeout\t
Fix for the last fix #96660. The FAX interface on the IP800/IP305/IP302 can not connect to a remote device because of wrong protocol events. This is fixed now.
H.323: Fast Unregister/Register operations could lead to failed registrations, in case of fixed signaling ports
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.
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
Bug Fixes
IPVA: Unused ETH1 Could Cause Out-Of-Memory Situation
Observerved with with headsets from various manufacturers. Most probably caused by strong electrostatic discharges to the USB connection cable. In such case also unplugging/plugging of the headset was not noticed by the driver anymore.
Logging: "Alarm and Event Forward Server" address could not be changed anymore once configured
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.
Beim Neuladen des Sequencers bei Änderung der LCD-Helligkeit kommt der UART-Takt gelegentlich zu schnell. Besser noch wäre 2 sequencerprogramme zu definiere, aber wie das geht ist nicht im Orchid/Titan Usermanual nicht offensichtlich -->
SIP: SDP version not increased when answering an offer where only media-mode has changed
Wrong value was calculated, if multiple maps were used in a single route blockdial timeout configuration of a map was lost, when another map was configured.
enabled state of an external directory configured via a PBX config template was lost in some cases,
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)
When the config of the template itself was changed an update was sent, but if it was changed which templates were used on a user object, no update was sent.
Phone: Could not configure fkey labels containing single quotation mark
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.
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).
phone: ip222,ip232: support USB headsets with two audio input channels from microphone
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. Must not re-use call-id and tags for re-subscription.
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.
Phone: NOTIFY(sipfrag) was missing after transfer complete
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.
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. 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.
phone: ip222,ip232: minimize delays in audio stream connect/disconnect operations
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. When processing INVITE with (multiple) History-Info header.