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Revision as of 14:15, 5 April 2016
This is the Firmware V9 Roadmap Document.
The release date of the next Hotfix is planed for the second monday of a month. Please note that this a scheduled and no fix date.
This article is generated automatically. Do not edit! Please see the disclaimer before using the information presented here!
V9 Release
This is the Version 9 Release. It is expected to be released 2011, Week 17. Definition
V9 und V8 PPP port configuration geht nicht richtig auf ip6010
| Status | Closed |
| Id | 63427 |
Da erscheint in V9 bei Auswahl von PPPOE das ISDN Menu, dafr ist bei <none> jetzt kein weiterfhrendes Men sichtbar. Das sollte auch bei V8 so sein.
Bei V8 erscheint derzeit auch bei <none> das ISDN Menu, dafr ist bei V8 das PPPOE Men korrekt vorhanden wenn man PPPOE0/1 als Schnittstelle auswhlt.
Fixed comparison of FTY_IM_MESSAGE in unit tests
| Status | Closed |
| Id | 126338 |
In fty_event_im_message::is_identical fty_event_im_message::data was handled as a null terminated string but it is a bufman buffer.
SRTP: Better diagnostics for SRTP on IP6000 IP2000
| Status | Closed |
| Id | 105043 |
!mem info srtp_socket
V9 Hotfix 3 (90600.03)
Changes included in Version 9 hotfix3 Definition
IPxx10: error handling in sata driver
| Status | Closed |
| Id | 67229 |
Old cards are producing DMA errors that were not handled properly. Try again read/write operation after error recovery.
DECT: IP6000/IP6010/... default config Master mode off
| Status | Closed |
| Id | 67479 |
Now the Dect Master is in mode off by default for the IP6000/IP6010/...
SoftwarePhone: DTMF to voice mail
| Status | Closed |
| Id | 67563 |
Now, sending DTMF with myPBX is correctly working. This fixes sending DTMF to voice mail.
VM: Trap while processing self-forwarded call
| Status | Closed |
| Id | 67570 |
VM: Trap while processing self-forwarded call
SIP: Uninitialized data in SDP offer/answer
| Status | Closed |
| Id | 67617 |
Applies to G.726 exclusive calls only.
| Status | Closed |
| Id | 67618 |
Phone: Main menu scrolling below last item broken.
1st item hould be activated upon down arrow press (done) and screen focus moved up (not being done - bug).
cpld update not working on ip241
| Status | Closed |
| Id | 67629 |
-
Status:
phone_orchid.cpp phone_orchid:config.h xilinx.cpp
and as consequence also ip6010:config.h
SIP: Interoperability with Lync and media-bypass
| Status | Closed |
| Id | 67645 |
Ack contained wrong To-Tag when calling a lync client in media-bypass scenario.
Results into call drop after 30 seconds.
Phone: Automatic key repeat did not work on IP222/IP232
| Status | Closed |
| Id | 67661 |
Automatic key repeat did not work on phone w/o alpha keyboard
Call to Voicemail did not work anymore with SRTP
| Status | Closed |
| Id | 67672 |
This was a collateral damage from fix
http://wiki.innovaphone.com/index.php?title=Support:DVL-Roadmap_Firmware_V9#Allow_configuration_of_SRTP_crypto_suite.2C_to_be_used_for_media_proposals
Phone: long function key titles hide idle screen information
| Status | Closed |
| Id | 67681 |
Description: Phone: long function key titles hide idle screen information.
Fixed: important idle screen information now shortens the amount of displayed function key name. Following information is now displayed over the function key text: a) crossed bell icon on do-not-disturb (lines 2+3) b) CFU + CFU-destination (lines 2+3) c) missed calls, unread messages and waiting callbacks (line 4)
PBX Waiting: Call forwarded with DTMF mapping was shown in myPBX for each registration
| Status | Closed |
| Id | 67682 |
The call was not sent with the original conferenceID, so myPBX could not detect that it was in fact the same call
PBX: License accounting in centralized licensing scenario wrong if master not available
| Status | Closed |
| Id | 67698 |
When the master is available the slave stores the licenses from the master including the usage. This stored usage included the licenses used by the slave itself, so if after a reset the master was not available the local usage just added to this.
Now from the stored usage the local usage is subtracted.
phone_orchid: microphone is not mute on a call intrusion in silent monitoring mode / microphone cannot be muted in a conference
| Status | Closed |
| Id | 67704 |
On a call intrusion in silent monitoring mode the microphone of the intruding party must be mute. In a conference the micro should be muted when the micro key is pressed and unmuted when the micro key is pressed again.
Muting the microphone did work when only one call was active but not when two calls were active as in a intrusion/conference.
Phone: CLIR on text messaging did not work
| Status | Closed |
| Id | 67710 |
CLIR on text messaging did not work
SIP: Interoperability with Lync: Handling of REFER from Lync
| Status | Closed |
| Id | 67713 |
REFER for blind transfer was rejected with "406 Not Acceptable",
due to absence of user part in refer-to URI.
PBX Trunk: Problem with Forking to trunk if multiple GWs are registered to Trunk
| Status | Closed |
| Id | 67720 |
If one of the gateways rejected the call (no channel, not connected, ...), the original call from which was forked was disconnected
Phone: Calls received with CLIR appear in call list with an empty entry
| Status | Closed |
| Id | 67722 |
Description: Phone: Calls received with CLIR appear in call list with an empty entry; now fixed to display the CLIR text: anonymous/unbekannt/...
PBX: License accounting did not work with Unknown Registrations under some special conditions
| Status | Closed |
| Id | 67731 |
It could happen that a registration to a user was not accounted for if the endpoint used for this already had an unknown registration at the time the user was created
User Interface: Alignment of tables fixed
| Status | Closed |
| Id | 67734 |
Some strange alignments were introduced with the last hotfix
Gateway: Interface Name with multiple "Umlaute" did not work
| Status | Closed |
| Id | 67737 |
There was a length limitation of the URL encoded output, which was already exceeded if three "Umlaute" (or any character which is encoded in more the one byte with utf-8), were used
IP-DECT: Packetization could change after handover
| Status | Closed |
| Id | 67738 |
On the new radio the RTP should be sent using the same packetization as was negotiated with the original call
SIP: Fix for early media from Waitng Queue
| Status | Closed |
| Id | 67775 |
PROGRESS after ALERT was not handled by SIP stack.
Now 183 Session Progress with SDP is send after 180 Ringing w/o SDP.
Leak checking improved
| Status | Closed |
| Id | 67783 |
sometimes leaks were falsely detected. Problem if objects are about to be deleted, which were not owned by any module anymore. This happend esspecially with httpclient.
SIP: Generate/add SRTP key on media-relay interfaces
| Status | Closed |
| Id | 67789 |
This enables even partial SRTP (SRTP on one side of media-relay).
Phone: Enable "Activate Registration" without user/password authentication if "Protect Configuration at Phone" set
| Status | Closed |
| Id | 67791 |
Description: Phone: Enable "Activate Registration" without user/password authentication if "Protect Configuration at Phone" set. Activating a registration is a state change, and not a configuration modification, so allow this option.
H.323: A name_id of length 0 resulted in invalid H.450 coding
| Status | Closed |
| Id | 67796 |
An empty name identification received was forwarded in H.323 as invalid H.450. Such a name is now forwarded as 'name not available'.
H.323 Malformed packet
| Status | Closed |
| Id | 67803 |
The ASN.1 encoder had a bug under one special condition: For a constrained character string with a maximum length of more or equal to 16bits, with an effective length of zero, the padding for octett alignment was missing for the zero length bitfield containing the string.
In H.323 this only happens for the CallIdentity used for H.450 call transfer message in case of blind transfer without consultation.
This fix breaks compatibility with earlier versions, for this reason this fix is available for version 9,8,7 and 6.
If phones and PBX with versions containing and not containing this fix are mixed the following problems will occur:
- A blind transfer without consultation (initiated with the redial key) is not possible
- A call which was transfered without consultation is not displayed at the transfered-to phone as transfered
SIP: Unwanted media-relay sessions when using forking/broadcast/multi-reg
| Status | Closed |
| Id | 67819 |
If in incoming SIP was routed to multiple destinations
the final session could be media-relay although not configured.
ip22/24/28/302/305: DSP affected by certain fax tones
| Status | Closed |
| Id | 67821 |
Brother fax problem, the first fax is transfered, the next fax transfers fail.
Switch to fax from remote is now done without reopening the channel.
Closing the channel waits until t38 is switched off.
Status:
ac_dsp3.cpp ac_dsp3.h
IP30x, IP1060, IP2010, IP6010: Fax did not work if rerouted from ISDN interface to a Voip destination
| Status | Closed |
| Id | 67823 |
DSP was not configured correctly
IP2x2: Handset microphone gain increased
| Status | Closed |
| Id | 67843 |
changed from 22.5db to 30db Status: ac_codec3.cpp
IP241: Remote party information truncated more than necessary
| Status | Closed |
| Id | 67872 |
Line 3 of call ctrl
IP-DECT: Changed channel option SRTP crypto suite
| Status | Closed |
| Id | 67904 |
Now DECT system channel configuration option 'Secure RTP' is a drop down box. The DECT Master correctly transmits the changed option. This feature was changed in V9 Hotfix 2, related case #66810.
SIP: DNS problem when SRV response provides no additional records
| Status | Closed |
| Id | 67907 |
If 2-step resolving is required (SRV and A) the service port
of the SRV response got lost and default SI Pport 5060 was used.
SIP: Trap when configuring STUN server on a SIP/TCP or SIP/TLS interface
| Status | Closed |
| Id | 67923 |
STUN is for SIP/UDP only.
myPBX: Always send an ID with CT_INITIATE
| Status | Closed |
| Id | 67932 |
This is needed for interoperability with phones using v9hotfix2 or older. (See #67803)
SIP: Must answer every request - even unknown/unsupported methods
| Status | Closed |
| Id | 67935 |
Lync sends proprietary NEGOTIATE request and waits for response.
Otherwise transport connection is blocked for any upcoming request.
PBX Waiting: A call rejected on a primary operator was never sent to all operators
| Status | Closed |
| Id | 67945 |
if again and again rejected on all primary operators. Problem happend also if rejected by "Do not disturb" on operator phone.
PBX: Master/Slave compatibility problem with version 9 and version 8 and non-ascii characters in PBX name
| Status | Closed |
| Id | 67956 |
In version 8 only latin1 characters were allowed, which means in unicode the high byte was always 0. So it could be ignored and when sending location information between master and slave sometimes the high byte contained 0xff.
In version 9 this non-ascii location information was not correct unicode at all.
The problem happened only if non-ascii characters were used when naming a PBX.
SIP: Generate new SRTP key on every incoming re-negotiation
| Status | Closed |
| Id | 67958 |
Required to have different encryption after transfer.
PBX-CDR: Mobility calls to Trunk, external number was missing in CDR
| Status | Closed |
| Id | 67978 |
only the number of the trunk itself was available
IP150: OEM specific WEB GUI modifications did not work
| Status | Closed |
| Id | 67992 |
manufacturer specific stylesheets have to be be adjusted to the new GUI style
PBX: End of call intrusion was not signaled to the phone
| Status | Closed |
| Id | 68007 |
The call intrusion tone was generated even if the intrusion was terminated
PBX: User Interface Inconsistency. Same thing was sometimes call 'Response Timeout'/' and 'No Response Time'
| Status | Closed |
| Id | 68008 |
Only Response Timeout used anymore
PBX-SOAP: Support UserClear for pending outgoing calls on Waiting Queues
| Status | Closed |
| Id | 68009 |
UserClear was ignored in this state
SIP: Fix for media negotiation with SRTP
| Status | Closed |
| Id | 68067 |
Exchange of SRTP key may fail after hold/retrieve
phone_orchid: DTMF Tones detected in voice data from microphone were propagated to remote side
| Status | Closed |
| Id | 68075 |
DTMF tones may be detected from audible feedback on pressing a dial key in connected state and also from some other source. It's better to propagate only tones requested explicitely via a dial key and not from some external source.
SIP: Trap handling 491 response on reliable transport
| Status | Closed |
| Id | 68093 |
Trap with MAX_BUSY_TICKS
phone_inca: "ETH0/Isolate PC Link" checkmark could not be cleared via WEB UI once set
| Status | Closed |
| Id | 68098 |
Only a WEB UI problem, a "config rem ETH0 /isolate-pc" did help.
SIP: Trap when outgoing SIP call is cancelled while DNS resolving is ongoing
| Status | Closed |
| Id | 68099 |
Trap when outgoing SIP is cancelled while DNS resolving is ongoing
Gateway: Allow configuration of username and password for ENUM/SIP interfaces
| Status | Closed |
| Id | 68147 |
For rare where remote destination server asks for authentication.
(And all remote destination servers ask for same auth or remote destination server s always the same.)
SIP: Interoperability with LinkSys SPA3102
| Status | Closed |
| Id | 68174 |
LinkSys SPA3102 gives "g729a" as RTP payload type mapping:
v=0
o=- 510843041 510843041 IN IP4 192.168.10.20
s=-
c=IN IP4 192.168.10.20
t=0 0
m=audio 16404 RTP/AVP 18 100 101
a=rtpmap:18 G729a/8000
a=fmtp:18 annexb=no
...
Needs to be handled.
IP241: Fix for display rendering
| Status | Closed |
| Id | 68181 |
Display of diversion destination was corrupt.
Pickup fkey labeling was wrong when party information needed truncation.
Gerneral/Admin page was broken if too many authentication servers were configured
| Status | Closed |
| Id | 68231 |
The number of authentication servers is now restricted to 10.
SIP: Interoperability with Lync
| Status | Closed |
| Id | 68232 |
Media negotiation problem on calls coming from on Lync client
and getting forwarded to another Lync client.
IP241: Updated some display text information
| Status | Closed |
| Id | 68234 |
No need to strongly abbreviate disconnect cause text.
Phone: Added loud note to web-ui that bool funnction key can only toggle if boolean object addressed by number
| Status | Closed |
| Id | 68240 |
Description: Phone: Added loud note to web-ui that bool funnction key can only toggle if boolean object addressed by number
phone: intrusion call started in handset mode is not terminated when going on hook when TAPI or operator run on PBX
| Status | Closed |
| Id | 68249 |
With TAPI or operator running on the PBX the the signaling of a busy condition is changed such that a disconnect instead of a release is sent. The disconnect was not handled correctly, the hookswitch state was lost and the next on-hook signal was ignored. TThus teh call could be terminated with the disc-key only.
Phone: Switch presence fkey light on if presence activity is set
| Status | Closed |
| Id | 68256 |
Switch presence fkey light on if presence activity is set
myPBX: Wrong charset was used when authenticating against reporting or changing user password
| Status | Closed |
| Id | 68258 |
Javascript uses Unicode. The PBX and the reporting use UTF8. Therefore strings have to be converted to UTF8 before encrypting them using RC4.
IP241: New fkey backgrounds
| Status | Closed |
| Id | 68304 |
Less transparency to avoid interference with personalized background images
myPBX: Chunked response from application sharing provider did not work
| Status | Closed |
| Id | 68311 |
After receiving HTTPCLIENT_RECV_RESULT, pbx_client did not send another HTTPCLIENT_RECV.
phone_orchid: call pickup via partner/pickup key fails when the key is pressed immediately after audible signal
| Status | Closed |
| Id | 68313 |
When "Audible Signal after alerting" was configured on a partner/pickup key and the key was pressed while or a short time after the audible signal was played then the picked call was mute. Status: files: ac_dsp3.cpp
| Status | Closed |
| Id | 68351 |
External background image source can be configured on web ui.
Background image can be selected on phone menu.
Now also external background image can be selected.
myPBX: Hide passwords for application sharing and reporting in config
| Status | Closed |
| Id | 68352 |
The passwords are moved from the config line to the VARS.
PBX0/MY-A - Application sharing password
PBX0/MY-R - Reporting password
SIP: Failed to register on dynamic PBX
| Status | Closed |
| Id | 68362 |
Attempt to register on a dyn PBX with SIP protocol was rejected with "301 Moved Permanently".
IP-DECT: Adding OEM radios to Kerberos realm did not work with passwords containing special characters
| Status | Closed |
| Id | 68377 |
The password was not URL-decoded when reading it from the UI.
SIP: Pending control calls on gateway
| Status | Closed |
| Id | 68378 |
Incoming unsolicited NOTIFY(message-summary) may cause pending control call on Gateway.
Control calls are calls (signaling connections) without media channel.
These calls are now released.
WEB GUI page cannot be scrolled completely when height of left hand logo is too big
| Status | Closed |
| Id | 68382 |
Height of FHF logo is bigger than that of the default logo, this must be considered when computing size for iframe below tab lines.
DTMF user configuration with invalid checkbox check for presence setting
| Status | Closed |
| Id | 68383 |
The check of the checkmark of the presence setting was wrong.
IP222: Alpha input using the num block
| Status | Closed |
| Id | 68398 |
Automatic switching back to numeric mode when leaving input ctrl.
SIP: Trap when using TLS as transport
| Status | Closed |
| Id | 68410 |
Only if remote side closes transport connection while requests are pending.
phone_orchid: begin of voice mail prompt was cut off sometimes
| Status | Closed |
| Id | 68416 |
Connection of Voip to DSP channel was delayed and thus the first packets of the RTP stream were lost.
body onload attribute can be extended and tab_active method doesn't crash anymore
| Status | Closed |
| Id | 68430 |
Custom PBX object XSL had no method anymore to set the onload attribute of the body. This can be now extended with a XSL template parameter.
Additionally the tab_active method has been called by default and the default value caused the method to crash.
X509: Fix for reading innovaphone info from flash
| Status | Closed |
| Id | 68435 |
Parsing the innovaphone info text was incorrect
License: Be safe against factory reset during license invalidation
| Status | Closed |
| Id | 68447 |
If factory reset is done before license invalidation procedure is complete,
will keep you from completing the license invalidation.
Now the procedure can be completed even after factory reset.
IP241 - handsfree speaker volume to low
| Status | Closed |
| Id | 68451 |
The handsfree speaker volume was too low even when configured to maximum. Now the general output volume is increased by 3 dB. In case of problems the general output volume can be changed by
config add AC-DSP0 RINGER /VoiceOutputGain n
with n = 1..63 -> (-32 + n)db, n = 32 -> 0dB, n = 0 -> mute
wrong link to PPP Interface State Info Help
| Status | Closed |
| Id | 68463 |
Link out of V9 namespace
phone: DHSG headset not reset to idle after a hookswitch signal in idle state
| Status | Closed |
| Id | 68567 |
most DHSG headsets generate a hookswich signal and enter voice mode when taken out of basestation. This hookswitch signal was simply ignored.
Now the voice mode is cleared after one second if there is no other DHSG event before.
SIP/TCP: Transport error when connection is closed by client
| Status | Closed |
| Id | 68578 |
If transaction client closes connection before final response has been sent,
the server tries to open a new connection toward ephemeral port of closed connection.
SIP: Fix for Dialog-Info notification
| Status | Closed |
| Id | 68581 |
Send an empty dialig-info XML after inbound subscription.
Required for interop with Grandstream GXP2010.
SIP: Use secondary proxy address on interfaces without registration
| Status | Closed |
| Id | 68635 |
When secondary proxy address resolved by DNS it should be used to send INVITE to.
SIP: Handling of 491 response on reliable transport
| Status | Closed |
| Id | 68652 |
Re-try of re-INVITE was missing.
SIP: Treat domain part of SIP URI case-insensitive
| Status | Closed |
| Id | 68653 |
According to RFC3261(19.1.4)
SIP: Fix for REFER handling
| Status | Closed |
| Id | 68663 |
Support for attended transfer between two calls from/to different SIP endpoints.
As long as both calls belong to the same signaling interface.
SIP: Problem decoding INFO(application/dtmf-relay)
| Status | Closed |
| Id | 68667 |
DTMF digit was not decoded from message body if whitespace between EQUAL and DIGIT.
E.g. Signal= 5
Phone: Changing config option /sip-hold does not call for reset
| Status | Closed |
| Id | 68691 |
Reset is required and 'reset required" must be displayed.
SIP: Web UI for cause code mapping (fix)
| Status | Closed |
| Id | 68772 |
Did not work as expected
SIP: Out-Of-Memory trap if Group Indications are enabled
| Status | Closed |
| Id | 68804 |
If 'Group Indications' are enabled on a PBX object where a SIP client registers,
an OOM trap may occur in case of heavy call activity on PBX.
Phone: "Keep Calling Party Info on Pickup Key" option from Phone->Preferences not working, fixed
| Status | Closed |
| Id | 68820 |
Phone: "Keep Calling Party Info on Pickup Key" option from Phone->Preferences not working, fixed
Phone: DnD Absence Message containing newline dumped to config-file breaks phone upon config-file upload
| Status | Closed |
| Id | 68898 |
Solution: All flashdir entries cotaining newline now dumped as binary.
IP30x V9 hf2 media problem: Echo canceller not reliable
| Status | Closed |
| Id | 68901 |
Sometimes voice from ISDN/Analog to the IP is muted after some time.
Seems to be a problem in the latetest echocanceller.
Change to old DSP code until fixed DSP code is available.
Status:
ip24.mak
Denial of Service filter in ethernet library did not work
| Status | Closed |
| Id | 68907 |
this filter can be useful to prevent DOS attacks on non routing devices
SIP: Trap when handling SUBSCRIBE on federation interfaces
| Status | Closed |
| Id | 68976 |
SIP client object may be deleted while DNS query is pending.
But only under critical timing conditions.
myPBX: Use display name for sending meeting urls and conference numbers
| Status | Closed |
| Id | 68992 |
For normal chat messages the DN is used as the sender name. For sending the links for application sharing or audio conferences the CN was used.
Phone: Hide calling party on Pickup key fixed
| Status | Closed |
| Id | 68995 |
Phone: Hide calling party on Pickup key fixed
SoftwarePhone: No CT setup with remote connected calls
| Status | Closed |
| Id | 68996 |
The call transfer setup facility is removed in the call setup if the call is a by remote control connected call used in case of outgoing calls with myPBX. This fixes an empty diverting party number information element in the PBX. Now it can be used with a trunk PBX object with the enabled option 'Set Calling=Diverting No', otherwise the calling party number was removed within this object.
PBX: Conference trap
| Status | Closed |
| Id | 68998 |
A zero pointer trap in the broadcast conference PBX object is fixed.
IP222/IP232: Hookswitch on some PCB not working
| Status | Closed |
| Id | 69001 |
Reduce LED frequency from 12kHz to 6 kHz Status: orchid_lcd.cpp
phone_orchid - builtin speaker test did not work
| Status | Closed |
| Id | 69026 |
works only over channel 0
myPBX: Login did not work with system names containing special characters
| Status | Closed |
| Id | 69033 |
Missing URL encoding when sending the connect-request message.
PBX Mobility: Mobility object hanging if call establishment with SOAP/myPBX canceled
| Status | Closed |
| Id | 69052 |
If a call for a mobile phone is initiated by SOAP or myPBX, a call is first sent to the mobile phone. If the mobile phone accepts the outgoing call to the destination is initiated. If the mobile phone did not accept the initial call, no other calls could be done from then on.
IP-DECT: Configuration of Media preferences did not work anymore
| Status | Closed |
| Id | 69056 |
Media preference were ignored. This was a collateral damage introduced with fixes from the last hotfix
SIP: Send OPTIONS at configurable interval
| Status | Closed |
| Id | 67519 |
For keep alive pruposes on interfaces without registration.
Required for Lync interoperability.
(config change TSIP /options-interval 30)
Status:
sip.cpp/h
siptrans.cpp/h
SIP: Fast re-routing on gateway interface w/o registration
| Status | Closed |
| Id | 67593 |
OPTIONS is used to check availability of remote peer.
If OPTIONS fails, not trying to send INVITE.
SIP: New config file option /no-ms-acceptedby
| Status | Closed |
| Id | 67665 |
Some IP phones get confused by this Microsoft extension.
Cisco SPA 303 and Cisco SPA 962 don't stop ringing anymore
ip200a/230/240: handset conversations can be monitored in a directly connected headset
| Status | Closed |
| Id | 67666 |
This feature is required for a special application and is supported only for ip200a/230/240 phones with a directly connected headset (non DHSG).
It is enabled via
config add INCA_DSP /handset-spy <volume>
whith <volume> in the range from 1..8
PBX: Translation of Cause "Call Rejected" to Cause "User Busy" for endpoint objects only
| Status | Closed |
| Id | 67668 |
If the cause is received from a gateway, it is forwarded transparently now
myPBX: Show active sessions on the admin interface
| Status | Closed |
| Id | 67708 |
The active myPBX sessions are shown on the page PBX/myPBX.
Phone: Added new function keys to Phone-UI (Toggle & Prepare Override)
| Status | Closed |
| Id | 67729 |
Description: Phone: Added new function keys to Phone-UI (Toggle & Prepare Override)
PBX: New presence activity "do not disturb"
| Status | Closed |
| Id | 67777 |
If a users sets this presence activity no calls are passed.
PBX: Allow CFB on Gateway Type Objects
| Status | Closed |
| Id | 67827 |
A CFB is triggered by a User Busy. If a CFB is used for example at a Trunk, the CFB is executed when the called remote user returns busy. Because this may be unexpected the CFB was not executed at a Gateway Type Object.
It is now enabled again, because it is useful when connecting external systems which return busy to indicate an out of channels situation
PBX Trunk/Gateway: Round robin within registrations to same device, different devices sequentially
| Status | Closed |
| Id | 67835 |
This way both round-robin or sequential usage of gateways can be configured
improved test for Timeslot Switch Chip of ip6010 ip3010 ip0010 ip1060 and ip6000
| Status | Closed |
| Id | 67840 |
intension is better analysis of hardware problems Status: idt72_drv.cpp, idt72_drv.h
ip22/24/28/302/305/6010/3010/1060/241/222/232: False DTMF detects
| Status | Closed |
| Id | 67844 |
Seen on IP222 and IP6010 Status: ac_dsp3.cpp
phone_orchid: builtin color display test
| Status | Closed |
| Id | 67845 |
After the builtin test function has been started the display test mode is entered when the 'Esc' key is pressed. Numeric keys trigger a full screen test display, all other keys stop the display test mode.
To the keys 0..9 the following patterns are assigned:
DarkGray, White, Grey, Black, Red, Green, Blue, Yellow, Cyan, Magenta
IP-DECT: Configuration XML data for OEM device
| Status | Closed |
| Id | 67858 |
A new configuration XML attribute is added for changing a OEM GUI.
Phone: "Function keys not modifiable on the phone" mask should disable creation of new function keys of masked type
| Status | Closed |
| Id | 67905 |
Description: Phone: "Function keys not modifiable on the phone" mask should disable creation of new function keys of masked type. Currently, only modification of preset function keys is disabled, but the creation of new ones enabled and possible.
Phone: Added command line option to hide Administration Menu and/or MAC/Serial completely
| Status | Closed |
| Id | 67943 |
Description: Phone: Added command line option to hide Administration Menu and/or MAC/Serial completely. See /hide-mask option to PHONE ADMIN-UI in wiki for more information.
IP-DECT: OEM registration string
| Status | Closed |
| Id | 68058 |
The Radio registration string of an OEM device is changed to check a OEM license in the DECT Master.
SIP: Diagnostics of transport error "Remote server certificate mismatch"
| Status | Closed |
| Id | 68142 |
If remote sverer certificate does not match the destination domain name
signaling connection is refused by client.
myPBX: Support for WebEx meeting passwords
| Status | Closed |
| Id | 68172 |
In WebEx a meeting can have a password that must be entered by the attendees when they join. Some WebEx accounts can only create meetings with passwords.
The possibility to configure a global meeting password is added to the PBX/Config/myPBX page.
Phone: Message function key
| Status | Closed |
| Id | 68208 |
Description: Phone: Message function key. Multifunctional depending on number of unread messages. Stores one prepared message (with destination and message text) and presents the new message screen when invoked. If incoming messages pending, display the letter/message icon and jump to incoming-messages subscreen upon invocation.
phone: Finnish translations updated
| Status | Closed |
| Id | 68395 |
no english placeholders anymore, Texts begin with upper case letter
IP150: dimming of key LEDs and LCD backlight
| Status | Closed |
| Id | 68414 |
to save power in special environments the key LEDs can be dimmed by
config add KEYS0 /light-off
The lcd backlight can be configured the usual menu way on the phone.
HTTP-Client: Allow user names longer than 16 characters
| Status | Closed |
| Id | 68499 |
Now user names with up to 64 characters are allowed.
| Status | Closed |
| Id | 68504 |
Phone: Call forwarding (always, busy, no reply) destination now choosable from dial-menu. Usage: enter number or search for phonebook entry, press menu-key, scroll down to choose call-forwarding (always, busy or no-reply) and acknowledge choice in CF-screen.
IP-DECT: Allow setting empty text for idle display
| Status | Closed |
| Id | 68553 |
Some handsets will not show signal strength and battery symbols if idle display is defined, they must be set to empty string.
ac_dsp3: support echo canceller trace
| Status | Closed |
| Id | 68649 |
required to analyze echo canceller problems
Phone: On IP240, make OK key a headset activation key (along with Space Key)
| Status | Closed |
| Id | 68774 |
Phone: On IP240, make OK key a headset activation key (along with Space key).
Enabled from Phone/Preferences : Use Newline/OK Key as Headset Key.
V9 Hotfix 4 (90600.04)
Changes included in Version 9 hotfix4 Definition
Kerberos: Protect against ping pong attacks
| Status | Closed |
| Id | 68822 |
Do not answer with an error message to unexpected or malformed messages.
This protects against the "Kerberos Server Spoofed Packet Amplification DoS" attack. The attack causes two Kerberos servers to send each other error messages in a ping pong style.
RTP-DTMF: Start handling of RTP-DTMF on reception of END event
| Status | Closed |
| Id | 68938 |
Workaround for Bug in MS Lync / Mediation server.
Mediation server changes destination port while sending RTP-DTMF redundancy retransmissions.
SIP: Presence interoperability with ESTOS UC server
| Status | Closed |
| Id | 69050 |
'Do Not Disturb' signaling without presence/tuple/contact element.
SIP: Support for Mediation Server Cluster
| Status | Closed |
| Id | 69051 |
Load balancing and fail-over acc. to spec.
Memory leak with each new telnet session
| Status | Closed |
| Id | 69157 |
Memory leak with each new telnet session (last packet upon exit not cleaned up)
SIP: Media negotiation problem
| Status | Closed |
| Id | 69159 |
On media-relay with exclusive codec we can answer an incoming SDP offer right away without passing to app.
PPP connection fails after LCP renegotiation with different authentication methods (PAP -> CHAP)
| Status | Closed |
| Id | 69167 |
the active authentication entity was not stopped when another authentication was started and and signalled layer down after some timeout
PBX Twinning: Blind transfer to other phone in a twin phone configuration was not possible
| Status | Closed |
| Id | 69170 |
Only normal call to other phone or consulation call was supported
Web-UI: Fixed layout of DynPBX configuration
| Status | Closed |
| Id | 69197 |
Use min-width for fieldset.left
IPVA didn't run on vSphere5
| Status | Closed |
| Id | 69232 |
IPVA didn't run on vSphere5
SIP: Interoperability of INFO(application/dtmf-relay) with Polycom
| Status | Closed |
| Id | 69247 |
Fail to decode DTMF signal, since "application/dtmf-relay" body does not contain any CRLF.
While CRLF is required according to "SIP INFO Package for DTMF".
IP222/IP232: residual echo in handset mode
| Status | Closed |
| Id | 69249 |
Change Codec config Status: ac_codec3.cpp
ip22/24/28/302/305: DSP affected by certain fax tones causing a trap
| Status | Closed |
| Id | 69250 |
Assert removed for testing
Status:
ac_dsp3.cpp
PBX: Phone config was not sent to phone, if phone was power cycled shorty after registration
| Status | Closed |
| Id | 69280 |
The new registration after the power cycle was not detected as new registration but as re-transmission of the previous registration, so it was not reported to the PBX and no phone config was sent
CX0-Wave-Encoding sometimes produced white noise
| Status | Closed |
| Id | 69286 |
CX0-Wave-Encoding sometimes produced white noise.
Also a memory leak was eliminated.
IP-DECT: Hanging call after call transfer
| Status | Closed |
| Id | 69309 |
The hanging call in dectradio is fixed which occurred if a user does an unattended call transfer to an unassigned number and the transferred call is not disconnected.
SRTP: Bad key exchange on H.323/SIP interworking
| Status | Closed |
| Id | 69353 |
MKI was added in SDP due to uninitialized data object.
Pass DNS-TTL to SIP
| Status | Closed |
| Id | 69374 |
In order to do load-balancing SIP needs to ge hold on the TTL of DNS resource records.
SIP: Support of maddr parameter in redirect response for REGISTER
| Status | Closed |
| Id | 69386 |
Set when redirecting REGISTER.
Read when processing redirect response.
G726 codec obsolete
| Status | Closed |
| Id | 69388 |
The G726 codec was rarely used (if ever) in real life. In addition there are signaling problems specially with DECT peers when G726 is selected. Thus G726 is removed from the list of supported coders in all products.
IP222: Input ip addresses does not work
| Status | Closed |
| Id | 69393 |
Entering an ip address on the phone's menu does not work.
All digits are doubled.
IP222: Call reroute did not work
| Status | Closed |
| Id | 69455 |
Using redial key on in incoming not-connected call did not work.
PBX: Call Diversion/Forward to '-' did not always show expected result
| Status | Closed |
| Id | 69457 |
A call diversion to a destination'-' can be used to explicitly no execute a diversion of this type. So if a user has an CFU to '-' and this diversion is valid for a given call (Filter, Boolean), the phone should ring.
In fact the call was rejected.
There was also a problem with CFB in case of "busy on ... calls"
IP-DECT: New radio BMC firmware PCS05Ak
| Status | Closed |
| Id | 69468 |
The new radio BMC firmware PCS05Ak for the IP1200 fixes a trap by the DECT system if more than 255 DECT users without an endpoint subscription are sent to it.
PBX: No CLIR on internal calls did not work for SOAP
| Status | Closed |
| Id | 69474 |
If the features "No CLIR on internal Calls" is activated on a PBX a CLI is sent to the called phone even if the call was sent with "CLI presentation restricted". The same should be case on SOAP/TAPI when monitoring this user.
Now when "No CLIR on internal Calls" is enabled all number information available is provided on SOAP.
PBX: Reject calls without media, if no known facility
| Status | Closed |
| Id | 69477 |
Fixes compatibility issues between versions. For example presence subscription sessions from v8 phones being forwarded to voicemail
PBX Waiting: Not possible to send DTMF to Waiting Queue from myPBX Numeric Keyboard
| Status | Closed |
| Id | 69482 |
The Waiting Queue prohibited the facilities used for this to be sent to calling phone
PBX: Filter for internal or external calls at CFs did not work CFB or CFNR if call already diverted
| Status | Closed |
| Id | 69483 |
Problem:
User A has CFU to User B
User B has CFNR for ext. Calls only to User C
An internal call to A was diverted to B (ok) and after no response diverted to C (nok)
PBX Waiting: In case of "Announcemen w/o Connect" together with Alert-Timeout 0, DTMF dialing was not possible
| Status | Closed |
| Id | 69496 |
In this case as only response to the incoming SETUP a PROGRESS was sent. This meant, that the caller was still in overlap dialing state, so a phone does not send DTMF, but translates input keys to INFO dialing messages.
A CALL-PROC is now sent before PROGRESS, which terminates the dialing.
SIP: Send BYE with Reason header with "Q.850 Recovery on timer expiry"
| Status | Closed |
| Id | 69500 |
If session refresh is outstanding the call is released with BYE with "Reason: Q.850;cause=16".
Better send BYE with "Reason: Q.850;cause=102"
PBX Waiting: No ringback when doing two-stage dialing to a Gateway/Trunk object
| Status | Closed |
| Id | 69531 |
A local ringback is now switched on, when receiving ALERT from called party
SIP: Bug when decoding SRTP keys from SDP
| Status | Closed |
| Id | 69545 |
Bug in decode base64.
Web-UI: Input field to small for SIP-URI
| Status | Closed |
| Id | 69578 |
Input control on "UC" tab of External UC properties dialog was too small.
Only part of SIP-URI was visible.
phone: assume an outbound call to be an external call if connected number info is missing in connect event
| Status | Closed |
| Id | 69581 |
In certain ISDN configurations the PBX can not provide the connected number info in the connect event for an outbound call. In this case the the call was assumed to be an internal call and consequently was not recorded when transparent recording of external calls was configured.
Now an external call is assumed in this case.
IP-DECT: Reset link
| Status | Closed |
| Id | 69584 |
The reset link of DECT System Config GUI page is fixed.
SIP: Do not check remote certificate name when calling PBX client with TLS
| Status | Closed |
| Id | 69598 |
Check is reqired only when calling an unregistered client or when giving INVITE to registrar.
| Status | Closed |
| Id | 69633 |
Under "Menu/Administration/IP Settings/VLAN" there was only a "VLAN Priority" menu item. This menu item did override the 'Priority RTP Data' value but not the 'Priority Signaling' value as entered via WEB configuration.
Now the items "Prio. RTP Data" and "Prio. Signaling" replace the "VLAN Priority" item.
IPxx10-sata: trap after config /trace /track activation
| Status | Closed |
| Id | 69642 |
Instruccion was accessing uninitialized pointer.
IP222: Alpha input using the num block
| Status | Closed |
| Id | 69646 |
Input mode changes back to numeric in screen "Wahlvorbereitung".
Should stay in alpha mode.
IP200: Manufacturing test fails is hwbuild is reprogrammed
| Status | Closed |
| Id | 69662 |
Old hwbuild was not cleared, because on INCA devices the hwbuild is located on a higher address was not cleared. Status: flash_firmware.cpp
H.323: Media Negotiation problem with Lync interop and SRTP
| Status | Closed |
| Id | 69687 |
With a retrieve from the lync after hold (which is signaled as a reinvite with sendrevc) new media parameters were sent, containing new SRTP keys. These new media parameters cannot be used, on the PBX which is initiating new end to end media negotiation at the same time. These media parameter were not ignored properly.
phone_orchid: wrong volume setting when monitor mode is entered
| Status | Closed |
| Id | 69734 |
when monitor mode ise entered by pressing the speaker key in a handset conversation the handsfree speaker is enabled in addition to the handset speaker.
the volume was reconfigured with the wrong value.
phone_orchid: Calls received with CLIR appear in call list with an empty entry
| Status | Closed |
| Id | 69737 |
Calls received with CLIR or without a number/name appeared in call list with an empty entry; now either "anonymous" (CLIR) or "unknown" is displayed instead of a name
SIP: Take SDP id and version as 64bit integer
| Status | Closed |
| Id | 69738 |
o-line of SDP offer/answer is defined as 64bit integer
Interoperability with MX-ONE
Relay: Forward facilities to local destinations
| Status | Closed |
| Id | 69764 |
This fix is related to the previous fix #66629 for V9 hotfix2. Now, facilities are only forwarded, if the destination is a physical interface, not e.g. a SIP provider.
SIP: Trap when handling NOTIFY(application/qsig)
| Status | Closed |
| Id | 69771 |
Traps if no progress indicator present in tunneled DISCONNECT message.
WEB-UI Config Upload screen blank after upload of a file with a lot of failing lines
| Status | Closed |
| Id | 69812 |
When uploading a file with about 50 failing lines the screen was left blank without any info about the failing operation.
phone: "Keep Calling Party Info on Pickup Key" option from Phone->Preferences did not work with very long numbers/names
| Status | Closed |
| Id | 69816 |
speciall when using this option in conjunction with the "Display Name on Pickup/Partner Key" option the calling party info was not correctly displayed
IP6010: SRTP using AES-192 and AES-256 did not work
| Status | Closed |
| Id | 69828 |
Due to a bug in the encryption driver of the IP6010, only AES-128 worked on this platform.
IP-DECT: Location update (OEM)
| Status | Closed |
| Id | 69863 |
Location update with message waiting information is fixed if the endpoint roams. This is only used by OEM devices.
Logging: Threshold for error event "SRTP authentication failed"
| Status | Closed |
| Id | 69920 |
Error event was triggered at the very first decrypt failure.
Some decrypt failure are expected during media re-negotiation.
Trigger this error event after 10 decrypt failures in line.
PBX Calls Page/SOAP wrong number
| Status | Closed |
| Id | 69921 |
In a configuration with escapes for calls from a slave and a node not the root node and the call forwarded to the master, because the number could not be resolved locally, wrong escapes were added to the called number
phone ip222: MWI LED not working
| Status | Closed |
| Id | 69924 |
The MWI LED was not switched on for pending voice mails and/or pickups
Trap during fax transfer
| Status | Closed |
| Id | 69950 |
Seen on Ip28, can happen on AC_DSP3 ( IP22/24/28/302/305/1060/301/6010 ) Status: ac_dsp3.h
phone_orchid: checkmark "Phone/User-x/General/Options/No DTMF Detection" has no effect
| Status | Closed |
| Id | 70114 |
If this checkmark is set DTMF digits entered via keyboard in a connected call shall be sent in-band as voice data, not encoded in RTP-DTMF packets as usual.
Improved protection against Denial of Service attacks
| Status | Closed |
| Id | 69166 |
flooding a box with different kinds of packets may lead to out of memory conditions. The Denial of Service filter in the ethernet layer is activated where required. TCP listening sockets have a backlog limit now. The http service restricts the number of half-open sessions and limits the number of concurrent sessions according to the total memory available on a box.
ISDN interop issue with SecuGATE LI 30 from Sirrix
| Status | Closed |
| Id | 69168 |
The SecuGATE LI30 is sending/receiving ISDN INFO messages in Call Proceeding State (State 3 and state 9), which was not supported
PBX Mobility: Support of transfer on mobile side, not using mobility
| Status | Closed |
| Id | 69275 |
In case that the mobile phone transfers the call to another destination, this call must be removed from the mobility function, so that the mobility function is available for another call
USB Driver merge from v10 to v9
| Status | Closed |
| Id | 69288 |
First step to support USB headsets. This is not functional yet.
SIP: New config file option /hold-notify-as-inactive
| Status | Closed |
| Id | 69293 |
If set, holdNotific is interworked into "a=inactive".
If not, holdNotific is interworked into "a=sendonly".
SIP: New config file option /prefer-pai2
| Status | Closed |
| Id | 69459 |
Interoperability with Telepo:
When receiving INVITE, get calling party id from second P-Asserted-Id header.
new: DHCP manufacturer specific option 'boot-cfg' provides an URL to read config from after any reset except 'creset'
| Status | Closed |
| Id | 69472 |
This option permits to boot boxes with a fresh config provided via TFTP/HTTP without storing the config on the device.
It is intended to be used as follows:
1. the box is started with DHCP enabled (no initial configuration)
2. the box contacts the DHCP server and gets the ip-address and also the Vendor Specific Information in option 43.
Suboption 249 of the Vendor Specific Information specifies the URL of the boot config file.
3. the box polls the TFTP/HTTP server for the config file.
4. the box reads the config file and executes the commands provided in the file
The URL may contain the same meta-character strings an Update Server URL, for example #m (mac-address)
The length of the URL in the DCHCP suboption is restricted to 127 characters.
The URL is polled in 5 second intervals.
The config file is read and executed by the update process in the usual way.
A 'creset' commmand as last command of the file will restart the box with the new configuration without writing any 'config' command options to the flash.
After a restart by the 'creset' commmand the boot-cfg URL is ignored.
After a restart by any other of the 'reset' commands or by a power cycle the boot-cfg URL is processed again.
On an Innovaphone DHCP-Server configuration of a boot-cfg URL and providing it to clients via suboption 249 must be explicitely enabled by
config add DHCP0 /boot-cfg
config write
config activate
Once enabled the URL may be entered under "IP4/ETH0/DHCP-Server/Boot Config URL" and is provided to all clients then.
If an Innovaphone DHCP-Client receivess a boot-cfg URL it is displayed under "IP4/ETH0/DHCP/Boot Config URL".
auto complete dtmf feature codes with '#' after 2 seconds
| Status | Closed |
| Id | 69561 |
Optional feature for phones, which are not able to send a '#', e.g. the iPhone.
They dial a feature like a cfu with a destination number and after two seconds, the feature code is automatically completed with a '#'.
PBX: HTTP request to initiate call for mobile phone
| Status | Closed |
| Id | 69570 |
To improve GSM client functionality
PBX: WSDL for SOAP API readable from box
| Status | Closed |
| Id | 69688 |
Simplifies Application development in some environments
SIP: Get display information from Call-Info header in register refresh response
| Status | Closed |
| Id | 69758 |
Get display information from Call-Info header in 200/OK
IP-DECT: Display update
| Status | Closed |
| Id | 69770 |
Now, the endpoint's display is updated if the registration forwards a display update.
Voicemail <pbx-query-obj> queries new properties
| Status | Closed |
| Id | 69781 |
"user_type": returns "1"(endpoint) or "2"(other object like gateway)
"pseudo_type": returns types alike "vm", "bc_conf", "trunk"
"pseudo_text": returns friendly texts alike "Voicemail", "Bc Conference", "Trunk Line"
V9 Hotfix 5 (90600.05)
Changes included in Version 9 hotfix5 Definition
SIP: Message decoding error
| Status | Closed |
| Id | 70116 |
Only concerns Message headers whose value starts and with quotes, but are not quoted.
E.g.
Referred-By: "Huvudnummer"<sip:400@abcdef.ghi;fnrid=1759>;from-tag=5decdf1a;to-tag=2515833546;org-cid="6afa95ede909d311906f00013e11cdb3@192.168.2.115"
Linux: VLAN GUI removed
| Status | Closed |
| Id | 70131 |
It is not necessary to configure a VLAN to Linux. Now the GUI page Linux VLAN is removed.
ip24/ ip6010/phone_orchid: in-band ring back tone generation into voice channel did not work
| Status | Closed |
| Id | 70146 |
sometimes a ring back tone must be sent in-band to an ISDN channel. the tone was generated but passed to the wrong destination.
SIP: Problems with CLEARMODE
| Status | Closed |
| Id | 70149 |
CLEARMODE was not offered as prefered codec outgoing INVITE.
Destination accepted G711a instead of CLEARMODE:
Offer:
v=0
o=- 18 1 IN IP4 172.16.66.77
s=-
c=IN IP4 172.16.66.77
t=0 0
m=audio 16422 RTP/AVP 4 18 8 0 96 97 101 13
a=rtpmap:96 G726-32/8000
a=rtpmap:97 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexa=yes
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20
Answer:
v=0
o=cp10 131833822144 131833822144 IN IP4 172.16.66.175
s=SIP Call
c=IN IP4 172.16.67.132
t=0 0
m=audio 33526 RTP/AVP 8 0 97
b=AS:75
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:97 CLEARMODE/8000/1
a=ptime:30
IP-DECT: Master radio list sync highlighting
| Status | Closed |
| Id | 70150 |
In the IP-DECT Master radio list the active sync highlighting is fixed, now.
SIP: No update of name presentation after transfer on QSIG interworking
| Status | Closed |
| Id | 70155 |
Fix for QSIG/SIP interworking.
Display name of callTransferComplete was not passed to SIP.
IP222/IP232: Handset volume with 90772 andHF4 too low
| Status | Closed |
| Id | 70226 |
This is a side effect of the echo canceller setting, volume in IP222/232 increased. IP241 uses old settings.
SIP: Cannot use SIP phone with myPBX
| Status | Closed |
| Id | 70252 |
Blind transfer (sending REFER) did not work since v9hotfix3.
SIP: Presence interoperability with ESTOS UC server
| Status | Closed |
| Id | 70257 |
Another fix for "Do not disturb".
| Status | Closed |
| Id | 70263 |
tftp was activated with alt-key, on on IP240 menu-key is used
Status:
platform_orchid.c
boot241.y
IP222/IP232: Sporadic Problem with the LCD Display after soft-reset
| Status | Closed |
| Id | 70269 |
It seem to be problematic to reset all orchid modules, e.g. the DMA module during software reset.
Now only USB and ENET modules are reset, the display gets also reset.
The display reset is released in the firmware.
Status:
start_orchid.S
platform_orchid.c
phone_orchid.cpp
boot222.y
boot232.y
boot241.y
IP241: Missing fkey icons
| Status | Closed |
| Id | 70353 |
Icons types list, face, mask-white and mask-black are available now.
PBX: Conference trap
| Status | Closed |
| Id | 70362 |
Some reworks of the PBX conference object. Fixes traps with call transfers of conference calls and conference calls to other PBX objects or mobility. Object update is also possible without call and chat clearing, now. Set maximum call number takes effect for maximum incoming calls, now.
IP241: Status icon for conditional call diversion
| Status | Closed |
| Id | 70363 |
Now having 2 different status icons for diversion.
One for unconditional and one for conditional forwarding.
IP241: Moving focus on screen "Call Diversion" activates diversion
| Status | Closed |
| Id | 70377 |
Using UP/DOWN/LEFT/RIGHT keys on the "Call Diversion" screen makes ON/OFF changing it state.
Webmedia: Recording of G.722 did not work
| Status | Closed |
| Id | 70382 |
Empty file was created.
myPBX: Closing a chat window sometimes causes a Java Script error
| Status | Closed |
| Id | 70439 |
client.js, line 2033
The Problem is a collision of closing the window and incoming messages for that window.
SIP: DNS resolving _stun._udp.xten.com did not work
| Status | Closed |
| Id | 70449 |
Because of unusual DNS response.
Target attribute of SRV answer records did not contain host domain name, but ip address.
<result val="0" title="SUCCESS">
<answer_rrs title="Answer Records">
<rr rr-type="33" rr-name="SRV" name="_stun._udp.xten.com" ttl="3407" priority="10" weight="0" port="3478" target="216.93.246.16"/>
<rr rr-type="33" rr-name="SRV" name="_stun._udp.xten.com" ttl="3407" priority="10" weight="0" port="3478" target="216.93.246.14"/>
</answer_rrs>
<ns_rrs title="Authoritative Nameserver Records">
</ns_rrs>
<ar_rrs title="Additional Records">
</ar_rrs>
</result>
H.323: Renegotiation to Fax did not work under some conditions
| Status | Closed |
| Id | 70462 |
Problem happened
- If switch to fax was done right after connect. This is typically done by IP Fax Servers
- If multiple signaling hops (e.g. multiple PBXs) were used
- If connect to a tone interface happened during dialing
myPBX: Norwegian translation updated
| Status | Closed |
| Id | 70477 |
Sent => Send
Logginn => Plogging
Permanent logginn => Forbli plogget
ethernet broadcasts bearing unicast IP packets with an arbitrary destination address were passed to the local IP-stack
| Status | Closed |
| Id | 70503 |
some load-balancing implementations send unicast IP packets (specially TCP-SYN) as ethernet broadcast packets.
Such packets must be silently discarded if the IP destination address is not the address of one of the local interfaces.
| Status | Closed |
| Id | 70569 |
ESC was taken as BS (backspace).
ESC must be handled as ESC when BS is no longer possible (empty input).
PBX: Presence update for PBX objects
| Status | Closed |
| Id | 70577 |
Now the presence is updated in myPBX for other PBX objects if the access is changed to allowed.
PBX: Conference presence info
| Status | Closed |
| Id | 70581 |
Now the broadcast conference object updates the presence info and is shown as callable in myPBX.
myPBX launcher: Hotkey does not work with some applications
| Status | Closed |
| Id | 70623 |
Try to get the selected phone number using WM_COPY before sending CTRL-C. This should help in some of the cases where the hotkey didn't work before.
SIP: Do not try to map local listen port with STUN when SIP/TCP is used
| Status | Closed |
| Id | 70654 |
Mapping of local listen port is only required fur SIP/UDP
IP222: Use R-key as BACKSPACE on edit fields
| Status | Closed |
| Id | 70664 |
Not ESC-key.
ESC-key is for leaving screen only.
Trap: When Dectmaster registers user at PBX using SIP protocol
| Status | Closed |
| Id | 70675 |
After closing regstration Dectmaster starts another call.
Call is rejected, but signaling enity is deleted before call object.
IP-DECT: Hanging call after failed call transfer
| Status | Closed |
| Id | 70756 |
The hanging call in dectradio is fixed which occurred if a user does an unattended call transfer and the call transfer fails.
SIP: Allow STUN to be used to map local media ports on SIP/TCP interfaces
| Status | Closed |
| Id | 70809 |
STUN cannot map signaling TCP port, but UDP media ports.
Status:
medialib.h
media.cpp
h323ch.h/cpp
sip.h/cpp
siptrans.h/cpp
IPVA, Keyboard Console, Credentials With Special Chars Couldn't Be Entered
| Status | Closed |
| Id | 70873 |
The Scancode table wasn't finished
SIP: Reduce memory footprint of SIP stack
| Status | Closed |
| Id | 70886 |
Free INVITE request buffer when receiving ACK. No need to keep any longer.
Delete INVITE client transaction when cancelling.
SIP: No route processing if neither Record-Route header nor Contact header is present
| Status | Closed |
| Id | 70971 |
Misleading trace message:
sip_call::process_routing(0xA8) Unsupported transport protocol: sip:user@domain.com;user=phone
IP6010: Wrong timer under high load
| Status | Closed |
| Id | 71001 |
-Clear IRQ in handle-interrupt after os_interrupt is too late, since IRQïs a enabled again and e.g. the timer irq is called again if a lower level IRQ like the enet occurs.
-The IRQ needs to be cleared in the serial-irq handler, in all case. After the serial-irq other interrupts are enabled.
Status:
ip6010.cpp
ip6010.h
SIP: Media negotiation problem
| Status | Closed |
| Id | 71009 |
Internal re-negotiation during early media on incoming SIP call.
Provide received offer to app again.
SIP: Offer CLEARMODE only if bearer capabilities are "Unrestricted Digital Information"
| Status | Closed |
| Id | 71162 |
On "Unrestricted Digital Information" only CLEARMODE is offered (no audio codecs).
On other bearer capabilities no CLEARMOE is offered (only audio codecs).
phone: dialog and presence subscriptions sometimes got lost after PBX restart when phone config was stored on PBX
| Status | Closed |
| Id | 71198 |
This happened specially when both "Store Phone Config" and "Discard Config on Phone" was checked in the user object because of a unsubscribe/subscribe race condition.\t
when editing a phone config template the dialing location inherited from a predecessor template was stored in the edited templat
| Status | Closed |
| Id | 71246 |
after a template has been edited unchanged information units inherited from predecessor templates must be removed from the edited template. this did not work for the dialing location and thus a later change in a predecessor template had no effect.
SIP: Endpoints behind NAT could not register at public PBX
| Status | Closed |
| Id | 71266 |
PBX must send response to the (public) IP address where request was received from.
Not to the (private) IP address in Via header or Contact-URI.
ipva, update to _platform_tracing.xsl
| Status | Closed |
| Id | 71274 |
ip6 missing
SIP: No media after accepting a waiting call
| Status | Closed |
| Id | 71288 |
Call waiting on a phone.
Going onhock while another call is waiting starts ringer.
After going offhook again the waiting call is accepted, but no media in both directions.
CX0 Wave-Encoding Not Working If Fact-Chunk Present In Header
| Status | Closed |
| Id | 71290 |
CX0 Wave-Encoding Not Working If Fact-Chunk Present In Header
DHCP Server Identifier could not be cleared via WEB interface
| Status | Closed |
| Id | 71305 |
When the field 'Server Identifier' was cleared and OK was pressed the just cleared value reappeared but after a reset 'Server Identifier' was clear.
IP241: Show ISDN display information in full length
| Status | Closed |
| Id | 71330 |
Use scrolling if required
NAT: Mapping to different internal UDP port did not work
| Status | Closed |
| Id | 71339 |
Configured destination port got lost after configuration
phone: send config to PBX only when the config was edited on phone
| Status | Closed |
| Id | 71387 |
A config from an older PBX may contain duplicate elements which are stripped by the phone. I such a stripped config is sent back to the PBX the PBX will return the old config again.
SIP: Send first NOTIFY(dialog) after sending 200/OK for SUBSCRIBE(dialog)
| Status | Closed |
| Id | 71413 |
NOTIFY(dialog) was sent before 200/OK for SUBSCRIBE(dialog)
\tRequest: SUBSCRIBE sip:150@192.168.200.14
\tStatus: 401 Unauthorized
\tRequest: SUBSCRIBE sip:150@192.168.200.14
\tStatus: 100 Trying
\tRequest: NOTIFY sip:158@192.168.200.214:5060
\tStatus: 200 OK
\tStatus: 481 Subscription Does Not Exist
SIP: Interop with Nortel CS1000 SIPLine GW
| Status | Closed |
| Id | 71426 |
Nortel sends 183/Progress with 'sendrecv' answer
followed by UPDATE with 'inactive' offer
followed by UPDATE with 'sendrecv' offer.
Innovaphone SIP stack remains in 'inactive' state.
H.323 Signaling Timeout event was generated for 'normal' RAS re-transmissions
| Status | Closed |
| Id | 71434 |
A single retransmission is normal under heavy load, so this is no reason for an event. Signaling Timeout events are now generated only if they cause a state change.
H.323 re-negotiation: Don't reuse media proposals if a select was already sent
| Status | Closed |
| Id | 71435 |
This fixes compatibility issues with SIP, especially when SRTP is used.
IP241: Show both parties of each call on screen "Pickup List"
| Status | Closed |
| Id | 71455 |
Currently only calling party infomation is displayed.
| Status | Closed |
| Id | 71469 |
The function assigned to the menu item (for example pickup) was executed first and then the key-function (for example dial).
PBX: Wrong web page when submitting an object and an error happens
| Status | Closed |
| Id | 71470 |
If for example a dupicate number is detected, the same web page should be displayed including the error message for the duplicate number. But not the same page was displayed but a page which could contain information not related to the object.
v8 to v9 upgrade problem with gateway registration names containing non-ASCII characters
| Status | Closed |
| Id | 71474 |
In general this was a problem with config line arguments seperated by ':'. This happened with the <number>:<name> argument within gateway definitions. The ':' was url-encoded and <name> interpreted as <number>
PBX: CFU was executed on PRESENCE_PUBLISH/SUBSCRIBE calls
| Status | Closed |
| Id | 71479 |
This was unexpected behaviour. You want to see the presence status of the configured user and not the presence status of the destination to which this user has configured a call forwarding
SIP: Interoperability with MX-ONE
| Status | Closed |
| Id | 71480 |
A semi-attended transfer fails if MX-ONE sends INVITE(Replaces)
instead of 200/OK when connecting a call.
Phone: Presence-Fkey did not always show presence set by myPBX
| Status | Closed |
| Id | 71487 |
Problem:
1) Set presence A with IP phone (fkey shows A)
2) Set presence B with myPBX (fkey shows B)
3) Delete presence with IP phone (fkey shows no presence)
Now Fkey shows presence B.
PBX: Tooltip on "PBX/Config/Log Calls" checkmark wrong
| Status | Closed |
| Id | 71507 |
It read "If not checked PBX calls are logged", should be "If checked PBX calls are logged"
| Status | Closed |
| Id | 71513 |
See http://wiki.innovaphone.com/index.php?title=Reference9:Concept_Fine_grained_function_hiding#More_Information
for more information
Media: Discard RTP packets from wrong source
| Status | Closed |
| Id | 71515 |
Packets arriving at RTP port must be discarded if the source if not the expected one.
To be save against DOS attack and for interop with Lync.
In some scenarios Lync starts sending RTP packets while having the call set to 'inactive'.
PBX: Blind transfer with consultation to BC-Conference failed
| Status | Closed |
| Id | 71540 |
The call was disconnected
H.323: No Alarm/Event should be generated by shutting down registration due to reset
| Status | Closed |
| Id | 71545 |
no event is sent to the application about this kind of unregistration
phone-orchid: micro-speaker loop of embedded test did not work
| Status | Closed |
| Id | 71566 |
Der Test war im Treiber nicht freigeschaltet
myPBX: Send existing invitation links to users that join the chat session later
| Status | Closed |
| Id | 71608 |
Users A and B have a chat session.
1. A starts application sharing or audio conference
-> A and B receive an invitation link
2. A adds a third user C to the chat and C accepts
-> C should also receive the invitation link
This worked only for audio conference but not for application sharing.
IP-DECT: Ring back tone after transfer
| Status | Closed |
| Id | 71610 |
If in-band ring back tone is sent and the call is transferred to a new destination with no in-band ring back tone, a local ring back tone must be played to the DECT handset. This is fixed now.
Ring Back tone missing after transfer when in-band tone was provided before but not after transfer
| Status | Closed |
| Id | 71612 |
The initial local ring back tone must be restarted when no data is received after transfer.
PBX: Send Name Identification with CLIR calls if "No CLIR on internal Calls"
| Status | Closed |
| Id | 71643 |
The feature "No CLIR on internal Calls" did not work completetly. The number was sent, but the Name Id was still suppressed
phone_orchid: Ring Back tone missing when Silence Compression is enabled
| Status | Closed |
| Id | 71670 |
When Silence Compression was enabled the tone generation was not triggered
IP232: Fix for touch handling
| Status | Closed |
| Id | 71674 |
Do not open touch keyboard on controls with CTRL_READONLY.
Do not open touch keyboard on controls without CTRL_ACTIVATE.
Using KEY_SHIFT has modified key to uppercase permanently.
Cursor positioning on text controls did not work.
Multi-line editor control was not displayed after hiding touch keyboard.
Hide overlay keyboard after next touched key.
Move and resize editor control when activating touch keyboard.
SIP: Trap on timer expiration during call release
| Status | Closed |
| Id | 71699 |
Media negotiation watchdog timer expired after final SIG_REL went to app.
But before app deleted the call object.
phone: display info provided by SETUP or CONNECT was ignored
| Status | Closed |
| Id | 71727 |
only the display info provided by an INFO event was handled
phone_orchid: continuous dialing tone stops after 10 seconds and is not restarted anymore
| Status | Closed |
| Id | 71753 |
the continuous dialing tone as used in most countries must be played as long as no digit is entered.
IPVA, Reset didn't work on VMware Player 4.0
| Status | Closed |
| Id | 71818 |
IPVA, Reset didn't work on VMware Player 4.0
IP241: Slow screen update when changing fkey type on fkey configuration screen
| Status | Closed |
| Id | 71831 |
Removing and adding config controls is very slow.
ip6010 - calling & dtmf tone timing did not work
| Status | Closed |
| Id | 71966 |
the tone time was calculated much too short
IP232: Redesign of touch keyboard
| Status | Closed |
| Id | 71968 |
Redesign of touch keyboard
IPVA, Trap After Failed FW Upload
| Status | Closed |
| Id | 71978 |
If a firmware upload failed with "wrong checksum" the box trapped occasionally, leaving the boot disk in an inconsistent state.
PBX Mobility: Potential Trap when initiating a call with myPBX or SOAP while another call is waiting
| Status | Closed |
| Id | 72637 |
A waiting means a call which was received while another call was active and the active call being disconnected and the waiting call not yet sent, or a call waiting for recall.
IP-DECT: Default frame size to 30ms
| Status | Closed |
| Id | 70140 |
Now the default frame size is 30ms.
SIP: Support for transparent message headers on transfer
| Status | Closed |
| Id | 70163 |
Interoperability with Telepo.
added support for USB tracing with Wireshark
| Status | Closed |
| Id | 70373 |
USB traces are now supported by the debug class and can be opened with Wireshark and a new innovaphone.dll.
SIP: Prefer P-Asserted-Identity with tel-URI
| Status | Closed |
| Id | 70417 |
Prefer P-Asserted-Identity with tel-URI over that with sip-URI.
In case there are two P-Asserted-Identity headers in INVITE.
Gateway: Forward Display Info received from ISDN Setup to H.323
| Status | Closed |
| Id | 70562 |
needed for compatibility with SecuGATE LI30
<pbx-getcallinfo> returning diversion reason
| Status | Closed |
| Id | 70697 |
as
<pbx-getcallinfo out-leg2-reason="..."/>
myPBX: Interface for IM provider
| Status | Closed |
| Id | 70867 |
Interface between the JavaScript client and the myPBX launcher.
Status:
IM to JS:
prepare_dial_name(value)
prepare_im(value)
ep_request(name,number)
JS to IM:
innovaphone_updateXml(xml)
innovaphone_sessionInfo(domain,name,number,cn,dn)
SIP: Diagnostics of transport error "SIP Overload"
| Status | Closed |
| Id | 70883 |
Limitation of buffer allocation
Status:
siptrans.cpp/h
sipmsg.h
IP-DECT: License restriction for OEM device
| Status | Closed |
| Id | 70887 |
A license restriction for an OEM device is added. It is not used in the IP1200.
SIP: Diagnostics of error "Registration expired"
| Status | Closed |
| Id | 70938 |
Missing registration refresh
Status:
sip.cpp/h
Possibility to clear the call list in mypbx
| Status | Closed |
| Id | 71202 |
A new button is now available in mypbx to clear the call list.
phone: LED mode of Join Group function key can be set both for idle and for active state
| Status | Closed |
| Id | 71247 |
sometimes the "not in group" state must be signaled as the exception
myPBX: Show implicit visibility in groups
| Status | Closed |
| Id | 71250 |
Active members of groups can see the presence and the calls of other group members. In order to make that clear to the user, now the visibility settings of myPBX show in what groups the user is visible.
IP-DECT: CSS changes for new OEM device
| Status | Closed |
| Id | 71282 |
For a new OEM device with a changed style some CSS classes are added.
PBX: New User property 'Do not Disturb'
| Status | Closed |
| Id | 71439 |
No calls are sent to the user if set.
Can be set by 'External UC' applications (e.g. Estos Procall)
PBX: Support up to six devices for a user
| Status | Closed |
| Id | 71506 |
Sometimes 4 devices are to little.
A little bit of cleanup on the user interface was done as well with this change
phone: Mic Off/On controllable via Soap:UserRc(<call>,14/15)
| Status | Closed |
| Id | 71721 |
To allow Soap app's control of the mute key
SoftwarePhone: New Tray Icon
| Status | Closed |
| Id | 71969 |
The SoftwarePhone has got a new tray icon differs from the myPbx one, now.
V9 Hotfix 6 (90600.06)
Changes included in Version 9 hotfix6 Definition
myPBX: Make more obvious that closing the chat window terminates the chat
| Status | Closed |
| Id | 71893 |
Display warning message when the user tries to close the chat window.
Show disconnect icon instead of close icon in the window.
IPVA, Optimize Flash-Related I/O Operations
| Status | Closed |
| Id | 71954 |
Specifically to reduce boot-time. Utilizing 64KB I/O accesses where possible.
H.323 RAS: Registration with authentication to account without authentication failed in a strange way
| Status | Closed |
| Id | 71958 |
Information was missing that no authentication was supported, so it was continued to retry instead of giving up right away
TCP: Roundtrip measurement wrong in case of packet loss
| Status | Closed |
| Id | 71985 |
In case of packet loss, way to high round trip values were measured. If the packet-loss was to high, this could result in a constantly increasing re-transmission timeout value.
SIP: Workaround for delays in media negotiation
| Status | Closed |
| Id | 72189 |
Workaround for delays in media negotiation caused by delayed Admission on H.323 calls.
SIP: Trap on IP-DECT when re-configuring PBX link
| Status | Closed |
| Id | 72190 |
85:2195:425:7 - REG_PRI.4 default(8102be48): serial_timeout
85:2195:425:7 - Assertion failed line 748 in common/os/os.cpp, object deleted
Status:
Merged to 09-80500
IP232: Input of capital Latin Extended-A
| Status | Closed |
| Id | 72226 |
Input of capital Latin Extended-A
myPBX: Queue chat messages for im calls that are not yet alerting
| Status | Closed |
| Id | 72237 |
The following fix did not work across PBXes:
#71608: myPBX: Send existing invitation links to users that join the chat session later
The problem was that messages were only queued in alerting state. When the call is made across PBXes this state isn't reached, yet.
Scheduling improved to avoid processes not being scheduled during long flashman operations
| Status | Closed |
| Id | 72243 |
In version 7 it could happen, that IP and other processes were not scheduled any more during periods of long flashman operations (e.g. bootcode update or reorganizing flash).
In version 8 and higher there was already a fix for this problem, but this included special handling of the flashman priority level, which was not a good solution even if it worked.
IP232: Touch keyboard key highlighting
| Status | Closed |
| Id | 72268 |
Added highlighting for key like SHIFT, BACKSPACE, etc.
PBX: H.323 Names in some places (e.g. Waiting Queue Maps) containing non-ASCII charcters could break the user interface
| Status | Closed |
| Id | 72277 |
This happened esspecially for Names as destinations for Waiting Queue maps, but it could happen at other places as well.
SIP: Cleanup failed (resources leaking)
| Status | Closed |
| Id | 72284 |
Call and channel objects were not freed sometimes
when INVITE was followed by CANCEL very fast.
IP241: Disable Link LED not working
| Status | Closed |
| Id | 72338 |
Timing problem with access to the paged MDIO registers Status: orchid_drv.cpp
Trap: When accessing web interface
| Status | Closed |
| Id | 72362 |
Seen once.
Not reproducable yet.
Ip6010 DSP Disconnect timeout after fax session
| Status | Closed |
| Id | 72403 |
Debugs added
* enable with http://addr/debug.xml DSP trace and DSP control message trace to printout all packets to the DSP with a descriptive string. That allows to analyse the message flow to the DSP after a trap.
* for further testing old fax disconnect procedure can be enabled with http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl with "t38 skip fax close".
Status:
ac_dsp3.cpp
ac_dsp3.h
ac_491.h
dsp.xsl
IP2x2: Fixed fkey handling of builtin test
| Status | Closed |
| Id | 72442 |
Fkey test did not work for touchkeys, key symbols were misplaced on screen
myPBX: Improve presentation of chat conferences
| Status | Closed |
| Id | 72444 |
Chat conferences should not be displayed as multiple calls but as a single item in the main window.
IP232: Parameters for touch sensor fixed for latest sensor
| Status | Closed |
| Id | 72455 |
latest touch was treated like the very old version that needed other parameters.
Old touch is not supported any more ( only 3 were build)
Status:
edt_touch.cpp
IP241,IP222,IP232: Make password configuration more convenient
| Status | Closed |
| Id | 72456 |
When entering a passwort on the phone's ui content is displayed as "****".
Show last enered character in plain text.
Ip6010 DSP Allow coder change from T38 to voice and back to T38
| Status | Closed |
| Id | 72457 |
..
Status:
ac_dsp3.cpp
| Status | Closed |
| Id | 72477 |
Center key should enter selected menu item.
Not leaving the current menu screen.
ESC key can be used to leave current menu screen.
Phone app will ask whether to save changes.
myPBX: Name and Number Display not correct on IM sessions across PBXs
| Status | Closed |
| Id | 72491 |
for local sessions (same PBX) the number and the Long Name was displayed for the remote party and this should be the same for remote session.
Sometimes the number was missing, sometimes the Name was displayed instead of Long Name.
The number was not adjusted correctly if different Nodes were involved
SoftwarePhone: Trap and media channel after conference
| Status | Closed |
| Id | 72511 |
If a call is disconnected during conference mode, the SoftwarePhone traps or there is no voice channel to the remaining party. This is fixed now.
HTTP: Accept authentication with both Latin1 and UTF-8 coding
| Status | Closed |
| Id | 72512 |
The server does not know what encoding is used in Basic- or Digest-Authentication. Therefore both encodings shall be tried.
ip28 codec hang when TEL port configured to '600Ohms'
| Status | Closed |
| Id | 72521 |
this option experienced problems in the past and should therefore not be selected. If selected the firmware will switch to default CTR21.
PBX: Cause for release/reject of a IM session not signaled
| Status | Closed |
| Id | 72533 |
Better feedback to the User
myPBX: New icons for unknown phone and im status
| Status | Closed |
| Id | 72536 |
The difference between open, closed and unknown was not understood well.
SoftwarePhone: Dialing outbound calls temporarily failed
| Status | Closed |
| Id | 72628 |
When there was a held call and a consultation call and the consultation call was released by the remote peer the SoftwarePhone did not accept further outbound calls until the held call had been released. This is fixed now.
IP232: Fix for touch keyboard handling
| Status | Closed |
| Id | 72651 |
When editing call diversions, one of the on/off controls may render across touch keyboard.
Content of multi-line-edit-control was mis-placed when touch-keyboard was activated.
Display of first matching directory entriy on indirect dialing screen.
PBX Config Templates: When editing config Templates wrong inherited values are displayed
| Status | Closed |
| Id | 72655 |
A config Template inherits only from the 4 other Templates configured in it directly, whereas any other object inherits from the Config Template configured with it and the four other Templates this is refering to.
ip241 - monitor mode (handset + speaker) did not work in V9hotfix5
| Status | Closed |
| Id | 72702 |
monitor mode (handset + speaker) was displayed but the speaker was mute
update - scfg command could hang when the HTTP session was broken or prematurely closed by the server
| Status | Closed |
| Id | 72708 |
in consequence update script processing was stopped until reboot
Web-UI: Wider input fields for domain names
| Status | Closed |
| Id | 72723 |
When configuring
- Remote Domain
- Local Domain
- Proxy
- STUN Server
on a Gateway interface there's need for more space.
Trap: When Dectmaster registers user at PBX using SIP protocol
| Status | Closed |
| Id | 72729 |
When Dectmaster registers user at PBX using SIP protocol
SIP: Add payload type for RTP-DTMF in case of media-relay
| Status | Closed |
| Id | 72732 |
Add payload type for RTP-DTMF (telephone-event) to SDP offer in case of media-relay.
Should help on Mobility scenarios.
PBX: Called Name displayed when calling an object with forking was wrong
| Status | Closed |
| Id | 72735 |
The name of the forking destination was displayed instead of the name of the called object
IP4001/IP6000 -UART driver did not work
| Status | Closed |
| Id | 72743 |
the hardware requires the registers to be written 32-bit wise which was not respected by a fix one year before.
PBX: No Audio if call thru Waiting Queue DTMF destination, was transfered to BC-Conf
| Status | Closed |
| Id | 72746 |
Problem caused by call state management error in PBX for calls connected without alert if alert was received later
IP241,IP222,IP232: Wrong call state displayed
| Status | Closed |
| Id | 72748 |
"Destination reached" displayed instead of "subscriber busy" if consultation call was rejected.
PBX Waiting: Ringback missing when using DTMF to dial from one WQ to another which is alerting
| Status | Closed |
| Id | 72766 |
This was a collateral damage for a fix for Waiting Queue announcements from a Boolean Object
IP222,IP232: Long key press on numeric block while being offhook does not open directory search
| Status | Closed |
| Id | 72831 |
Long key press during offhook is expected to open directory search.
| Status | Closed |
| Id | 72833 |
Re-design
PBX: Profiles/Access got lost when writing an User Object with SOAP
| Status | Closed |
| Id | 72849 |
The respective tags were not allowed
IP241,IP222,IP232: Support for unicode 0x308 (diaeresis/umlaut)
| Status | Closed |
| Id | 72859 |
When ,, are encoded as a,o,u followed by diaeresis from unicode block 'Combining Diacritical Marks' only a,o,u where displayed.
Now ,, are displayed.
IP241,IP222,IP232: Support for hebrew and arabic presence notes
| Status | Closed |
| Id | 72905 |
Hebrew and arabic text passages are rendered frmo right to left.
PBX: Conference no media
| Status | Closed |
| Id | 72915 |
There is sometimes no media for a conference member. This is fixed, now.
SIP: Memory leak during transfer
| Status | Closed |
| Id | 73003 |
Occured on internal testing only (002-conf-with-bcast.xml)
IP241,IP222,IP232: Update for CFU indication on idle screen
| Status | Closed |
| Id | 73013 |
Not displaying "cfu:241" on header of idle screen
but "Diverted to 241"
or "Umgeleitet zu 241"
or "Renvoy sur 241"
etc.
myPBX: Simplify adding multiple users to a chat session
| Status | Closed |
| Id | 73016 |
Do not unselect the chat session, after a person has been added. Replace "start chat" buttons by "add to chat" buttons when a chat session is selected.
RTP-DTMF: Digit may get lost during media re-negotiation
| Status | Closed |
| Id | 73037 |
Receiver starts handling on END event.
Sender may stop sending before END event was sent.
myPBX: Do not show bubble for outgoing chat calls and messages
| Status | Closed |
| Id | 73039 |
Notifications are only needed for incoming calls and messages.
myPBX: Skript errors in chat window
| Status | Closed |
| Id | 73040 |
Closing the chat window caused different script errors in Internet Explorers and Firefox.
LDAP replication from v7 could stop
| Status | Closed |
| Id | 73052 |
LDAP replication from v7 could stop
SIP: Don't write SRTP key into T.38 part of media description
| Status | Closed |
| Id | 73112 |
v=0
o=- 1295 1 IN IP4 10.120.55.3
s=-
c=IN IP4 10.120.55.6
t=0 0
m=audio 20026 RTP/SAVP 8 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -
a=inactive
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9xHxSg836505XOXwdIHfQ8Cm2ZYezNPpjvHNPCvb
m=image 0 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:9xHxSg836505XOXwdIHfQ8Cm2ZYezNPpjvHNPCvb
PBX External UC: Support of multisite setup
| Status | Closed |
| Id | 73115 |
This requires all calls to the External UC to be routed to one interface. Before this fix the calls were routed within the local PBX only.
Now a Name (not Long Name as before) has to be configured for the External UC interface.
IP241,IP222,IP232: Show number of missed calls in status bar
| Status | Closed |
| Id | 73140 |
New extended symbol is more eye-catching.
PBX: Called party number wrong at original called phone with forking
| Status | Closed |
| Id | 73145 |
If the forking destination and the original called phone are assigned to different nodes
Media Negotiation: v9 XPARENT not compatible to v8 or earlier XPARENT
| Status | Closed |
| Id | 73153 |
In v9 a dynamic payload type is used for XPARENT to be compatible to SIP, whereas in v8 an earlier payload type 0 was used. Within the media negotiation this should be detected and switched back to payload type 0.
Phones: Update of phone text data
| Status | Closed |
| Id | 73211 |
Some translations missing (Polski,Eesti).
Support for language typical letters (Polski).
Some abbreviation removed (for color phones).
PBX: Conference GUI update with IE
| Status | Closed |
| Id | 73256 |
The PBX broadcast conference object window is not closed if IE is used. This fixed now.
SIP: support for "Content-Type: multipart/alternative"
| Status | Closed |
| Id | 73260 |
Lync interoperability
IP222/IP232: voice quality in handset mode
| Status | Closed |
| Id | 73263 |
- Neuer DSP code von Audiocodes ( patch Version 660 02 mit AMR_WB ).
* Kein StrongEC mehr ntig
* Codec PCM Bus luft mit 16KHz
* Wideband Codecs zur Zeit disabled
* Konferenz erfordert noch Anpassung im Treiber
Status:
ac_494e.cpp
ac_codec3.cpp
ac_codec3.h
ac_dsp3.cpp
ac_dsp3.h
phone_orchid.mak
IP2x2: Keyboard handling on call reroute
| Status | Closed |
| Id | 73268 |
Redial key launches "Transfer" screen.
After entering the destination number, Redial key should cause execution of call reroute.
PBX Waiting: When using a waiting queue for outgoing dialing, the announcement was cut off at the beginning
| Status | Closed |
| Id | 73274 |
This was because the announcement was started already when the call was sent out and not when the call was connected
myPBX: Allow spaces in last LDAP attribute
| Status | Closed |
| Id | 73276 |
Currently "j doe innovaphone" finds all internal users starting with j. It should only find internal users that start with "j doe innovaphone".
IP222/IP232: hand receiver volume too high
| Status | Closed |
| Id | 73306 |
Die IP222/IP232 wird gelegentlich als zu laut empfunden.
Die Prozentanzeige mit den krummen Balken sieht "komisch" aus.
0% (Slider ganz links) wre gut.
Mehr Werte wren gut.
Status:
ac_codec3.cpp
ac_codec3.h
IP232: Cursor positioning when touching ip address control
| Status | Closed |
| Id | 73407 |
Make sure input cursor is at right side of touched octet.
SIP: Configuration of SIP response code mappings did not work on IPxx10 and IPVA
| Status | Closed |
| Id | 73453 |
PBX: Reporting licenses counting could be wrong if config templates were used
| Status | Closed |
| Id | 73466 |
It could happen that a reporting license was used up by a config template object, if reporting was checked and other config templates where referenced
myPBX: Allow overriding implicit allows
| Status | Closed |
| Id | 73471 |
It was not possible to define an allow with the same name. This is needed to override allows that are inherited from templates or group memberships.
IP241,IP222,IP232: Trap when scrolling down long call list
| Status | Closed |
| Id | 73494 |
Out of memory trap when scrolling down long call list.
Trap: When using Diagnostics/Ping
| Status | Closed |
| Id | 73536 |
When leaving Diagnostics/Ping or Diagnostics/Traceroute web screen a trap may occur.
| Status | Closed |
| Id | 73567 |
Internet explorer buggy concerning url encoding of href attributes
IPVA, crash-dump without backtrace, if tracing was switched off internally
| Status | Closed |
| Id | 73574 |
IPVA, crash-dump without backtrace, if tracing was switched off internally
ISDN: Do not provide tones and no signaling of inband info for unrestricted digital information calls
| Status | Closed |
| Id | 73603 |
There is equipment, which is doing unrestricted digital information ISDN calls, which gets confused if there is an ALERT message indicating inband tones (ringback).
IP241,IP222,IP232: Symbol "new messages" and symbol "headset" do overlap in status bar
| Status | Closed |
| Id | 73608 |
Moved "headset" symbol to right side of status bar.
IP-DECT: DECT endpoint update for multicast identities
| Status | Closed |
| Id | 73613 |
The update information to DECT is changed if the user logs out a handset. This is used to update the multicast identities in some OEM handsets.
Ip6010 DSP Allow calls to start with T.38
| Status | Closed |
| Id | 73626 |
Needed for some T38 fax applications Status: ac_dsp3.cpp
IP241,IP222,IP232: Missing "Recall possible" display text on call completion
| Status | Closed |
| Id | 73628 |
Missing display text on 'recall possible' notification.
SIP: Media negotiation problem on outbound call from BC Conference object
| Status | Closed |
| Id | 73652 |
INVITE was sent without SDP offer.
test/9.00/pbx/conf/SIP/002-conf-with-bcast.xml
IP241,IP222,IP232: Extra digits dialed when using RIGHT key on screnn 'indrect dialling'
| Status | Closed |
| Id | 73694 |
'123123' instead of '123' was dialed.
ISDN Trunk: Transfer to ISDN Trunk with TONE interface failed
| Status | Closed |
| Id | 73695 |
There was not media after the transfer
H.323: Renegotiation to SRTP after dialtone failed
| Status | Closed |
| Id | 73698 |
Happens with Escape Dialtones configured in PBX and calls with SRTP enabled
Option added to wait for the DSP to disconnect fax, needed only for some fax devices.
| Status | Closed |
| Id | 73748 |
Use the prodedure of
Mantis 67821: ip22/24/28/302/305: DSP affected by certain fax tones
only if the option /t38-wait-fax-close set.
This avoids traps caused by disconnect timeouts if the DSP response is not received in time.
Status:
ac_dsp3.cpp
ac_dsp3.h
dsp.xsl
SIP: Using wrong remote port when registering
| Status | Closed |
| Id | 73784 |
Only affects IP-DECT when handset is switched OFF and ON and if the SIP runs on non-standard port.
IP232: Memory leak in display rendering
| Status | Closed |
| Id | 73864 |
Memory leak in display rendering
IP232: Presence fkey shows current activity, but not current note
| Status | Closed |
| Id | 71611 |
Show current note in second line of fkey label.
myPBX: Cause codes for instant messaging
| Status | Closed |
| Id | 71878 |
Display causes why a chat session has been terminated.
For example: Unassigned number, rejected, no response...
myPBX: Make call state icons clickable
| Status | Closed |
| Id | 71886 |
Clicking a call state icon should have the same effect as clicking the text.
myPBX: Notification on disconnected chat sessions
| Status | Closed |
| Id | 71918 |
Display a text that the chat has been terminated.
Debug information on assertion
| Status | Closed |
| Id | 71961 |
More debug information on default event handler.
myPBX: Display reason why a browser is not supported
| Status | Closed |
| Id | 71972 |
Currently it only says "Browser version not supported".
It should also tell what feature isn't supported:
- XmlHttpRequest
- WebStorage
- PostMessage
myPBX: Forward invitation links to incoming chat calls that are added to a chat conference
| Status | Closed |
| Id | 72280 |
The same way like the invitation links (for application sharing and audio conferences) are forwarded to additional outgoing chat calls.
myPBX: Display invitation links in the chat window of the organizer
| Status | Closed |
| Id | 72282 |
The organizer of an application sharing session or a audio conference wants to see all corresponding links in the chat window.
SoftwarePhone: Password encryption tool
| Status | Closed |
| Id | 72431 |
A password encryption tool was added.
SoftwarePhone: Crash dump
| Status | Closed |
| Id | 72433 |
Now, the SoftwarePhone writes a crash dump file.
myPBX: Display shortened links in chat messages
| Status | Closed |
| Id | 72445 |
Shorten the displayed part of the URL in order to avoid text overflow.
Cut overflowing texts at the right boundary of the chat box.
PBX: New presence access flag 'online'
| Status | Closed |
| Id | 72538 |
With this flag a user allows access to his online status
myPBX: Configure visibility of online state and presence separately
| Status | Closed |
| Id | 72632 |
Phone status and IM status are now called online status.
Activity and note are now called presence.
The visibility of both is now configured separately.
PBX Mobility: Better support for Opticaller
| Status | Closed |
| Id | 72701 |
HTTP request to initiate call for mobile phone
myPBX: New translations
| Status | Closed |
| Id | 72768 |
For the other v9hotfix6 items regarding myPBX.
Support for more automated Fax Tests
| Status | Closed |
| Id | 72826 |
Switching from Audio to Fax to Audio to Fax. Special case which could happen if switch to Fax happened before connect.
myPBX: Highlight incoming alerting calls
| Status | Closed |
| Id | 73093 |
Highlight the alerting icon using an orange border.
SIP: Interworking with KIRK Wireless Server 300
| Status | Closed |
| Id | 73110 |
If "KIRK Wireless Server 300 PCS10__ r3327"
calls into PBX and is connected with Voicemail,
Voicemail may send re-INVITE with SRTP key.
Instead of accepting or ignoring the SRTP option,
KIRK Wireless Server 300 rejects the whole SDP offer.
Now we retry the re-INVITE w/o offering SRTP key.
PBX: Forward original received ISDN display element to picking up or forwarded call
| Status | Closed |
| Id | 73278 |
In the display element from ISDN there could be vital information from equipment like crypto gateways. This should be available also if the call was picked or forwarded.
SoftwarePhone: Support for Jabra SPEAK 410 USB with product id 0x0410
| Status | Closed |
| Id | 73424 |
Now, the SoftwarePhone supports Jabra SPEAK 410 USB with product id 0x0410.
SIP: Overriding cause code mappings did not work
| Status | Closed |
| Id | 73477 |
Mapping of SIP response codes into Q.931 cause codes could not be overwritten.
Linux: Empty file check
| Status | Closed |
| Id | 73554 |
Now, an empty file is not accepted as a successful file upload.
V9 Hotfix 7 (90600.08)
Changes included in Version 9 hotfix7 Definition
Ip6010 DSP Allow coder change from T38 to voice and back to T38 with local DSP
| Status | Closed |
| Id | 73193 |
..
Status:
ac_dsp3.cpp
ISDN: Enable fax detection only after connect
| Status | Closed |
| Id | 73213 |
On ISDN networks it can happen that the Connect message is delayed. This way fax tones are forwarded to the caller before the caller has received this Connect.
This way a renegotiation on voip to fax could happen before the connect, which is not supported by sip.
Phones: Update of phone text data
| Status | Closed |
| Id | 73775 |
External input
SIP: Handling of collision of transfer and release
| Status | Closed |
| Id | 73936 |
If one end releases a call while the other initiates an attended transfer, a "ghost call" may remain.
Resource leak.
IP241,IP222,IP232: Limitation of background image size
| Status | Closed |
| Id | 73957 |
Limit is 512 kBytes now.
To keep phone from out of memory trap.
H.323: Renegotiation after PBX Node dialtone failed sometimes
| Status | Closed |
| Id | 74020 |
This depended on timing. This happened if the call was sent after the dialtone to a master over a slow link.
| Status | Closed |
| Id | 74046 |
Touching header bar and moving vertically down opens the phone menu.
No need to use center key of 4-way control.
Dect User Gui: AD-replicated objects weren't listed under certain conditions
| Status | Closed |
| Id | 74051 |
Deleted object thate were re-replicated werent't visible in the Dect user GUI. A mandatory Ldap attribute was missing under such circumstances.
IP222: Handset Microphone level too low
| Status | Closed |
| Id | 74073 |
Handset micro increased by 6db
Handsfree micro unchanged
Status:
ac_codec3.cpp
IP222,IP232: Language configuration not saved
| Status | Closed |
| Id | 74087 |
Language can be selected on phone menu, but change is neither saved nor applied.
SIP: Don't take NOTIFY(application/simple-message-summary) with "Do-Not-Disturb: no" as FTY_MWI_DEACTIVATE
| Status | Closed |
| Id | 74162 |
Interop with Nortel CS1k PBX
Phones: Fine grained function hiding did not disable Fkey configuration
| Status | Closed |
| Id | 74174 |
Value 0x00000800 (PHONE_HIDE_US_FUNCTION_KEYS) hides function keys entry from user-list,
but fkeys can still be (re)configured by long fkey press.
For more information see:
http://wiki.innovaphone.com/index.php?title=Reference9:Phone/Protect
myPBX: Allow asterisk as a wildcard in LDAP search strings
| Status | Closed |
| Id | 74200 |
As it is done on our phones.
IP232: Pixel noise with some color values
| Status | Closed |
| Id | 74221 |
- changed to lower pixel clock Status: orchid_lcd.cpp
PBX: Conference to VM
| Status | Closed |
| Id | 74222 |
Now a call to the VM as a conference member works again.
RTP-DTMF: Digit may get lost during media re-negotiation
| Status | Closed |
| Id | 74232 |
Receiver starts handling on END event.
Sender may stop sending before END event was sent.
No media for calls from SIP provider forwarded back to SIP provider in case of NAT and MediaRelay
| Status | Closed |
| Id | 74290 |
NAT traversal depends on a packet being sent from inside the NAT to outside, to fix the RTP destination of the outside endpoint. This does not happen if both endpoints are outside.
Dummy packets are sent from the Media Relay function in this case to achieve this.
ISDN: Send HLC with mobility calls
| Status | Closed |
| Id | 74296 |
Some ISDN networks refuse the forwarding of a call to a mobile network if no HLC (High Layer Compatibility) Information Element indicating Telephony is included in the call.
IP241,IP222,IP232: Improved phone screen rendering
| Status | Closed |
| Id | 74301 |
Reduce map copy operations.
SIP: Accept INVITE's from alternative proxy
| Status | Closed |
| Id | 74310 |
If alternative proxy is configured, calls should accepted from alternative proxy,
even if primary proxy is alive and kicking.
Flash Directory: Substring match could miss results
| Status | Closed |
| Id | 74320 |
Searching by substring worked case sensitive in some cases, instead of functioning case insensitive.
PBX: CF at Gateway Type objects - additional dialed digits should be added to the destination
| Status | Closed |
| Id | 74348 |
This way a CFNR at a trunk object can be used to reroute the call to another trunk.
Gateway: Deleting of routes could result in duplication of routes
| Status | Closed |
| Id | 74356 |
This happened if an interface registration was disabled, for which automatic routes have been generated and then a route was deleted. The last route was duplicated.
IP241/222/232: Monitormode (Lauthren): Level too low
| Status | Closed |
| Id | 74362 |
Use independent analog codec channels for speaker and headset receiver.
Speaker volume in monitor mode is configured as in handsfree mode.
To change to speaker level in monitor mode the gain of the speaker can be configured with
config change AC-DSP0 RINGER /DualOutputModeGain level
config activate
level is from 0..63 0
0 -> -32dm
32 -> 0db
63 -> 31db
Status:
ac_codec3.cpp
ac_codec3.h
| Status | Closed |
| Id | 74367 |
Could not activate first builtin background image from phone menu if external background image is currently active.
IP241,IP222,IP232: Show multiple diversions on incoming calls
| Status | Closed |
| Id | 74371 |
If a call was diverted more than once, the phone shows first diverting party (original called number) and last diverting party.
(Not only the last diverting party)
myPBX launcher: Unhandled exception when accessing browser object
| Status | Closed |
| Id | 74380 |
The myPBX launcher could crash when accessing the browser document because of an unhandled exception.
IP232: Pickup fkey displays no information in active state
| Status | Closed |
| Id | 74392 |
Pickup fkey displays no information in active state
NAT: Permanent UDP forwarding did not work
| Status | Closed |
| Id | 74549 |
Permanent UDP forwarding was unusable after first forwarded datagram.
PBX Waiting: If used as outgoing dialing device, coder list sent with the call was not correct
| Status | Closed |
| Id | 74593 |
Should be based on the coders paramter in the configured URL
Phone_orchid: Residual echo in handset ( sporadic)
| Status | Closed |
| Id | 74671 |
Echo canceller NLP sensitivity Mode changed to 2 Status: ac_dsp3.cpp
IP222,IP232: Function key "Hotdesk" did not work
| Status | Closed |
| Id | 74723 |
Function key "Hotdesk" did not work
IP232: Fix for long touch on fkey
| Status | Closed |
| Id | 74743 |
Opens fkey config, but mist not toggle control on touch-off.
Web-UI: Configuration of "Sync Server" does not return with HTML page
| Status | Closed |
| Id | 74778 |
Configuration of "Sync Server" does not return with HTML page
| Status | Closed |
| Id | 74792 |
Even after confirming to save
IPxx10: Flashdir Segments Default to 129(was 51)
| Status | Closed |
| Id | 74826 |
IPxx10: Flashdir Segments Default to 129(was 51)
Interop: Don't add an error log "SRTCP authentication failed" on non-RTCP packets
| Status | Closed |
| Id | 74903 |
Microsoft Lync send invalid RTP and RTCP packets at the beginning of each call.
In case of encrypted media, decryption of these packets fails.
phone_orchid: ethernet statics counters sometimes displayed as negative numbers
| Status | Closed |
| Id | 74913 |
the unsigned counters were printed as signed integers
IP241,IP222,IP232: Volume control not displayed when already at maximum
| Status | Closed |
| Id | 74927 |
Volume control is not displayed when trying to increase while being at maximum.
IP6000 crypto driver: Trap when buffers are depleted
| Status | Closed |
| Id | 74935 |
Avoid the trap and log an Event when the buffers are depleted.
IP241,IP222,IP232: Mute status indication does not work
| Status | Closed |
| Id | 74964 |
"MICROPHONE OFF" is displayed on status bar now.
SIP: Rare problem with blind transfer
| Status | Closed |
| Id | 74991 |
callIdentity must be zero when giving ctInitiate to PBX on blind transfer.
TLS: Flow control for incoming data
| Status | Closed |
| Id | 75004 |
The TLS socket has to wait for the application to process incoming data before sending the next RECV.
H.323: Renegotiating to Fax was rejected, if PCM switch and local networks were used
| Status | Closed |
| Id | 75006 |
If the side which initiated a switch to T.38 has configured PCM and the media address was classified as local due to local network configuration, the T.38 was rejected.
SIP: Interworking of divertingLegInformation1 improved
| Status | Closed |
| Id | 75025 |
Honour subscriptionOption
IP222,IP232: Changes not saved in some cases
| Status | Closed |
| Id | 75088 |
Some config screens did not write changed settings directly after "Save Changes" dialog.
If menu was left with DISC key (instead of ESC key) the changes have been discarded.
Also affects other phones: IP241,IP240,IP230,IP110
Changes are saved immediately when leaving the current screen.
Not when leaving "User Settings" or "Phone Setting" screen.
PBX Waiting: Diverting leg1 info not correct when diverting to a Waiting Queue
| Status | Closed |
| Id | 75091 |
If Waiting queue not defined in root node and escapes are used
PBX: Adjusting received leg1 info wrong
| Status | Closed |
| Id | 75097 |
In context of nodes with escapes
VM: <pbx-upd-obj type="cfu"..> without effect when invoked multiple times
| Status | Closed |
| Id | 75121 |
Statement <pbx-upd-obj type="cfu"..> failed to work properly after being used for diversion manipulation multiple times within a single script session.
SIP: No interworking of "Q.931 CALL PROCEEDING" into "183 Session Progress"
| Status | Closed |
| Id | 75140 |
Causes trouble on other vendor PBX's.
(Aastra 5000, Advoco/Arcstel, Nortel SESM)
IP28 Watchdog leads to endless interrupt and trap
| Status | Closed |
| Id | 75186 |
.
IP800/6000: Problem mit FAX und CNG detect: Option to disable CNG detect added
| Status | Closed |
| Id | 75278 |
config change AC-DSP0 /t38-cng-detect-disable 1
config write
config activate
Status:
ac_dsp3.cpp/h
ac_fax3.cpp/h
SIP: Decrement Max-Forwards and interwork to H.323
| Status | Closed |
| Id | 75288 |
Instead of sending always "Max-Forwards: 70"
H.323: Renegotiation to T.38 did not work for slowstart call thru media-relay to efc endpoint
| Status | Closed |
| Id | 75305 |
A common scenario when this happend is an XCAPI (slowstart) calling thru a PBX with media-relay configured to an ISDN interface
Fix for MIPS counter
| Status | Closed |
| Id | 75310 |
MIPS counter was incorrect
PBX: Allow 'Max Calls' configuration of 0
| Status | Closed |
| Id | 75362 |
So that no call at all is sent to the object but a CFB (if configured) is executed
SIP: Send "305 Use Proxy" if INVITE is received from unexpected source
| Status | Closed |
| Id | 75380 |
Applies to registered interfaces only (e.g. phones).
TLS: Possible trap when restoring sessions
| Status | Closed |
| Id | 75394 |
Only in v9, depending on the certificate.
PBX Exec Object: Leg2 information send to secretary not adjusted correctly for nodes
| Status | Closed |
| Id | 75431 |
The leg2 information is used to display at the secretary the number of the exec which was called. This number was not correct if nodes with escapes were used
Gateway Interface Maps: Should be applied to leg1 info also
| Status | Closed |
| Id | 75437 |
The same rules used for a Calling Party Number are applied to leg1 info
SoftwarePhone: Timer tick resolution
| Status | Closed |
| Id | 75447 |
The timer tick resolution is increased. This fixes the call RTCP calculation transmitted to the PBX.
IP241,IP222,IP232: Indirect dialing of names not possible
| Status | Closed |
| Id | 75451 |
Could not dial a name from 'indirect dialing' screen
by moving the focus to input field and going offhook.
(Only fkey worked)
PBX: CFNR Loop check detected loops that weren't
| Status | Closed |
| Id | 75465 |
Esspecially a CFNR at a Waiting Queue was not executed if the destination of the CFNR has transfered the call to the Waiting Queue
Gateway: Event "No Media Data Received" was created for each SWITCH-PCM call
| Status | Closed |
| Id | 75483 |
There should be no event in this case, this is normal.
PBX: Conference id prefix/suffix configuration
| Status | Closed |
| Id | 75507 |
The broadcast conference PBX configuration is changed: now, the third party conference unit option is saved and must be enabled to use the configured id prefix and suffix. Otherwise they are ignored now and default values for the innovaphone conference interface of the current device firmware version are used. This fixes the configuration if the firmware is updated from V8 to V9 and the innovaphone conference interface is used.
Disabling the "Create Dynamic Conference Id" option in firmware V9 hotfix 5 and 6 is also fixed now.
IP241,IP222,IP232: Call duration display wraps after 100 minutes
| Status | Closed |
| Id | 75508 |
Now after 60 minutes the display changes from mode [mm:ss] to [hh:mm].
TLS: Duplicate alert message on malformed ClientHelloV2
| Status | Closed |
| Id | 75509 |
Only one alert should be sent per session.
TLS: Improved negotiation of protocol version
| Status | Closed |
| Id | 75510 |
TLS server unnecessarily rejected ClientHello messages with TLS 1.1 and higher. Instead of rejecting it should tell the client that it wants to use TLS 1.0.
TLS: Skip empty records
| Status | Closed |
| Id | 75511 |
TLS record layer should ignore records with zero length without doing anything.
myPBX: Script error after waking the PC up from hibernation
| Status | Closed |
| Id | 75532 |
Sometimes window.open throws an unhandled exception in this case.
SIP: Check proxy availability did not work in any scenario
| Status | Closed |
| Id | 75542 |
OPTIONS can be used to poll remote proxy's availablity
to avoid TCP timeout when INVITE is to be sent.
Signaling interface is marked as down and not used anymore.
IP232: Hiding touch keyboard by touching a control
| Status | Closed |
| Id | 75576 |
Touching a text control activates the touch keyboard.
Touching the text control again now deactivates the touch keyboard.
IP241,IP222,IP232: CFU information in header bar
| Status | Closed |
| Id | 75654 |
CFU information in header bar is now displayed even if there's not is enough space between name and number.
Either name or number is omitted is required.
IP241,IP222,IP232: Replace triangle by arrow to display diversion/transfer information
| Status | Closed |
| Id | 75689 |
Replace quite heavy 'BLACK RIGHT-POINTING POINTER'
by much lighter 'RIGHTWARDS ARROW'
to display diversion/transfer information
on call control, fkeys and call lists.
H.323: A forwarded HopCount>32 could result in a very small HopCount
| Status | Closed |
| Id | 75765 |
There are only 5 bits for transmitting a HopCount in H.323. A HopCount from SIP is typically 70 and this value was not reduced to 32 but only the 5 lower bits were transmitted, which resulted in a HopCount of 6
IP: Minor memory management change
| Status | Closed |
| Id | 75770 |
no use of malloc in ipproc
IP241,IP222,IP232: Two waiting calls are now displayed (instead of one)
| Status | Closed |
| Id | 75789 |
Waiting calls are displayed below the two main calls.
Only two lines of information and grey background.
SIP: Handling of some more proprietary Alert-Info and Call-Info in INVITE
| Status | Closed |
| Id | 75824 |
Alert-Info: <Bellcore-dr3>;info=alert-recall
and
Alert-Info: <http://not_used.com>;info=alert-autoanswer
and
Call-Info: <sip:127.0.0.1>;answer-after=0
PBX Waiting: Leg2 information sent with calls to operators not adjusted correctly for nodes
| Status | Closed |
| Id | 75830 |
If a Waiting Queue was configured in a Node not the root node, the leg2 info was not adjusted corrcectly. The leg2 information is used to signal to the operator which Waiting Queue is forwarding the call
SIP: Decoding of RFC-4412 definitions
| Status | Closed |
| Id | 75848 |
The 'Resource-Priority' Header Field
The 'Accept-Resource-Priority' Header Field
The 'resource-priority' Option Tag
417 Unknown Resource-Priority response
SIP: Handling of 422 Session Interval Too Small
| Status | Closed |
| Id | 75873 |
Get Min-SE and re-try INVITE
SIP: Send b=TIAS attribute in media description
| Status | Closed |
| Id | 75881 |
Send b=TIAS attribute in media description
remove indirect calls to os_mem_alloc() via malloc()
| Status | Closed |
| Id | 75886 |
malloc() shall not be used anymore in sources dedicated to run on innovaphone hardware
IPv6: Memory leak if packets received, which are not handled locally
| Status | Closed |
| Id | 75920 |
E.g. an IPv6 multicast UDP packet, for some other application was not deleted.
SIP: Fix for video negotiation
| Status | Closed |
| Id | 75954 |
Fix for video negotiation
phone_orchid: after leaving a conference the active call was mute sometimes
| Status | Closed |
| Id | 75955 |
The conference was not stopped internally in all cases.
Flashman: Show meaningful result after OEM certificate upload in production
| Status | Closed |
| Id | 75958 |
"Certificate" instead of "-unknown-"
SIP: Fix for Supported header
| Status | Closed |
| Id | 76016 |
Re-implemented to easily add more tags
IP222,IP232: Alpha mode did not work on multi-line edit controls
| Status | Closed |
| Id | 76165 |
Could no enter alpha chars using the num block
SIP: Trap when parsing very large History-Info header
| Status | Closed |
| Id | 76198 |
Trap when parsing very large History-Info header
SIP: SIP message may exceed available buffer
| Status | Closed |
| Id | 76203 |
Trace message like this can occur:
ERROR: SIP message buffer (1025) exceeded! (9,14,801d781c)
SIP: CSeq missing
SIP: Message encoding failed!
SIP: Contact not allowed in BYE/CANCEL/PRACK request
| Status | Closed |
| Id | 76210 |
Acc. to RFC-3261 there must not be Contact header line in BYE/CANCEL request
Acc. to RFC-3262 there must not be Contact header line in PRACK request
reset could fail, if issued during firmware update
| Status | Closed |
| Id | 76266 |
and even worse any following reset command were not accepted as well
IP241,IP222,IP232: Show "New message from ..." on active phone screen
| Status | Closed |
| Id | 76283 |
Show "New message from ..." on phone screen for 3 seconds when active with calls (non-idle).
Like on b/w phones.
IP241,IP222,IP232: Toggle of display-name/name-alias/number was buggy
| Status | Closed |
| Id | 76290 |
String termination was missing.
Garbage data was displayed.
IP232: Open call details when touching entry in call list
| Status | Closed |
| Id | 76292 |
Instead of initiating call to remote party in call list entry.
H.323: Merges from v10 for Media Negotiation fixes and special OEM protocol features
| Status | Closed |
| Id | 76301 |
- Problem with transparent forwarding of SDP
- special SRTP key exchange
- Video fixes
- Unit Testing support
IP-DECT: Memory leak with rejected radio registration
| Status | Closed |
| Id | 76308 |
Memory leak occurred with a rejected radio registration is fixed, now.
PBX-SOAP: When monitoring Waiting queue, the peer number was not indicated
| Status | Closed |
| Id | 76315 |
The number was considered presentation restricted, because of some bit manipulation error
ip22/24/28/302/305/6010/3010/1060: Fax failed if tones are sent during the fax call
| Status | Closed |
| Id | 76342 |
In some cases the Fax messages were missing due to tones send during the fxx call Status: ac_dsp3.cpp
Gatway: Transfer handling may cause re-routing after regular hang-up
| Status | Closed |
| Id | 76357 |
Re-routing is expected to take place on non-connected calls only.
PBX Number Map: Not possible to use overlap dialing to Number map with incomplete destination
| Status | Closed |
| Id | 76377 |
If a Number Object with incomplete destination was called and the number was to be completed with overlap dialing a wrong number was called.
This is a usefull feature to use Number Maps as quick dial to other nodes. In this case Number Maps are used with a destination of the remote node, so the number is incomplete, the number within this node has to be dialed in addition to the Number of the Number Map object.
SIP: Support for media recording
| Status | Closed |
| Id | 76396 |
Media recording to HTTP URL
Gateway: Conference interface, no voice
| Status | Closed |
| Id | 76419 |
The ADSP firmware is changed to version 122. This fixes a bug in the conference interface of IP6000/IP6010/... which results in conference calls without voice in one direction for a single member.
phone_orchid: spurious trap in long conference calls
| Status | Closed |
| Id | 76445 |
in long confernce calls the phone may trap because the DSP delivers an unexpected zero size packet.
IP241,IP222,IP232: Call held indication
| Status | Closed |
| Id | 76454 |
If remote side has put the call on hold
the user should get a visible indication
(in addition to hearing music on hold).
New call status "held" is displayed.
LDAP-Expert, Edit Object Dialog: Could show more than one object
| Status | Closed |
| Id | 76464 |
LDAP-Expert, Edit Object Dialog: Could show more than one object
Qsig: Connected Number could be encoded wrong
| Status | Closed |
| Id | 76472 |
Qsig: Connected Number could be encoded wrong
Gateway: Better handling of call-reroute requests
| Status | Closed |
| Id | 76515 |
Apply interface maps to numbers in reroute request
PBX-SOAP: Better error handling when using a Waiting Queue for outgoing calls
| Status | Closed |
| Id | 76544 |
- call was hanging if an invalid number or user without registration was called
- trap if outgoing call was done on operator connect
Spurious leaks in leak checking
| Status | Closed |
| Id | 76589 |
A problem which mostly shows up in automated release testing and then requires manual investigation.
SoftwarePhone: Auto start configuration
| Status | Closed |
| Id | 76595 |
The duplicate back slash occurred with the install directory is fixed now. Used by the run mode configuration 'At login'.
myPBX launcher: Window position was not always remembered
| Status | Closed |
| Id | 76626 |
When the user just moved the window without resizing it, the window position was not remembered for the next session.
IP241,IP222,IP232: Some diagnostics to check phone's ui memory consumption
| Status | Closed |
| Id | 74049 |
Where have all the memory gone.
SIP: Support for "Content-ID" in "multipart/alternative" bodies
| Status | Closed |
| Id | 74161 |
Support for "Content-ID" in "multipart/alternative" bodies
| Status | Closed |
| Id | 74278 |
To make end of list of menu entries more discernible.
myPBX: Support static URLs for application sharing
| Status | Closed |
| Id | 74389 |
In some application sharing solutions a fixed link can be used to create and join meetings (GoMeetNow, BeamYourScreen).
There are two URLs configured, one for the presenter and one for attendees. When the user clicks the aplication sharing button the links are sent using chat messages.
PBX: Tracing flag turns on tracing in all dyn PBX's as well
| Status | Closed |
| Id | 74390 |
Helpfull to debug dyn PBX setups
debug method raw_ethernet
| Status | Closed |
| Id | 74897 |
A new debug method to trace raw ethernet data.
Support for new hardware
| Status | Closed |
| Id | 74990 |
.
possible use of push-pull drivers for new hardware
| Status | Closed |
| Id | 75051 |
not visible to customers
PBX/IP-DECT: User password length
| Status | Closed |
| Id | 75453 |
The maximum user password length in the PBX and IP-DECT is increased to 23.
H.323: Support for switch back to local media on endpoint
| Status | Closed |
| Id | 75522 |
needed to resume normal call after a conference using external conference unit
IP232: Touch gesture to open directory-search/indirect-dialing screen
| Status | Closed |
| Id | 75549 |
Vertical upwards on idle screen opens directory-search/indirect-dialing screen.
Vertical downwards on idle screen opens main-menu screen.
PBX-SOAP: Wsdl Versioning mechanism fixed
| Status | Closed |
| Id | 75552 |
Better mechanism, to avoid constantly adding arguments to the Version function with new Versions.
different port naming convention for new hardware
| Status | Closed |
| Id | 75575 |
BRI1..5 instead of TEL1..4+PPP
PBX-SOAP: UserHold without MOH to local User
| Status | Closed |
| Id | 75577 |
UserHold was sending MOH to the local and the remote User. With the argument remote=true, the MOH is sent to the remote user only
myPBX launcher: Use default configuration from local machine registry hive
| Status | Closed |
| Id | 75634 |
If no user configuration is given, the following values are copied from the local machine hive.
"Software\\innovaphone\\myPBX\\URL"
"Software\\innovaphone\\myPBX\\secondaryURL"
Also the autostart checkmark is disabled, if autostart is activated, globally. Autostart is controlled with the following registry key.
"Software\\Microsoft\\Windows\\CurrentVersion\\Run\\innovaphone myPBX"
SIP: support for proprietary message header
| Status | Closed |
| Id | 75864 |
Support for proprietary message header (CAL)
H.323: Call to SIP Trunk with MediaRelay and exclusive coder renegotiation from TONE to early media did not work
| Status | Closed |
| Id | 75959 |
No ringback and no audio after connect
SIP: Added Call-Info header to re-INVITE for hold/retrieve
| Status | Closed |
| Id | 76011 |
Call-Info: <urn:X-cisco-remotecc:hold>
Call-Info: <urn:X-cisco-remotecc:resume>
SIP: Support for active call pickup acc. to RFC-3891
| Status | Closed |
| Id | 76035 |
Sending INVITE with Replaces header to do pickup at SIP PBX's
phone ip222, ip232 : USB headset support (beta)
| Status | Closed |
| Id | 76463 |
general support for USB headsets added, some headsets are tested and working, others will follow soon
phone - Headset Function Key can be configured now as enable/disable or as call control key
| Status | Closed |
| Id | 76468 |
a headset key mode can be configured to use the key either to enable/disable the headset (Mode: Enable) or to start/accept/clear calls via headset (Mode: Control)
V9 Hotfix 9 (90600.11, withdrawn)
Changes included in Version 9 hotfix9 Definition
add programmable tftp retry limit to httpclient
| Status | Closed |
| Id | 74936 |
currently there is a fixed retry limit of 4. some OEMs want to set this dependent on expected file size. Status: checked in to 10.00, 9.00, 90600
H.323: Media Negotiation problem with conferences on IP-DECT
| Status | Closed |
| Id | 76314 |
A channel was not switched to the conference after a hold/retrieve cycle
PBX Mobility: Dialed digits could get lost, when using Opticaller
| Status | Closed |
| Id | 76598 |
There could be a collision of a dialed digit with media renegotiation. For example if with the first digit a media was switched to inband information from a carrier.
phone: Ring Tone Titles containing apostrophes garble phone configuration
| Status | Closed |
| Id | 76745 |
When under "Phone/Ring Tones/Add Ring Tone" a title containing apostrophes is entered the page "Phone/User-x/Preferences" cannot be edited anymore because of a XML-Error. Status: checked in to 10.00, 9.00, 90600
SIP: Record-Route handling on outbound subscriptions
| Status | Closed |
| Id | 76831 |
Processing of Record-Route in SUBSCRIBE response.
Simply missing.
IP241: Headset noise during firmware update
| Status | Closed |
| Id | 77111 |
. Status: phone_orchid.cpp
phone: Pickup fuction key is not displayed while all alerting calls are displayed on a Partner key
| Status | Closed |
| Id | 77393 |
alerting calls displayed on a Partner key are not displayed on the Pickup key.
if nothing has to be displayed on the pickup key the key should not disappear but display the 'idle' label
Status:
checked in to 10.00, 9.00, 90600
H.323: Media Negotiation problem with transfer in Gateway (not PBX)
| Status | Closed |
| Id | 77683 |
Under special conditions a blind transfer happend in the Gateway could result in a call without media. This only happened if the call was transfered twice and the destination of the first transfer was a physical interface.
PBX Twinning: When calling another (twin) phone, the call was sent to the original phone also
| Status | Closed |
| Id | 77829 |
It still works, but this waiting call could be confusing
PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX
| Status | Closed |
| Id | 77874 |
A call from a object within a node with escapes on a slave PBX was not routed to the master if the destination was within the same node and not known on this slave but was sent to the node-extern destination directly
PBX Waiting: Name Id missing in calls initiated with SOAP
| Status | Closed |
| Id | 77911 |
When a Waiting Queue is used by applications to initiate outgoing calls, the name of the waiting queue should be sent with these calls as calling name. This name id was missing
RTP: Potential random trap when closing channels
| Status | Closed |
| Id | 77918 |
Happens if there is a collision with a received packet and closing of the channel. Window for this is very small, so it should happen very rarely. Probability can increase with high load.
Web-UI: Visually separated commands
| Status | Closed |
| Id | 77954 |
Visually separated commands 'Clear' and 'Save' on Maintenance/diagnostics/Events.
Also 'download' and 'delete' on General/License.
IP241,IP222,IP232: Show special symbol for call completion entries in call list
| Status | Closed |
| Id | 77966 |
Show special symbol for call completion entries in call list to separate from usual missed calls.
IP241,IP222,IP232: Status message "Recording" not displayed
| Status | Closed |
| Id | 77992 |
Status message "Recording" needs to be displayed during recording.
Gateway: Wrong Media info sent for calls forwarded from a VOIP interface to another VOIP interface
| Status | Closed |
| Id | 78024 |
This could cause wrong information to be displayed on a PBX calls page. Esspecially SRTP was not indicated.
Better to not send any Media Info in this case.
IP241,IP222,IP232: Wrong display name in call list
| Status | Closed |
| Id | 78047 |
Name identification of remote party is to be displayed,
but name identification of diverting party was displayed instead
on incoming calls list.
phone: the call which was the active call at start of a conference was not automatically cleared on a release from remote
| Status | Closed |
| Id | 78086 |
The call which was the active call at start of a conference had to be cleared manually but the call which was on hold at start of conference was cleared automatically.
Now any call will be automatically cleared when relesed from remote.
Status:
checked in to 10.00, 9.00, 90600
phone_orchid: remaining call mute after remote relase for the call which was the active call at start of a conference
| Status | Closed |
| Id | 78091 |
did not happen after a remote release for the call which was on hold at start of the conference. Status: checked in to 10.00, 9.00
IP1060 IP3010 IP6000 IP6010: T1 mode: wrong pulseshape for short lines.
| Status | Closed |
| Id | 78157 |
Slew rate and level for T1 pulse for short lines was too high
Status:
falc56_drv.cpp
falc56_drv.xsl
permit hid usage report sizes of up to 64 byte, some devices send more than 32
| Status | Closed |
| Id | 78158 |
Jabra LINK 14201-30 responds with 33 byte to certain requests
phone: ip222,ip232: Muting USB headset micro via microphone key did not work but MICROPHONE OFF was displayed
| Status | Closed |
| Id | 78159 |
. Status: checked in to 10.00, 9.00, 90600
IP241,IP222,IP232: Trap in font rendering
| Status | Closed |
| Id | 78210 |
Some characters at beginning of line may cause trap.
Gateway: Trap when interworking Call Completion
| Status | Closed |
| Id | 78228 |
Trap when interworking Call Completion.
LOG CALL 6 A:Call -> / PRI2::->*::
R_CALL free error c18a59b8
phone: ip241: memory leak when ip230x extension module is attached and used
| Status | Closed |
| Id | 78311 |
one packet lost per keystroke on extension module
PBX: Dynamic group function keys did not work with non-ascii characters in group name
| Status | Closed |
| Id | 78322 |
Name was not correctly converted to internal utf-8 representation
phone: ip222,ip232: trace USB media data only if explicitely requested
| Status | Closed |
| Id | 78326 |
use "config add USB-HOST AUDIO /trace" if data is needed
H.323: Potential Trap in case of special malformed RAS registration
| Status | Closed |
| Id | 78368 |
A NULL pointer access could happen
PBX MWI Object: Malformed H.450 interrogation result was sent if no message
| Status | Closed |
| Id | 78376 |
It still worked, because the wrong message was usually interpreted as no message available, which was correct, but it does not look nice in wireshark
SIP: Removed b=TIAS attribute from media description
| Status | Closed |
| Id | 78379 |
Causes trouble on some SIP providers (neotel.at)
DHCP server leases with hostnames containing non-ascii latin1 characters (for example Umlauts) could not be displayed
| Status | Closed |
| Id | 78382 |
In pre-V9 firmware hostnames were stored latin1-encoded. Names contaning non-ascii latin1 charaters must be converted to UTF8 before display. Status: checked in to 10.00, 9.00, 90600
MyPBX: Changing password did not work any more
| Status | Closed |
| Id | 78391 |
Collateral damage of fix #75453: PBX/IP-DECT: User password length
phone: ip222, ip232: USB headset sometimes mute after disconnect
| Status | Closed |
| Id | 78414 |
depending on isochronous transfer state at disconnect time. seems to happen more often with relatively cheap headsets.
SIP: Be save against sudden death of SIP caller
| Status | Closed |
| Id | 78460 |
Lifetime of an INVITE trasnaction is not limited by any timeout
after provisional response has been send/received.
Sudden death of a caller make calls hang forever.
Now overall lifetime of an INVITE server transaction is limited to 3 minutes.
After expiration fimnal reject response is sent and call is released.
IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax failure after transfer
| Status | Closed |
| Id | 78487 |
channel was reconfigured on remote switch to T38, without a new activate fax relay command Status: ac_dsp3.cpp
PBX Boolean: Avoid unnecessary load by boolean monitoring
| Status | Closed |
| Id | 78504 |
A call was sent to the monitoring endpoint every 10s
IP241,IP222,IP232: No RTP send on outgoing call in some cases
| Status | Closed |
| Id | 78550 |
No RTP send on outgoing call in some cases
IP6000: Traps in DSP driver under high load
| Status | Closed |
| Id | 78591 |
under high load timing may change. Checks in driver relaxed to take this into account.
PBX-SOAP: Limit on size for Admin call too small
| Status | Closed |
| Id | 78592 |
Only 4k was allowed
SoftwarePhone: Default ToS value for RTP
| Status | Closed |
| Id | 78593 |
Now the default ToS value for RTP packets is 0xB8.
PBX Filter: Maximum length increased from 13 to 17 digits
| Status | Closed |
| Id | 78594 |
13 digits could be too little if an international number including a prefix of an external line was to be checked
IP241,IP222,IP232: Wrong call direction indication on PARTNER fkey
| Status | Closed |
| Id | 78642 |
When partner calls someone else it my displayed as if partner is called.
(Only if option "Show connected Party in busy State" is enabled)
IP222 IP232 IP241: Sporadic noisy sidetone
| Status | Closed |
| Id | 78649 |
Analog and digital sidetonegain were enabled,now only the analog sidetonegain is used. Status: ac_codec3.cpp
Gateway: Handling of blind transfer gets stuck in routing
| Status | Closed |
| Id | 78709 |
Routing of transfer call stops with: reason='incomplete'
But during transfer no dialing digits will follow.
PBX-SOAP: UserCall new option diversion override
| Status | Closed |
| Id | 75773 |
To initiate calls, which are not diverted by a configured CFU.
Use Info[] element with type="fty-no-cf"
SIP: Support for "Allow-Events: ccnr" and "Allow-Events: ccbs"
| Status | Closed |
| Id | 76658 |
For interoperability:
Support for "Allow-Events: ccnr" and "Allow-Events: ccbs"
Phone->PBX LDAP Search returns normalised number to be dialled by phone
| Status | Closed |
| Id | 77261 |
1)Phone->PBX LDAP Search returns normalised number to be dialled by phone.
2)Phone receives info about escape digits when registering at its PBX.
With 1) the PBX includes an object's normalized number into the LDAP search result.
With 1) the phone is able to dial that normalized number.
With 2) the phone is able to prefix required escape digits to the received normalized number.
Status:
checked in to 10.00, 9.00, 90600
PBX soap method SetPresence
| Status | Closed |
| Id | 78092 |
PBX soap method SetPresence added to set the presence of a PBX user.
phone ip222, ip232 : USB headsets need not to be configured anymore
| Status | Closed |
| Id | 78114 |
If an USB headset with a known signature (vendor/product id) is plugged it is automatically enabled. This is indicated by the headset symbol in the status line.
"Phone/Preferences/Start Outbound Call on Electronic Hook Switch (EHS) Signal" is implied in this case because some headsets will loose state if a hoook signal is ignored.
Status:
checked in to 10.00, 9.00, 90600
phone: ip222, ip232: USB headset support - Plantronics C420, GN2000 USB - MS OC Version
| Status | Closed |
| Id | 78201 |
. Status: checked in to 9.00, 90600
PBX Mobility: No-Alert checkmark as workaround for provider which do not provide Alerting
| Status | Closed |
| Id | 78410 |
Some SIP provider do not provide an Alerting signal when a mobile phone is called. This could result in no ringback signal to the caller or the min/max-alert feature not working.
This new checkmark provides a fake Alerting in case Progress is received
show linux shutdown warning on firmware reset page
| Status | Closed |
| Id | 78665 |
If linux is running, a warning is now shown, that linux should be shutdowned before a firmware reset is performed.
Linux: Enable/Disable support link
| Status | Closed |
| Id | 78786 |
Now the Linux menu is always shown and a link is provided to enable or disable the Linux support (RAM reservation). The support state is also saved in the downloaded configuration file and restored with the upload.
Update: The support state is only saved in the downloaded configuration with password. The state is not saved in the configuration file with standard password or if downloaded by the update server. Please use the next or a later hotfix instead, see also fix #78836.
V9 Hotfix 10 (90600.12)
Changes included in Version 9 hotfix10 Definition
PBX: Trap if enabling Unknown Registrations
| Status | Closed |
| Id | 78904 |
Collateral Damage from
fix: #77261: Phone->PBX LDAP Search returns normalised number to be dialled by phone
Linux: Enable/Disable configuration
| Status | Closed |
| Id | 78836 |
With the new feature #78786 the configuration is only saved in the downloaded configuration file with password. Now the information is also included in the configuration file with standard password and in the file downloaded by the update server.
V9 Hotfix 11 (90600.14)
Changes included in Version 9 hotfix11 Definition
IP222/IP232/IP241: Ethernet link configuration
| Status | Closed |
| Id | 78067 |
In some case ethernet link configuration and display didnt work.
Now the autoneg status is shown if available, and the phy status if no autoneg didnt complete.
An option to use autoneg with fixed speeds and a link trace option (/ltrace) are added.
Status:
orchid_drv.cpp
orchid_drv.h
orchid_drv.xsl
Media Relay: Don't send dummy RTP data on incoming calls
| Status | Closed |
| Id | 78252 |
Dummy RTP data is sent just in case a NAT router is within the media path to set a UDP mapping in case both legs of the call contain a NAT router.
An example for such a situation is a call coming from a SIP provider thru a NAT router to the PBX, which forwards the call back out to the SIP provider. The NAT router won't get RTP data from inside to set the mappings.
The dummy RTP was sent to all legs of the call, but it is better to send it to outgoing call legs only, because endpoints calling in may turn off a local ringback tone when receiving dummy RTP
H.323/SIP: Avoid delayed SDP within outgoing calls as far as possible
| Status | Closed |
| Id | 78466 |
If media renegotiation is needed, to one side of the call an request for a media proposal (in SIP terms, this is an INVITE without SDP) is sent. The media proposal (in SIP terms SDP offer) is then forwarded to the other side.
The request for an offer should if possible not sent with the initial call, because there is equipment which does not handle 'delayed SDP'
Media Relay: Don't terminate T.38 protocol in media relay, forward transparently
| Status | Closed |
| Id | 78610 |
This should add robustness and reduce CPU load
Flash Directory: Display Error Messages
| Status | Closed |
| Id | 78656 |
Flash Directory: Display Error Messages
Shedding light on replication problems.
currently only in 10.00
RTP: No check for changed media destination because of received multicast
| Status | Closed |
| Id | 78698 |
When receiving RTP normally a check is done if the source of the RTP is the same as we are sending to. If this is not the case, we assume the destination of the RTP is behind a NAT router and we change the destination address to the source address of the received RTP.
This does not make sense for received multicast.
IP22 IP24 IP28 IP305: Faxempfang gelegentlich gestrt
| Status | Closed |
| Id | 78803 |
DSP erhlt zum falschen Zeitpunkt ein close-rtp. Status: ac_dsp3.cpp
IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax receive max packet size increased to 1024
| Status | Closed |
| Id | 78811 |
For internet fax devices Status: ac_fax3.cpp
phone: ip222, ip232: USB controller sometimes hangs
| Status | Closed |
| Id | 78824 |
happened with certain headsets after fast connect/disconnect sequences
Status:
checked in to 10.00, 9.00
merged to 90600
SIP: SDP answer sometimes contains 2 media descriptions for audio
| Status | Closed |
| Id | 78833 |
... instead of one for audio and one for video.
Offer:
v=0
o=- 1 2 IN IP4 130.30.1.111
s=CounterPath Bria
c=IN IP4 130.30.1.111
t=0 0
m=audio 1902 RTP/AVP 0 8 18
a=alt:1 1 : 1A98X9/s 31UwK5IA 130.30.1. 111 1902
a=fmtp:18 annexb=yes
a=rtpmap:18 G729/8000
a=sendrecv
a=x-rtp-session-id:ACE43888C33A4AC8895C7F8F69380914
m=video 28212 RTP/AVP 115 34 123 124
a=alt:1 1 : yjz/mbyQ aWGuD8xJ 130.30.1. 111 28212
a=fmtp:115 QCIF=1;CIF=1;I=1;J=1;T=1
a=fmtp:34 QCIF=1;CIF=1
a=fmtp:123 profile-level-id=42801e; packetization-mode=0; max-mbps=48600
a=fmtp:124 profile-level-id=42801e; packetization-mode=1; max-mbps=48600
a=rtpmap:115 H263-1998/90000
a=rtpmap:34 H263/90000
a=rtpmap:123 H264/90000
a=rtpmap:124 H264/90000
a=sendrecv
a=x-rtp-session-id:694677A9F65D419FB7B9E451FFCA02CC
Answer:
v=0
o=- 3141 1 IN IP4 10.230.5.1
s=-
c=IN IP4 10.230.5.1
t=0 0
m=audio 16410 RTP/AVP 0 13
b=TIAS:64000
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
m=audio 16410 RTP/AVP 0 13
b=TIAS:64000
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
Conference: Permanent noise in conference channel
| Status | Closed |
| Id | 78838 |
Permanent noise in conference channel.
Increasing with every new participant.
PBX: When doing a show users, unnecessary data was sent to the browser
| Status | Closed |
| Id | 78878 |
The data could be reduced and thus speeding up the display of big users lists.
PBX Admin UI: Editing profiles was not possible if the user had too many contacts
| Status | Closed |
| Id | 78883 |
The form is now submitted using POST instead of GET.
SIP: Wrong number of waiting messages (MWI)
| Status | Closed |
| Id | 78890 |
MWI: Number of voice messages not decoded from incoming NOTIFY(application/simple-message-summary).
Was either 1 or 0.
SIP: Send even anonymous Diversion header
| Status | Closed |
| Id | 78954 |
If diverting party's identity is unknown/hidden.
Diversion: ;reason=user-busy
SIP: Failed to decode presence XML from CUCM
| Status | Closed |
| Id | 78967 |
Support for namespace "urn:ietf:params:xml:ns:pidf:status:rpid" added.
IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: Fax failure after transfer #2
| Status | Closed |
| Id | 78969 |
tone flag was not disabled Status: ac_dsp3.cpp
H.323: Media problem with SIP call-completion on IP-DECT
| Status | Closed |
| Id | 78970 |
There was no media on the successful completion call (SDP recived in Progress was not forwarded)
SIP: Send 'Connected Number' in P-Asserted-Identity header of 200/OK
| Status | Closed |
| Id | 78974 |
Send 'Connected Number' in P-Asserted-Identity header of 200/OK
if different from original called number.
SIP: Re-transmission of ACK with SDP answer
| Status | Closed |
| Id | 78975 |
Handling of re-transmissions of 200/OK with SDP offer.
ACK must contain SDP answer.
IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305: DSP packet debug didnt show some packets, version endian was wrong
| Status | Closed |
| Id | 78985 |
cleanup
Status:
ac_491.cpp
ac_494.cpp
IP-DECT: Radio/Master calls switched to slowstart
| Status | Closed |
| Id | 78987 |
If first media answer was received with a PROGRESS message, the call leg from the radio to the master was switched to progress (no EFC features are defined for the Progress message).
This could cause media problems later in the call with hold/retrieve/transfer.
SIP: REFER does not work as expected on Gateway interfaces without registration
| Status | Closed |
| Id | 79008 |
Bad side effect of a previous fix in HF8:
http://mantis.innovaphone.com/view.php?id=76309
Introducing new SIP interop tweak "Take Refer-To URI as Remote Target URI"
to be found on the Gateway interface config (GWx).
See http://mantis.innovaphone.com/file_download.php?file_id=66299&type=bug
IP-DECT: Better handling when receiving a call, while a previous call is being released
| Status | Closed |
| Id | 79022 |
Call to service number, which cause a callback can create problems.
Edss1 Interworking: Sending divertingLegInformation2 as PublicPartyNumber(was UnknownPartyNumber)
| Status | Closed |
| Id | 79025 |
Such that the type of number resulting from an interface cgpn-map can also be asn.1 encoded.
Gateway: Mapping of Message Center Number of MWI
| Status | Closed |
| Id | 79039 |
Mapping of Message Center Number of MWI according to CGPN mapping.
Qsig Interworking: Apply interface CGPN map to DGPN
| Status | Closed |
| Id | 79060 |
Qsig Interworking: Apply interface CGPN map to DGPN
PBX: A CFB at a PBX object was executed on call to busy user
| Status | Closed |
| Id | 79199 |
It should only be executed, if the calls exceeded the Busy On number.
PBX: Mapping of Message Center Number of MWI
| Status | Closed |
| Id | 79232 |
Mapping of Message Center Number of MWI.
PBX-SOAP: Conference Id was missing in call-info for parked calls
| Status | Closed |
| Id | 79243 |
This caused problems with the innovaphone operator
PBX-SOAP: UserClear cause should not be used for remote party
| Status | Closed |
| Id | 79249 |
This can result in strange call clearing without tones
PBX: Include error in resulting XML if submit-object with insufficient rights was attempted
| Status | Closed |
| Id | 79289 |
Problem with innovaphone operator.
phone: ip222,ip232,ip241: wrong AM/PM time display on status line
| Status | Closed |
| Id | 79302 |
15:19 PM was displayed instead of 03:19 PM Status: checked in to 10.00, 9.00, 90600
Voicemail: Default for mwi.basicService now Speech(1) (was allServices(0))
| Status | Closed |
| Id | 79303 |
Voicemail: Default for mwi.basicService now Speech(1) (was allServices(0))
May from now on be overriden by variable "$_pbxmwiservice".
PBX: Local objects could not be called from Nodes with escapes as expected
| Status | Closed |
| Id | 79317 |
Was not possible to call at all, or the number was wrong
PBX-Trunk: Handling for Incomplete, Invalid, Busy, No Anser destinations for Media calls only
| Status | Closed |
| Id | 79319 |
For example this should not be done for presence/dialog subscriptions
Refactored some ASN.1 BER Handling
| Status | Closed |
| Id | 79357 |
Became necessary in the run of: #79260: ASN.1 BER: Support for indefinite-length encoding
PBX: Conference & VM
| Status | Closed |
| Id | 79359 |
The last connected user is not disconnected although it is configured. It occurs if a VM PBX object forwards the call to the BC Conference object (like the innovaphone conferencing script). It is fixed now.
IP-DECT: Subscription could get lost randomly with logout/login cycle
| Status | Closed |
| Id | 79397 |
Problem with user database handling
H.323: Media problem if PBX rtp-proxy is activated for CFNR to external destination
| Status | Closed |
| Id | 79430 |
no voice
HTTP: Could not have sockets with same ports but different remote addresses
| Status | Closed |
| Id | 79458 |
This was a general problem with the new TCP stack, used together with IP6. Problem only happened for HTTP because only for HTTP this new stack is currently used.
ISDN: Fix for call completion interworking
| Status | Closed |
| Id | 79498 |
CCBS/CCNR does not work in some cases
SIP: From-URI may got lost after call transfer
| Status | Closed |
| Id | 79536 |
UPDATE may be send out missing From-URI.
AD Replication stuck after connection loss
| Status | Closed |
| Id | 79541 |
Didn't reconnect. Side effect of after-hf10 fix
PBX Waiting/Broadcast: Incomplete CDRs if CFNR configured on object
| Status | Closed |
| Id | 79549 |
If the CFNR was executed, the CDR for the call to the Waiting Queue/Broadcast object ended without release.
PBX: Web User interface problem with IE and non-ascii PBX names
| Status | Closed |
| Id | 79568 |
For example groups at a user assigned to a PBX with non-ascii characters could not be edited.
The problem is a bug in IE XSL translation which does special handling of href attributes. Same thing with onclick attribute works.
ip3010 TEL port not working in NT mode
| Status | Closed |
| Id | 79570 |
TEL port of ip3010 gateways configured in NT mode do not get Physical Link up indication. This problem applies to V9hotfix7 up to V9hotfix10. Status: ip6010.cpp
SIP: Max forward value of 32 could be too small for some provider
| Status | Closed |
| Id | 79578 |
For a starting value of max-forwards a value of 32 was used, because this is the maximum value in H.323. This was too small for some sip providers. Starting value now increased to 64 and on H.323 the half value is transmitted.
SIP: Re-negotiation for T38 did not work in media-relay scenarios
| Status | Closed |
| Id | 79583 |
Bad SDP answer was generated.
Status:
Introduced by
#77277: SIP: SDP answer must have the same number of media descriptions as received offer
HTTP-Client: MD5-sess authentication
| Status | Closed |
| Id | 77773 |
HTTP Digest Authentication with alogrithm=MD5-sess.
Choose the first supported "WWW-Authenticate" line from 401 response headers.
Needed for new versions of IIS.
Status:
http://wiki.innovaphone.com/index.php?title=Support:DVL-Feature_Requests#HTTP_Client
Phone: Possibility to reject incoming SIP calls with customized reason phrase
| Status | Closed |
| Id | 77928 |
E.g. "SIP/2.0 480 Do not disturb" instead of "SIP/2.0 480 Temporarily unavailable"
Active Directory Replication: Editfield for Poll Timer added
| Status | Closed |
| Id | 78631 |
If change notifications cannot be received from an AD, a poll timer can be specified. A re-replication is going to take place after the poll timer expired.
X.509: Support for PKCS#12 files
| Status | Closed |
| Id | 78820 |
Support for certificate import using password encrypted PKCS#12 files.
Currently the following encryption types are supported:
pbeWithSHAAnd128BitRC4 (1.2.840.113549.1.12.1.1)
pbeWithSHAAnd40BitRC4 (1.2.840.113549.1.12.1.2)
pbeWithSHAAnd3-KeyTripleDES-CBC (1.2.840.113549.1.12.1.3)
pbeWithSHAAnd128BitRC2-CBC (1.2.840.113549.1.12.1.5)
pbewithSHAAnd40BitRC2-CBC (1.2.840.113549.1.12.1.6)
myPBX: Pass language code to reporting
| Status | Closed |
| Id | 78874 |
For localization of the call list.
Phone: Show number of voice messages on label of MWI fkey
| Status | Closed |
| Id | 78894 |
Use "%u Message(s)" as label text for active state of MWI fkey.
IP-DECT: Cisco SIP conferencing
| Status | Closed |
| Id | 78988 |
Cisco proprietary softkey events implemeneted to start a 3pty conference
IP-DECT: Feature Code for calling predefined service URIs
| Status | Closed |
| Id | 79028 |
User for Cisco compatibility
Cisco Softkey Features, Status updates and Park
| Status | Closed |
| Id | 79030 |
suported_mask needs more flags
Softkey Feature Park
IP-DECT: Cisco Park
| Status | Closed |
| Id | 79032 |
Send Park softkey event
| Status | Closed |
| Id | 79036 |
.
IP-DECT: SRTP for an OEM device
| Status | Closed |
| Id | 79061 |
Now SRTP is configurable for an OEM device.
| Status | Closed |
| Id | 79104 |
.
Merge of v10 changes: Dynamic Timer Management/oem serial no
| Status | Closed |
| Id | 79127 |
Better power management on some platforms
Display of oem serial no
stanard behaviour of delete on NULL
Merge of v10 changes: Memory Management
| Status | Closed |
| Id | 79129 |
for OEM needs
Merge of v10 changes: Firmware File check when updating improved
| Status | Closed |
| Id | 79130 |
for OEM needs
Merge of v10 changes: More ciphers
| Status | Closed |
| Id | 79134 |
for OEM SRTP and other Certificate formats
Merge of v10 changes: Internal interfaces enhanced
| Status | Closed |
| Id | 79135 |
for OEM support
Merge of v10 changes: ZipZip Tone definition updates
| Status | Closed |
| Id | 79136 |
should overlay normal voice
Merge of v10 changes: Library update
| Status | Closed |
| Id | 79140 |
.
Merge of v10 changes: permit DHCP mode change between client/disabled without reset
| Status | Closed |
| Id | 79152 |
for OEM support
ASN.1 BER: Support for indefinite-length encoding
| Status | Closed |
| Id | 79260 |
needed for PKCS#12 import
only decoding
ASN.1 BER: Support for constructed octet strings
| Status | Closed |
| Id | 79261 |
needed for PKCS#12
only decoding, only one level of nesting
IP810: Config Wizard
| Status | Closed |
| Id | 79284 |
Mechanism needed to add platform specific files (IP810 uses same firmware as IP6010, IP3010, ...)
Merge of v10 changes: Support for SIP specific phone features
| Status | Closed |
| Id | 79306 |
for OEM support Status: checked in to 9.00
phone: permit to select the directories to be searched on indirect dialing
| Status | Closed |
| Id | 79307 |
if there is for example a huge external directory used for inbound name resolution and dialing is restricted to internal partners it may be hard to find internal numbers via combined directory search.
config add PHONE DIR-UI /dial-mask <selection>
selection:
0x02 - local directory
0x04 - PBX directory
0x08 - external directory
0x06 = local + PBX
0x0E = local + PBX + external
IP-DECT: Signal waiting calls to handset
| Status | Closed |
| Id | 79349 |
Was only acoustic information.
IP-DECT: LDAP phonebook (IP1202)
| Status | Closed |
| Id | 79483 |
The new feature LDAP phone book for the IP1202 is added now.
IP-DECT: Three party conference with innovaphone PBX
| Status | Closed |
| Id | 79503 |
Now it is possible to make a three party conference with DECT handsets with an innovaphone PBX (an innovaphone device with the CONF interface). The conferencing unit must be configured in the DECT master. The conference call is established with the feature code 'R' + '3'.
This fix also includes a rework of the DECT radio module. It can handle more than one waiting or hold call now.
Gateway: 'cn' attribute for test interfaces
| Status | Closed |
| Id | 79506 |
The 'cn' attribute is included in the test interfaces SIG0/1 for internal tests now.
V9 Hotfix 12 (9061009)
Changes included in Version 9 hotfix12 Definition
PBX: CFNR at Gateway object with incomplete destination
| Status | Closed |
| Id | 79605 |
A CFNR at a gateway object is executed if there is no registration. Any additional digits dialed should be added to the CFNR destination. This did not work if the original CFNR destination was incomplete and only completed with additional digits dialed.
PBX Waiting: Filter did not work anymore for CFNR
| Status | Closed |
| Id | 79610 |
Collateral damage from
fix: #75465: PBX: CFNR Loop check detected loops that weren't
LDAP Server caused crash on port-scanner attack
| Status | Closed |
| Id | 79634 |
LDAP Server caused crash on port-scanner attack
myPBX: Quotes in connected names caused java script errors
| Status | Closed |
| Id | 79636 |
When the connected name of a call contained the ' character there was a script error.
PBX: Status displayed on boolean function key could be wrong
| Status | Closed |
| Id | 79643 |
In case of severe network problems, it could happen that the status displayed on a Boolean function key was wrong and was only corrected when the boolean status changed.
DHCP Server Identifier was cleared after editing the DHCP-Server page
| Status | Closed |
| Id | 79651 |
The value of "IP4/ETHx/DHCP/Server Identifier" was cleared when the OK or Renew button was pressed on the "IP4/ETHx/DHCP-Server" page. This bug was introduced with V9hotfix5.
SIP: Wrong Contact-URI in outgoing NOTIFY
| Status | Closed |
| Id | 79700 |
Contact-URI should match the Request-URI of the SUBSCRIBE.
Also the Message-Account URI in "simple-message-summary" was wrong as result of the wrong Contact-URI.
Config: Could not dynamically set or reset /trace on the LICENSE module
| Status | Closed |
| Id | 79709 |
Missing handling of MODULE_UPDATE.
PBX: Allow call completion only if recall is not prohibited by CFU
| Status | Closed |
| Id | 79739 |
A CFU loop results in a rejection with busy. A subsequent call completion attempt was allowed and a recall possible was signaled right away. This was very confusing.
SIP: Handling of Call Pickup
| Status | Closed |
| Id | 79741 |
Failed to handle INVITE with Replaces as Call Pickup.
Must redirect Pickup call to alerting party.
myPBX: Problem with checking browser capabilities in Firefox
| Status | Closed |
| Id | 79757 |
When checking the supported browser features, Firefox thows an uncaught exception if cookies are deactivatd by the user. Therefore myPBX is stuck in the "Loading" screen instead of displaying a configuration hint.
IP241,IP222,IP232: Show lengthy number information on Partner fkey
| Status | Closed |
| Id | 79779 |
Toggle between 'show head' and 'show tail' presentation.
SIP: DNS resolving of STUN server failed
| Status | Closed |
| Id | 79788 |
Wrong STUN server port used (5060 instead of real port).
Fax: Channel memory leak
| Status | Closed |
| Id | 79815 |
Memory leak is fixed in fax channel now.
PBX: Export to CSV, utf-8 byte order mark was missing
| Status | Closed |
| Id | 79844 |
Without the utf-8 BOM the file was not correctly interpreted as utf-8 by Microsoft Excel and other applications.
PBX: For registrations containing name and number, use name to identify the device
| Status | Closed |
| Id | 79846 |
It did depend on the sequence of the name and the number. If the number was first, the first device was selected and the name was ignored. Now the name is used to select the device regardless of sequence.
This is a problem with endpoints which always send name and number for registration.
SIP: CLEARMODE does not work since Hotfix10
| Status | Closed |
| Id | 79858 |
Invalid SDP answer is send:
v=0
o=- 4 3 IN IP4 10.28.108.8
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 0 RTP/AVP 125 101
a=rtpmap:125 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000
memory violation in http client when processing an URI > 8000 byte
| Status | Closed |
| Id | 79860 |
happens when the digest authentication header is constructed
Status:
checked in to 10.00, 9.00
PBX Broadcast: Forking/Mobility configured at Broadcast object did not work
| Status | Closed |
| Id | 79863 |
Mobility can be used now for forking with delay
Potential trap in packet libraray
| Status | Closed |
| Id | 79866 |
Ther was a not obvious size limit for packet::put_head and packet::put_tail, which caused a trap if the size was exceeded. This could happen if a CDR exceeded a certain size.
SIP: Domain Name System (DNS) names compared case sensitive
| Status | Closed |
| Id | 79887 |
Domain Name System (DNS) names are "case insensitive"
SIP: Interworking with "AUDC-IPPhone"
| Status | Closed |
| Id | 79892 |
RFC-3551 4.5.2
Even though the actual sampling rate for G.722 audio is 16,000 Hz,
the RTP clock rate for the G722 payload format is 8,000 Hz because
that value was erroneously assigned in RFC 1890 and must remain
unchanged for backward compatibility. The octet rate or sample-pair
rate is 8,000 Hz.
Codec description must be: a=rtpmap:9 g722/8000
but "AUDC-IPPhone" sends: a=rtpmap:9 G722/16000
in SDP offer.
SIP: Problems with DNS resolving of proxy adresses
| Status | Closed |
| Id | 79907 |
If resolving of the primary proxy failes, the backup proxy is never resolved.
IP-DECT: Reset required if Radio password changed
| Status | Closed |
| Id | 79929 |
Now reset required is shown if the Radio password for the Master registration is changed.
Phone: PBX (operator) initiated outbound call was connected but mute when another call was ringing on phone already
| Status | Closed |
| Id | 79943 |
the phone was not switched from ring mode to handsfree/headset mode
SIP: Interworking with HD audio client
| Status | Closed |
| Id | 79953 |
Support for unknown codecs.
Passing transparentley.
Phone: Out-Of-Memory-Trap
| Status | Closed |
| Id | 79980 |
If LOG server is configured but not reachable
the device will buffer arising LOG entries until a limit of 300kB.
This limit was to high for old black/white telephones.
Now not more then 1% of DRAM size is used for LOG buffer.
PBX: When editing a Node object it was changed to a PBX object
| Status | Closed |
| Id | 79982 |
This was a collateral damage of fix: #78878: PBX: When doing a show users, unnecessary data was sent to the browser
ISDN: Problem configuring negative volume levels
| Status | Closed |
| Id | 79992 |
Collateral Damage from
79028: IP-DECT: Feature Code for calling predefined service URIs
Dect: AD replication makes IPEI disappearing on all users
| Status | Closed |
| Id | 80015 |
Side effect of a previous fix #72672 aiming at ad-replicated and deleted objects coming back to life.
myPBX: Setting CFx with boolean did not work in IE8
| Status | Closed |
| Id | 80018 |
Using Internet Explorer 8 the selected Boolean object was not saved.
Internet Explorer 8 requires option tags to have a value attribute.
IP-DECT: Handover with IP1202
| Status | Closed |
| Id | 80022 |
Only one handover is possible since the last fix with the IP1202. It is fixed now.
Name and Number configured for a registration must be completely replaced by registration result
| Status | Closed |
| Id | 80039 |
If for example Number and Name is configured but the registration result provides the Number only, the configured Name must not be used in further signaling operations (diversion queries ...)
Status:
checked in to 10.00, 9.00
PBX: Possible trap on calls from misconfigured nodes/PBXs (node parent loop)
| Status | Closed |
| Id | 80093 |
If a node or PBX is configured with a parent node configured to itself in the most simple case, a call from an endpoint configured for this node to a destination which cannot be found in this node, will cause a trap.
This is a collateral damage of fix: #79317: PBX: Local objects could not be called from Nodes with escapes as expected
IP1060 IP3010 IP6000 IP6010 IP22 IP24 IP28 IP302 IP305 IP222 IP232 IP241: receveid RTP packets limited to 480bytes
| Status | Closed |
| Id | 80111 |
increased to 640 Bytes to allow G711 80ms.
DHCP Survivability mode doesn't work after a DHCP restart
| Status | Closed |
| Id | 80112 |
``Survivabilityïï mode is used by WLAN phones. In this mode the settings of the 'saved lease' (IP addr, mask, ...) are used until a fresh lease is received. The 'saved lease' is the last lease received from a server, it is kept over a reboot.
When WLAN coverage is lost for a while and then regained a DHCP restart is requested to get a fresh lease from a server in a possibly different network. If this happened while using the 'saved lease' the phone lost it's (saved) IP address.
Status:
checked in to 10.00, 9.00
AC-DSP3: Switch trace off if the DSP Host interface shows an error
| Status | Closed |
| Id | 80130 |
SIP: Keep backup registration while calls are active
| Status | Closed |
| Id | 80137 |
Terminating backup registration will active calls (through backup system) get disconnected.
SIP: Content of Allow-Events header must be treated case-insensitive
| Status | Closed |
| Id | 80173 |
Event names are case insensitive.
IP-DECT/Analog Features: Call Park should be done on the last active call
| Status | Closed |
| Id | 80209 |
Szenario is an active call, then a waiting call comes in, which is accepted, then call park is executed. This call park should be done on the accepted waiting call and not the original.
IP-DECT/Analog Features: Call completion
| Status | Closed |
| Id | 80212 |
The type of the call is changed back to normal state if the call completion is executed, and facility conversion is added for the call completion state.
This fixes the reusing of features for a call completion callback call, used if IP-DECT/analog features are enabled. This also fixes missed remote hold and retrieve events to the gatekeeper.
SIP: Handling of 488 for encrypted media
| Status | Closed |
| Id | 80246 |
If SRTP offer is rejected by remote endpoint with "SIP/2.0 488 Not Acceptable Here"
we should re-try offer without encryption keys.
Phones: Show 'tel' presence on configuration screen
| Status | Closed |
| Id | 80274 |
Show latest presence on 'Presence' fkey,
but show 'tel' presence on phone config menu.
PBX-Waiting: Calls to a Waiting queue object in altering state could not be cleared with SOAP
| Status | Closed |
| Id | 80291 |
A SOAP UserClear issued for the call monitored at the Waiting Queue object didn't do anything
Phones: Sometime fkey could not be deleted
| Status | Closed |
| Id | 80389 |
Sometime fkey could not be deleted when using the fast edit mode (long fkey press).
IP-DECT: Handset display
| Status | Closed |
| Id | 80402 |
Names with special characters are not correctly shown in the handsets. This is fixed in the IP1200 now.
IP210 IP230 IP240: Handset receiver volume increased, especially at lower frequencies
| Status | Closed |
| Id | 80403 |
For better performance with jinlida receiver.
Gateway: Pass through ctSetup facility
| Status | Closed |
| Id | 80437 |
Lync sends diverting party information inside Referred-By header.
Referred-By is interworked to ctSetup facility.
ctSetup facility needs to be passed through by Gateway application.
Linux: IP address with external DHCP server
| Status | Closed |
| Id | 80471 |
Assigning a IP address to Linux by a external DHCP server is not working if the network interface which is used is configured with a fix IP address (DHCP disabled). This is fixed now.
IP232: Invalid text on second line of fkey label
| Status | Closed |
| Id | 80484 |
Invalid text on second line of fkey label when changing registrations.
PBX: Routing problem from sub-slave to master to object in sub-slave node, but registered to master
| Status | Closed |
| Id | 80591 |
Collateral damage from #77874: PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX
SIP: Calls may remain in clearing state
| Status | Closed |
| Id | 80623 |
SIP calls may remains undeleted.
SIP: Code optimization
| Status | Closed |
| Id | 80635 |
Reduce number object constructions/destructions during message encoding.
PBX: Update of conference id did not always work after multiple transfer accross PBXs
| Status | Closed |
| Id | 80656 |
A scenario which did not work was A calls B, B does consultation to C, B Transfers, C does consultation to D, C transfers with B on different PBX then A.
After this the conference id on the call on A should be identical to the conference id on D. This was not the case.
SOAP/TAPI applications which are keeping track of transfered calls could have a problem with this.
AD Replication: LDAP filter encoding failed, when Poll Timer was configured
| Status | Closed |
| Id | 80658 |
AD Replication: LDAP filter encoding failed, when Poll Timer was configured
PBX: More consistent use of conferenceId after transfer
| Status | Closed |
| Id | 80660 |
The conference ID is used (SOAP/TAPU, CDRs) to associate different call legs to the same call. After a transfer two calls, which have been seperate are connected, so one of the call legs has to change its conference ID, so that the resulting call has a single conference ID again.
There was a complicated logic implemented in the PBX to decide which conference ID should be used, this is now changed to a simple logic: The conference ID of the call on which the transfer is performed, is used.
Example:
A calls B, B does a consuktation to C, and B transfers A to C - This means the transfer is performed on call leg A, so the conference ID of the original call A-B is used for A-C
Gateway: Routing of incoming SIP calls may not work
| Status | Closed |
| Id | 80709 |
... because the To-URI is not reduced to a called number (CDPN).
Gateway routing is based on CDPN and does not work for URIs.
Before this fix:
.LOG CALL 2 Alloc
.LOG CALL 2 A:Call -> / GW8::->*::
.LOG CALL 2 B:Call :081604998@212.13.249.90->:018108680@192.168.5.230 / GW8::->GW1::
After this fix:
.LOG CALL 1 Alloc
.LOG CALL 1 A:Call -> / GW8::->*::
.LOG CALL 1 B:Call 081604998->018108680 / GW8:081604998:->GW1:018108680:
SIP: Memory leak when closing signaling interface
| Status | Closed |
| Id | 80752 |
Memory leak when closing signaling interface while DNS request is pending on a call.
Phone: Message function key cannot be configured
| Status | Closed |
| Id | 80766 |
Configuration is not saved
SIP: Diversion information was wrong on incoming SIP calls
| Status | Closed |
| Id | 80812 |
Parsing of History-Info header was wrong, but only if multiple History-Info headers were present.
PBX: Routing problem with nodes/escapes/slaves with calls to extern numbers in other PBX defined nodes
| Status | Closed |
| Id | 80853 |
Could happen that the number dialed when sending the call to the extern interface was not adjusted correctly
Build Number format changed
| Status | Closed |
| Id | 80925 |
The build number of the hotfixes changes from the 90600.xx format to the 9.061xxx format. This is due to organizational changes without any other significance.
IP-DECT: Memory leak for SIP calls
| Status | Closed |
| Id | 80937 |
buffer for received name-id was not freed. Could happen with other facilities as well.
SIP: Transcation handling was wrong
| Status | Closed |
| Id | 80989 |
Double delete on a INVITE client transaction object.
0:0027:988:5 - SIP_TAC_INVITE:serial::delete(805b0c80) caller=800440ec
PBX: Adjusting called number, when sending call to extern
| Status | Closed |
| Id | 81005 |
depending on the node of the extern object and the called node, the called party number has to be adjusted (escapes added, prefixes added/removed). This did not work unders some conditions.
myPBX: Syslog for sessions and application sharing
| Status | Closed |
| Id | 77289 |
Syslog entries should be useful for debugging session timeouts and problems with the external application sharing solution.
SIP: Pass display names of <dialog-info> to phoneapp
| Status | Closed |
| Id | 79543 |
Pass display names of <dialog-info> to phoneapp.
Gateway: Overlap dialing timeout configurable
| Status | Closed |
| Id | 79639 |
The default of 4s is not good for all applications. Now configurable between 500ms and 6000ms.
IP-DECT: OEM Configuration of Cisco Features changed
| Status | Closed |
| Id | 79684 |
should depend on installed license
myPBX launcher: Automatically move main window into visible area of screen
| Status | Closed |
| Id | 79697 |
The window might be outside the visible area if the user changed the screen resolution or disconnectes one of the screens.
If the main window is not visible on any screen, it is now moved to the center of the primary screen, when the user clicks the tray icon.
PBX-SOAP: FindUser optional argument 'nohide'
| Status | Closed |
| Id | 79734 |
Needed if the FindUser is used to look for users which have the 'Hide from LDAP property' set. Status:
PBX Mobility: Pick mobile call on fixed phone
| Status | Closed |
| Id | 79794 |
by dialing number of Mobility object on fixed phone
Gateway: Allow setting of system time from ISDN time
| Status | Closed |
| Id | 79889 |
In ISDN connect messages a public network is sending a local date and time. By a configuration option this can now be used to update the system time
IP-DECT: New Master/Radio behavior for license incompatibility
| Status | Closed |
| Id | 79914 |
The behavior of the DECT Master is changed, if an unlicensed Radio tries to register in. This is only used for an OEM license model.
PBX: Better handling of presence information without activity
| Status | Closed |
| Id | 80006 |
This type of presence information could contain a note which refers to the future
RPCAP trace: Indicate transmit/received packets by setting the remote mac address to 00-90-33-00-00-00
| Status | Closed |
| Id | 80152 |
When reading a trace it is currently not obvious if a packet is sent or received, we need to find out the devices IP address, e.g. by reading the config. If the devices on MAC adress is used a source only if a packet is sent and as destination only if a packet is received this process is simplified.
This is now done for UDP/TCP traces!
SIP: Support for Alert-info:<Bellcore-dr1>
| Status | Closed |
| Id | 80174 |
Proprietary tagging of internal calls
phone: ip222,ip232: more USB headsets supported
| Status | Closed |
| Id | 80224 |
for a complete list see http://wiki.innovaphone.com/index.php?title=Reference9:Concept_USB_Headset
Status:
checked in to 10.00, 9.00
Phones: Switch for phoneapp to disable auto-answer
| Status | Closed |
| Id | 80233 |
Disable/enable auto-answer support on phoneapp level.
PBX: DECT attributes for DECT security
| Status | Closed |
| Id | 80300 |
In preparation for the new DECT feature DECT security there will be new attributes for the endpoint data which must be taken over. With this fix the innovaphone PBX supports the new attributes if the user is edited.
phone: ip222, ip232: Jabra USB Headset feature "Reject incoming call" supported now
| Status | Closed |
| Id | 80401 |
A double tap on talk button rejects a ringing call with 'busy'
IP-DECT: Signal waiting calls to handset
| Status | Closed |
| Id | 80408 |
New event to the DECT system to add a waiting call to the call list.
This is step two of the implementation and related to the case #79349.
IP-DECT: Anonymous endpoint information
| Status | Closed |
| Id | 80421 |
Information about anonymous endpoint is added to the event to the DECT system. In preparation for the new feature "easy subscription".
IP-DECT: DECT security
| Status | Closed |
| Id | 80424 |
Some changes in preparation for the new feature "DECT security". This feature will be available later.
Phone: Accept MWI from Exchange Server
| Status | Closed |
| Id | 80446 |
MS Exchange Server sends unsolicited NOTIFY(message-summary) to served user
with served user's number as destination and origin.
But phones expect to receive MWI message center number as origin.
MWI fkey would not light up.
SIP: New config option /take-zero-addr-for-hold
| Status | Closed |
| Id | 80516 |
From now on "c=IN IP4 0.0.0.0" is no longer accepted as hold signaling.
Config option /take-zero-addr-for-hold is introduced to get back old handling.
IP-DECT: User log in, endpoint data added
| Status | Closed |
| Id | 80620 |
During the user log in with a change of the IP-DECT Master the endpoint data (product id/software version) is sent to this new one now.
SIP: Pass display information to application
| Status | Closed |
| Id | 80632 |
Pass display information received in Call-Info header in 200/OK for BYE to app.
IP-DECT: Idle display update
| Status | Closed |
| Id | 80654 |
Handling of idle display update message in the call release message is added. Used with an OEM PBX.
myPBX launcher: Configurable hotkey action
| Status | Closed |
| Id | 80684 |
Options are:
- Copy selected phone number to myPBX
- Show myPBX
myPBX launcher: Autocomplete configured URLs
| Status | Closed |
| Id | 80689 |
If the user configures just an IP address, it shall be replaced by the full myPBX URL.
For example 192.168.0.10 will be replaced by http://192.168.0.10/PBX0/MY/client.htm.
V9 Hotfix 13 (9061024)
Changes included in Version 9 hotfix13 Definition
IP222 equalizer update
| Status | Closed |
| Id | 80004 |
- handsfree speaker equalizer enabled
- handset mic and receiver equalizer smoothed
- ADC gain reduced, input gain increased ( after ec ) to avoid clipping
- halfduplex mode disabled
IP222 IP232 IP241: repeated ethernet link status 1000M wrong
| Status | Closed |
| Id | 80029 |
.
IP222 IP232 : Default LCD backlight standby brightness reduced
| Status | Closed |
| Id | 80031 |
IP241 handsfree equalizer update, IP222 handsfree mic equalizer update
| Status | Closed |
| Id | 80157 |
- IP241 handsfree speaker equalizer enabled
- IP241 handsfree micro equalizer enabled
- IP222 handsfree micro equalizer enabled
- All three filter calculated with measured frequency response
- IP241 ADC gain reduced, input gain increased ( after ec ) to avoid clipping
phone: ip222, ip232: Pressing Talk button on USB Headset when there is an active and a held call does not transfer
| Status | Closed |
| Id | 80413 |
Instead of transfer the active call is disconnected and the held call is retrieved Status: checked in to 10.00
IP241,IP222,IP232: Show name of diverting party in incoming calls
| Status | Closed |
| Id | 80839 |
Show name of diverting party in incoming calls (not only number).
PBX-SOAP: UserRedirect - original called number got lost
| Status | Closed |
| Id | 80854 |
For example if an application used an Waiting Queue object to monitor for incoming calls and redirected these calls to agents. The agent receiving the call could not see if the call was diverted to the waiting queue already.
SIP: Send 200/OK for MESSAGE(text/plain) when accepted by application
| Status | Closed |
| Id | 81017 |
Application gives "NormalCallClearing" to SIP stack.
SIP stack better sends "200 OK" instead of "603 Decline".
SIP: Un-escape content of XML elements
| Status | Closed |
| Id | 81019 |
XML element content requires some resevered characters to be escaped (<>).
These escape sequences (> or <) must be un-escaped onthe receiving end.
IP241,IP222,IP232: Two status symbols may overlay each other
| Status | Closed |
| Id | 81027 |
Symbol for "Call diversion" and symbol for "PIN-locked" overlay each other.
Hide "Call diversion" as long as phone is PIN-locked.
SIP: Handling of multiple 401/407 responses
| Status | Closed |
| Id | 81045 |
Implemented handling of up to 3 401/407 responses per transaction.
SIP: Trap while releasing call
| Status | Closed |
| Id | 81089 |
Timer fires during call release and causes trap.
H.323: Memory leak when sending special OEM H.235 key elements
| Status | Closed |
| Id | 81160 |
Only happens if Avaya SRTP is enabled in OEM products
H.323: Potential Trap in case of high load and media renegotiation/call clearing collision
| Status | Closed |
| Id | 81173 |
There is a very unlikely situation when media-renegotiation is started and then the call is cleared, which could cause a message related to the media-renegotiation to be sent to a already deleted call object. High load could make this situation more likely.
IP241 handset equalizer
| Status | Closed |
| Id | 81174 |
Enable IP241 handset micro and speaker equalizer.
Same values as on IP222.
Reduces noise in G711 mode.
IP222: Handset receiver volume increased
| Status | Closed |
| Id | 81175 |
By 6db, so the volume similar to IP241, but still has low THD+N
IP1060/3010/6010/22/24/28/302/305: sequence number to DSP changed from byte to word, caused jiter buffer warnings
| Status | Closed |
| Id | 81189 |
IP222/232/241/1060/3010/6010/22/24/28/302/305: min jitter buffer changed to 10ms
| Status | Closed |
| Id | 81197 |
was 35ms, now 10ms as on ip6000/800
IP-DECT: Duplicate call setup
| Status | Closed |
| Id | 81213 |
The IP-DECT Master sends in some circumstances a call twice to the same radio in the same time. This affects only the IP1202 and OEM devices, not the IP1200, and is fixed now.
IP-DECT: OEM configuration option
| Status | Closed |
| Id | 81214 |
The configuration format of an OEM configuration option was changed for the backward compatibility.
phone_orchid: micro mute when a waiting call was connected after an outbound call setup had been cancelled
| Status | Closed |
| Id | 81229 |
- A goes offhook
- B calls A, a waiting call from B is indicated on A
- A goes onhook, phone rings
- A goes offhook again and is connected to B
- A hears B, B doesn't hear A
SIP: Pass display information to application
| Status | Closed |
| Id | 81230 |
Pass display information to application when registration comes up.
IP0010 IP1060 IP3010 IP6010: During long reset the ready LED was blinking orange and the Ethernet LEDs were swaped
| Status | Closed |
| Id | 81236 |
Now the ready LED shows the green blinking during long reset.
The Ethernet LEDs are initialized directly after reset to overwrite the default setting that swaps link and speed.
X.509: Creating certificate containing IPv4 address did not work
| Status | Closed |
| Id | 81251 |
When creating a certificate or a request with an IPv4 address, the IPv4 address was mapped to an IPv6 address. This was not the expected behaviour.
172.16.10.32 -> ::ffff:172.16.10.32
Now both IPv4 and IPv6 addresses can be used.
ip222,ip232: waiting calls mute after having been accepted/connected on USB headsets, mainly seen with Jabra LINK 14201-30
| Status | Closed |
| Id | 81312 |
In some cases only the accepted call was mute and the next call was OK again but the Jabra LINK 14201-30 lost the USB connection in most cases.
Delaying the HID-commands sent to the headset solves this problem.
PBX: Objects list filter for numbers did not work correctly anymore
| Status | Closed |
| Id | 81376 |
only top level nodes were displayed
Collateral damage of fix
79982: PBX: When editing a Node object it was changed to a PBX object
SIP: Pass display information to application
| Status | Closed |
| Id | 81379 |
Pass display information received in REGISTER response to app.
IP241,IP222,IP232: Show H323-ID instead of Display Name if Display Name is not available
| Status | Closed |
| Id | 81419 |
Like on old b/w phones.
IP-DECT: Memory leak with special Cisco features
| Status | Closed |
| Id | 81463 |
Only with special OEM features.
SIP: Problems with DNS resolving of proxy adresses
| Status | Closed |
| Id | 81522 |
Not always re-tried when failed at startup.
IP241,IP222,IP232: Rendering errors on Fkey configuration screen
| Status | Closed |
| Id | 81556 |
When scrolling or leaving Fkey configuration screen.
List (Toggle) controls were not rendered correctly.
IP-DECT: Mobility Master registration
| Status | Closed |
| Id | 81571 |
The Mobility Master does not accept clients with the OEM name IP1202. This is fixed now.
HTTP: Possible trap on many simultaneous sessions
| Status | Closed |
| Id | 81597 |
Sorting the TCP sessions did not work correctly, which caused an assertion because it could happen that it was not possible to remove a session
IP-DECT: System GUI, disabled local coder options
| Status | Closed |
| Id | 81603 |
If the PARI function (only IP1202) of the IP-DECT Master is disable, configuration changes on the System GUI do not effect anything. The settings for the local coder are disabled on this GUI page now.
IP-DECT: System settings not to dynamically connected radios
| Status | Closed |
| Id | 81612 |
System settings should not be sent if the radio-master connection is a dynamic one (IP1202 only). This is fixed now.
Support for old versions of Jabra BIZ 2400 headset with 48 kHz output sampling frequency
| Status | Closed |
| Id | 81613 |
the newer versions come with the usual 16 kHz sampling frequency but have the same signature (vendor/product) as the older ones.
SoftwarePhone: Trap with trace and mute ringer event
| Status | Closed |
| Id | 81619 |
A trap occurs if the trace is enabled and the ringer mute option is toggled.
IP222 IP232 IP241: Force same speed of the switch ports for 1000M/100M scenarios
| Status | Closed |
| Id | 81693 |
In scenarios that operate one Ethernet port with 1000M and the other with 100M the switch througput was low.
Now the 1000M port is reconfigured to 100M, and the throughput is high.
SIP: Support for multiple audio media descriptions
| Status | Closed |
| Id | 81712 |
One without encryption and one with encryption:
v=0
o=OpenStage-Line_0 968610650 446118927 IN IP4 10.30.1.144
s=SIP Call
c=IN IP4 10.30.1.144
t=0 0
m=audio 5012 RTP/AVP 9 8 0 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv
m=audio 5010 RTP/SAVP 9 8 0 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ujVU8G6kgknZnPflRwx8tadNskkp9glas/DFCbC3
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:+uoFBNVVhDY5OOGvMOdAlvpxvc98hX/VeProhlwH
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv
IP232: Show more of collapsed fkeys
| Status | Closed |
| Id | 81807 |
140px when expanded.
100px when collapsed (80px before).
SIP: Double Replaces header in INVITE after REFER
| Status | Closed |
| Id | 81847 |
Reject by "Swyx IpPbxSrv/8.1.0.246" with "400 Bad Request(Multiple values in single-value header Replaces)"
PBX Mobility: No response for call to busy mobile phone, if no fixed phone
| Status | Closed |
| Id | 81850 |
If a user is called with mobility configured and no fixed phone and the mobile phone was busy, then the call did not complete and was hanging as if the number was not complete. The call should be answer with busy instead.
Chrome compatibility issue on PRI statistics page
| Status | Closed |
| Id | 81859 |
If clear was clicked an error message was displayed
PBX Admin UI: Not possible to change DECT parameters for users with cf/grp admin rights
| Status | Closed |
| Id | 81871 |
The user interface prohibited the editing of DECT parameters based on the rights of the object to be edited not based on the login
phone: ip222,ip232: when a call is released from remote while another call is waiting the waiting call cannot be accepted.
| Status | Closed |
| Id | 81907 |
When the call is released from remote a buys tone is generated for two seconds. Therafter phone rings to indicate that the waiting call can be accepted now.
When trying to accept this call by pressing the headset talk button the call was disconnected instead.
SIP: Memory leak in SIP stack
| Status | Closed |
| Id | 82058 |
Memory leak in SIP stack.
myPBX: Interface for tracing
| Status | Closed |
| Id | 80973 |
The web application can now write messages into the trace file of the myPBX launcher.
Oem Code: Accelerating Boot Snmp Traps
| Status | Closed |
| Id | 81132 |
Oem Code: Accelerating Boot Snmp Traps
PBX: Handling of call limits at PBX objects improved
| Status | Closed |
| Id | 81232 |
A call from a PBX, which is sent back to the same PBX is not counted anymore. This can happen because of node-extern.
Incoming calls at master, which are above the limit are rejected now. They can be rerouted on the slave with "Route Master calls if no Master to"
SIP: New config option for endpoints not refreshing their registration during call
| Status | Closed |
| Id | 81243 |
Interoperability:
New config option /keep-active-endpoints for endpoints not refreshing their registration during call.
AVM FRITZ!Box Fon WLAN 7270 v2 (UI) 54.05.21 (Apr 2 2012)
UI hint where to update Linux AP
| Status | Closed |
| Id | 81295 |
Changed a hint on the upload/update tab and added a link to the update/upload page on the firmware side.
SIP: New config option /no-certificate-check
| Status | Closed |
| Id | 81601 |
New config option to disable validation of remote certificate name
when opening outbound TLS connection.
IP-DECT: Own priority for idle display set by messages
| Status | Closed |
| Id | 81616 |
The idle display set by messages has got an own priority now.
SoftwarePhone: Support for Jabra PRO 930
| Status | Closed |
| Id | 81618 |
Support for Jabra PRO 930 with product id 0x1016 added.
SIP: New config option "No Inband Disconnect" on GW interfaces
| Status | Closed |
| Id | 81803 |
Introduced VOIP_OPTION_NO_INBAND_DISC.
Option was missing on Gateway interfaces.
SIP: New config option /product-id-format
| Status | Closed |
| Id | 81880 |
Change the User-Agent string from
User-Agent: (innovaphone IP232/10.00 dvl [90910/90879/501])
into
User-Agent: innovaphoneIP232x90910x501
with /product-id-format 1
phone: ip222,ip232: more USB headsets supported
| Status | Closed |
| Id | 81972 |
Sennheiser and some more Jabra Headsets added. Look for "(since V9hotfix13)" in
http://wiki.innovaphone.com/index.php?title=Reference9:Concept_USB_Headset
Debug information on assertion
| Status | Closed |
| Id | 81973 |
More debug information on default event handler.
phone: ip222,ip232: support for advanced USB headset functions (redial, reject call, accept waiting call)
| Status | Closed |
| Id | 81975 |
A lot of USB headsets generate special indications to request redialing of last number dialled, to reject a ringing call, to accept a waiting call and to put the active call on hold or to switch between an active and an held call.
V9 Hotfix 14 (9061046)
Changes included in Version 9 hotfix14 Definition
Incorrect disk usage calculation for more than 4GB
| Status | Closed |
| Id | 81209 |
The disk usage calculation was wrong, if more than ~4 GB of the card were used.
PBX Waiting: Call state shows "Disconnecting" after switch from announcement 1 to announcement 2
| Status | Closed |
| Id | 81370 |
Announcement disconnect changed state
PBX: Switch from Music on Hold to inband ringback on Alert after unpark oder transfer
| Status | Closed |
| Id | 81407 |
To give the caller feedback that soon somebody may answer the call
PBX: Don't forward in-band info indicator if no media channel
| Status | Closed |
| Id | 81879 |
Not good to indicate in-band info available if no media channel can be negotiated
SIP: Trap when configuring user presence
| Status | Closed |
| Id | 81996 |
Trap may occur when configuring user presence.
HTTP-Server: Configuration of "Public compact flash access" did not work for all cases
| Status | Closed |
| Id | 82064 |
E.g. /DRIVE/CF0/Neuer Ordner/ does not work, because HTTP request contains escaped sequences.
phone: ip222,232.241: no notification tone on a successfull redial attempt
| Status | Closed |
| Id | 82166 |
On a failing or unanswered call the menu key opens the "Recall" menu. If "Redial" is selected the call is automatically redialed for 20 minutes in intervals depending on the result of the previous attempt. On success the user should be notified about the connection.
IP-DECT: Trap during subscribing handsets
| Status | Closed |
| Id | 82190 |
A trap occurs during subscribing handsets on the IP1200. This is a fix for the previous fix #80424 and fixed again.
IP-DECT: Wrong release code
| Status | Closed |
| Id | 82195 |
The IP-DECT sends the wrong release code "User not reachable" if the handset was not connected and the remote parts disconnects. This is fixed now.
IP-DECT: No delay with semi-attended call transfer and SIP
| Status | Closed |
| Id | 82197 |
If the SIP protocol is used and the user do a semi-attended call transfer, the call transfer is directly confirmed again. The semi-attended call transfer is stored in the base station and executed as an attended call transfer if the target party connects.
IP-DECT: Display info with failed user log-in
| Status | Closed |
| Id | 82200 |
If the user log-in fails, the cause is shown in the display.
IP-DECT: Short tone info by PBX only one time
| Status | Closed |
| Id | 82204 |
Tone information with defined length and requested by the gatekeeper is played only one time to the handset. This is used by an OEM PBX and fixed now.
IP-DECT: Hanging aborted semi-attended call transfers with SIP
| Status | Closed |
| Id | 82205 |
If the SIP protocol is used, a semi-attended call transfer is done by the user and the call transfer can not be executed, the remaining call party is not disconnected. This is added now.
IP-DECT: LDAP server GUI description
| Status | Closed |
| Id | 82258 |
The LDAP server GUI description is wrong and corrected now. (The LDAP server can not be a HTTP server.)
DHCP: Name registration at WINS was not refreshed as requested by TTL in registration response
| Status | Closed |
| Id | 82289 |
If the DHCP-client gets a lease containing a WINS-server address and a NETBIOS node type P or M (1 or 2) the client tries to register it's NETBIOS-name (ipxxx-xx-xx-xx) with the WINS-server. The TTL returned by the server in the registration response determines when a name refresh has to be sent.
Phones: Presence info during ringing state may show garbage data
| Status | Closed |
| Id | 82306 |
In case a presence update arrives at the phone while phone is in ringback state.
Have been observed in conjunction with call forking with mobility only.
myPBX: Support contact names containing a single quote
| Status | Closed |
| Id | 82323 |
Configuring contacts with a H.323 id containing a single quote (') caused script errors in the web application.
PBX: RTP-DTMF was disabled by Voice Mail object during re-negotiation
| Status | Closed |
| Id | 82332 |
RTP-DTMF acc. to RFC-2833 was disabled by Voice Mail object during re-negotiation.
Gateway CDR with '0. 0' charge amount
| Status | Closed |
| Id | 82359 |
Should be '0.00' instead
phone: call completion did not recover when DND(busy) was set at the phone requesting the call completion
| Status | Closed |
| Id | 82390 |
when DND(busy) was set on the phone reqesting the call completion and was cleared some time later a "Recall possible" was not indicated anymore although a pending call completion was indicated on the called phone.
ipv6: memory leak when sending fragmented packets
| Status | Closed |
| Id | 82394 |
H.323:No Media for calls with reverse media to a H.323/SIP exclusive Code Media Relay interface
| Status | Closed |
| Id | 82408 |
The execlusive coder/media relay config is used to avoid media negotiation problems with carrier which do not support media renegotiations. In case of a call with reverse media to such an interface, this did not work. This happens for example if a CFNR is configured at a Waiting Queue which redirects a call, which received an announcement from the Queue to such interface.
IP-DECT: DECT Users Administration Log-in
| Status | Closed |
| Id | 82422 |
The GW-DECT module's users administration log-in is fixed.
Kerberos: Allow editing multiple fields in admin UI
| Status | Closed |
| Id | 82425 |
This is a fix for the page General/Kerberos in the admin UI.
Editing more that one item at a time could result in strange effects. Now it is possible to edit multiple fields.
phone_inca: when a call completion was set up with CLIR active the called party was not displayed on a possible recall
| Status | Closed |
| Id | 82435 |
A sets CLIR, A calls B, B is busy
A sets a CCBS request via Menu/Recall
B goes on Hook
A rings and sees 'anonymous' instad of the number of 'B', status line is empty (should show "Recall possible")
IP-DECT: Display info with remote control call
| Status | Closed |
| Id | 82444 |
The display info shown after accepting a remote control call is fixed now.
IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: enable modembypass
| Status | Closed |
| Id | 82458 |
Modembypass is enabled on all calls with disabled T.38 and coders G711A or G711U.
Switch to modem bypass is indicated in the trace by "switch to modembypass".
The feature can be disabled with http://addr/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl
Modembypass works best if T38 is disabled on both sides.
If T38 is enabled on the called side the CED may trigger a T38 session, this changed back to voice and modem bypass is enabled (if G711 is active). The first modem tone is interupted, but we still have modembypass on both sides.
If T38 is enabled on the calling side the calling side stays on regular G711.
IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: reduce probability of false DTMF detection
| Status | Closed |
| Id | 82470 |
Change DTMF signal to noise ratio from 12db to 18db
IP800 IP6000: reduce probability of false DTMF detection
| Status | Closed |
| Id | 82471 |
Change DTMF detector signal to noise ratio from 12db to 18db.
Gateway: 'Enable PCM' option added at the CONF interface of the IP800
| Status | Closed |
| Id | 82481 |
The option 'Enable PCM' is available at the CONF interface of the IP800 now.
Gateway: PCM mode of the CONF interface fixed
| Status | Closed |
| Id | 82482 |
The PCM mode of the CONF interface is not activated for a call even though it is configured and possible. This is fixed now.
IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: T38 trace flag added
| Status | Closed |
| Id | 82484 |
Allows to trace T38 connection on the PCM port and on the DSP host interface.
Use this if fax modem problems are suspected.
Enable at http://addr/debug.xml at trace->T38 trace.
Gateway: Routing problem with blockdial route and following matching non-blockdial incomplete routes
| Status | Closed |
| Id | 82486 |
Example:
Blockdial Route 00->
After this a non-blockdial route with 0...
If now a number of 001 was dialed, the first route should match and after the enbock dialout the call should be sent to the destination of the route. Instead the call was rejected with "no destination found"
failure of analog ports of ip28
| Status | Closed |
| Id | 82488 |
ip28 analogue ports do not react to incoming calls and hook-off. Problem could only be solved by reset.
Debug "HTTP_GET LOG_HTTP.1: retry, authentication failed" removed
| Status | Closed |
| Id | 82499 |
Phone: Display text received with BYE
| Status | Closed |
| Id | 82525 |
Pass display text to phoneapp.
SIP: Trap during call handling
| Status | Closed |
| Id | 82544 |
Trap during call handling
IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: update to DSP code 660.27
| Status | Closed |
| Id | 82551 |
660.27 is the released version on the audiocodes web.
Fax relay and modem bypass improved.
IP222 IP232 IP241: update to DSP code 660.27
| Status | Closed |
| Id | 82554 |
660.27 is the released version on the audiocodes web.
Improved acoustic echocanceller.
phone: call forwarding options offered even when call forwarding was not possible
| Status | Closed |
| Id | 82567 |
Call forwarding is not supported when running SIP. But when the menu key was pressed after entering a number call forwarding options were offered (happened with the primary registration only).
PBX Mobility: Trap in case of Transfer of a call from a mobile endpoint to another mobile endpoint
| Status | Closed |
| Id | 82584 |
The Trap happens in the following call scenarion
- Mobile endpoint calls in, using mobility two-stage dialing
- call is accepted at local phone
- on local phone a consultation call is initiated to another user with mobility
- when mobile phone rings, a transfer is initiated on local phone
- the called mobile phone accepts the call
- the trap happens when the called mobile phone hangs up
There could be other call scenarions where the trap happens as well
SIP: SRTP key exchange failed
| Status | Closed |
| Id | 82616 |
Bug in base64 decoding of SRTP key.
PBX Trunk: Number to Name Feature did not work for calls to busy endpoints
| Status | Closed |
| Id | 82619 |
Such a call was not rejected with cause 'User Busy', but was just hanging.
PBX Mobility: Call to mobile phone was sent with invalid diverting information
| Status | Closed |
| Id | 82622 |
A call to a mobile phone is sent with a diverting leg2 info, which means, the call contains the information, that it was diverted by the called user to the mobile phone. So in theory this could be displayed on a mobile phone.
The coding of this information was wrong and created interop problems with some networks.
SIP: Trap on subscription handling
| Status | Closed |
| Id | 82623 |
Trap on RAS_REGISTRATION_VERIFY between sending SUBSCRIBE and receiving 200/OK.
SIP: Removed cisco-special retrieve signaling
| Status | Closed |
| Id | 82637 |
Removed cisco-special retrieve signaling.
phone: ip222,ip232: Plantronics APU70 - Savi 7xx - Radio Link not cleared on release from remote
| Status | Closed |
| Id | 82654 |
On a release from remote for a call set up by pressing the Talk button (headset or base) the Radio Link between base and headset was not cleared until the Talk button was pressed again.
SIP: Mobility did not work due to RTP-DTMF
| Status | Closed |
| Id | 82674 |
DTMF must be passed through signaling channel to get mobility working.
Suppress RTP-DTMF capability in SDP answer also.
IP3010 IP810: number of DSP channels and number of conference channels was wrong.
| Status | Closed |
| Id | 82675 |
Now the IP810 shows 20 DSP channels and 30 conference channels.
Now the IP3010 shows 42 DSP channels and 60 conference channels.
SIP: Restart NAT discovery if failed
| Status | Closed |
| Id | 82676 |
Restart NAT discovery if failed
IP222 IP232 IP241: Reduce DTMF level
| Status | Closed |
| Id | 82685 |
During DTMF receive and transmit levels similar as on IP240 are used
-10db level
0xc0=208--> 22db attenuation
also insgesamt ein level von -32db ( bei Vollauschlag ) oder -29dbm.
Der alte Wert beim ac_phone3.cpp war -9db
Weitere Diskussion:
Es gibt den Fall das inband DTMF zum IP Netz geschickt wird, da gabs in Fall 59846 die nderung zum IP mit LEV=0x28 -->-10db und Attenuation 0xff-->18db, also mit -28db zu senden.
Da das gut funktioniert und die beiden Pegel nicht so unterschiedlich sind unde der ac_dsp3 nicht unterschiedliche Pegel zum IP und zum Codec kann nehmen wir die -28db=-25dbm.
Laut www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-Q.24-198811-I!!PDF-E&type=item Tabelle A-1 sind die -25dbm am unteren Ende, da ist vom Wert A
-22..-30 je nach Administration die Rede. Wahrscheinlich ist das kein Problem,
bei Audiocodes (ac_dsp2) kann man die sensitivity von -28 bis -38db einstellen (DTMF_DETECTION_ENERGY_THRESHOLD__28dBm)
-->
IP222,IP232: Going offhook in call list always dials first list entry (not touched list entry)
| Status | Closed |
| Id | 82732 |
Going offhook in call list always dials first list entry (not touched list entry)
Memory leak after Firmware or Bootcode download
| Status | Closed |
| Id | 82740 |
After a Firmware or bootcode download via Maintenance/Download the associated command processor instance was not released.
IP-DECT: Call transfer compatibility with SIP
| Status | Closed |
| Id | 82742 |
For compatibility reasons with the SIP protocol the call transfer initiate result message should not be sent until the connect message is received. This is changed now again.
SIP: Trap if multiple calls arrive at same time
| Status | Closed |
| Id | 82743 |
Trap if multiple calls arrive at same time.
SIP: Bug in digest authorization
| Status | Closed |
| Id | 82761 |
Sometimes wrong method is used in digest calculation.
phone: ip222,ip232: USB headset mute after activation of changes in codec parameters, headset icon cleared on status line
| Status | Closed |
| Id | 82823 |
After for example
config add AC-DSP0 HEADSET /InputGain 32
config activate
the headset icon was cleared on status line and the headset was mute although the headset Talk key was handled.
PBX Waiting: Evaluate Busy on ... Calls for calls to an operator
| Status | Closed |
| Id | 82858 |
If a operator has configured Busy on 1 call, there should be no call from the Waiting Queue if the operator is already busy.
PBX: Avoid signaling loop after call transfer
| Status | Closed |
| Id | 82865 |
A signaling loop could be created by calling from a phone registered at one PBX to a phone at another PBX, then putting the call on hold and do the same call again, accept on the other side the waiting call. If both parties do then a transfer there is the signaling loop.
Such loop ist now detected and the call is cleared.
PBX Broadcast: No diverting name sent with broadcasted call
| Status | Closed |
| Id | 82880 |
The diverting and original-called name info was missing from the diverting leg2 info generated by the broadcast object
Web-UI: Misplaced reset-required indication
| Status | Closed |
| Id | 82896 |
Misplaced reset-required indication on ISDN interface config (TEL1,TEL2,...).
IP0010 IP1060 IP3010 IP6010 IP810 IP302 IP305: ISDN: Enable fax detection only after connect - fixed
| Status | Closed |
| Id | 82900 |
Fax detection was not enabled in all cases.
Happens in test/10.00/box/dsp/ip6010 with
fix: #78316: SIP/H.323: Don't complete media negotiation if no media can be seen
IP1060: Memory size is not correctly shown
| Status | Closed |
| Id | 82906 |
The memory size is not correctly shown in the IP1060 in diagnostics counter page.
phone: a calling party name found by inverse directory lookup for an external call was sometimes not stored in call list
| Status | Closed |
| Id | 82919 |
When the external call setup came in with a name identification provided by the external source and there was another name found by inverse directory lookup the name from directory was displayed on the call screen but the name identificication was stored in the call list. Now the name found by inverse directory lookup will be stored.
PBX Routing: Node extern did not work for calls from a trunk marked as local object
| Status | Closed |
| Id | 82948 |
The call was not routed back to the originating slave
IP222 IP232 IP241: New equalizer and volume setting
| Status | Closed |
| Id | 82952 |
- use equalizer up to approx 6db
- use digital volume to adjust volume
SIP: Trap on out of memory
| Status | Closed |
| Id | 83004 |
SIP-Client allocations not deleted.
Outbound control calls without facility interworking.
PBX Routing: A CFNR at a PBX object for WAN re-routing did not work if it contained escapes
| Status | Closed |
| Id | 83017 |
The number configured at the PBX object is interpreted in the context of the node of this PBX object. If escapes were needed to dial the WAN trunk, it did not work.
External-UC: Presence info assigned to wrong PBX object
| Status | Closed |
| Id | 83020 |
When presence eventlist is received from External-UC
all presence info was assigned to same PBX object.
PBX: Editing Config Templates impossible, if by some old firmware a strange config ended up in a User object
| Status | Closed |
| Id | 83033 |
The problem was a empty hardware id (hw=""). This is now ignored.
PBX mobility: The forking destination put in for mobility at a user should be dialed from the node of the user
| Status | Closed |
| Id | 83040 |
This number was dialed from the node of the Mobility object. This was confusing, because this number was configured at the user and it was also different behaviour as with forking without mobility
PBX: Standyby PBX generated alarms for missing slave registrations, even if active PBX up
| Status | Closed |
| Id | 83089 |
There should be only an alarm, if the standby PBX is active
SIP: One-way audio after mutual hold on dect systems
| Status | Closed |
| Id | 83126 |
Dect ep gets FTY_HOLD_NOTIFY but no FTY_RETRIEVE_NOTIFY.
X.509: Avoid alarms on missing system time after reboot
| Status | Closed |
| Id | 83137 |
Allow 60 seconds for setting the system time before an alarm is set. Also certificates are now rejected silently, in that time.
SIP: Problems with presence signaling on External-UC link
| Status | Closed |
| Id | 83177 |
Missing parameters on Contact-URI in 200/OK for SUBSCRIBE(presence):
maddr
transport
Wrong Contact-URI in presence XML in PUBLISH.
IP-DECT: Support for more than 32 LDAP attributes
| Status | Closed |
| Id | 83191 |
The IP-DECT devices support up to 256 LDAP attributes now.
PBX: Support for more than 64 LDAP attributes
| Status | Closed |
| Id | 83192 |
The PBX supports up to 256 LDAP attributes now.
IP-DECT: LDAP replication alarm loop
| Status | Closed |
| Id | 83249 |
If the LDAP replication is configured, but it can not be synchronized, there is an alarm loop. This is fixed now.
Gateway: On IP24, IP302, IP305 it could happen that multiple routes disappeared if a route was deleted
| Status | Closed |
| Id | 83278 |
Hard to predict when this happened.
Gateway: No Media received event was generated for T.38 calls, which started without actual RTP Traffic
| Status | Closed |
| Id | 83433 |
This happend for calls to/from Fax servers, which switch to T.38 without sending any RTP packets first.
SIP: Multiple subscriptions for 'message-summary'
| Status | Closed |
| Id | 83469 |
After every RAS_START another subscription for 'message-summary' is established.
phone: support PBX-directory access via TLS
| Status | Closed |
| Id | 78275 |
A "Use TLS" checkmark has been added to the PBX directory config of the phone. If set, port 636 is used instead of port 389.
PBX-Mobility: Data Call Thru
| Status | Closed |
| Id | 81513 |
To speed up dialing with the mobility client, the called number can be posted to the PBX via HTTP before the call
IP-DECT: DECT security (2)
| Status | Closed |
| Id | 82191 |
Some changes in preparation for the new feature "DECT security". This feature will be available later (IP1202).
IP-DECT: No fall-back after unattended call transfer
| Status | Closed |
| Id | 82198 |
Now there is no fall-back after an unattended call transfer and the behavior is consistent with the other call transfer types (attended, semi-attended). To switch back to the hold call the R-key must be pressed.
IP-DECT: Idle display update
| Status | Closed |
| Id | 82199 |
Handling of idle display update message in the call release message is added. Used with an OEM PBX.
phone: ip222,ip232: the variable KEYS0/HID-MAP permits to map new USB headsets to builtin descriptors
| Status | Closed |
| Id | 82635 |
Sometimes USB headsets come with a signature different from the signature of similar headsets which are already supported.
A "vars create KEYS0/HID-MAP p <map>" maps the new signature to an existing one.
<map> format is
manufacturer:product=manufacturer:product
the second manufacturer:product tuple is the signature of an already supported headset,
'manufacturer' and 'product' are plain 4 digit hex numbers without a "0x" prefix.
IP2x2, IP241: Coder Preferences for prefered coder G.722 suboptimal
| Status | Closed |
| Id | 82815 |
In case G.7222 was selected as prefered coder and the called endpoint did not support G.722, as next best coder G.729 was selected. This is typically not what is desired in such a case, G.711 is the better alternative in this case
myPBX: Show version of launcher in the list of sessions
| Status | Closed |
| Id | 82821 |
The version of the used launcher is now shown on the page PBX/myPBX.
Status:
checked-in: 10.00
checked-in: 9.00
H.323: Support for Avaya SRTP with AES128/80
| Status | Closed |
| Id | 82829 |
Needed for OEM
SoftwarePhone: HID Support for new headsets
| Status | Closed |
| Id | 83184 |
With this new version the following headsets are supported for call control:
- Jabra GO 6430 (Jabra LINK 350 USB with firmware 5.4.17 or later) with product id 0xa342. Please select the first device.
- Jabra SUPREME UC (Jabra LINK 360 USB) with product id 0xa346. Please select the first device.
- Jabra PRO 9470 with product id 0x1042.
- Sennheiser VoIP USB headset (SH 350 IP) with product id 0x0008.
- Sennheiser DW Office with product id 0x740a. Please select the first device.
- Sennheiser CEHS-CI 02 (USB adapter cable) with product id 0x0030. Please select the second device.
PBX: Description was missing for DECT System object
| Status | Closed |
| Id | 83198 |
A configurable description is useful for the DECT System object as for all other objects
Permit logging to a second (shadow) log server
| Status | Closed |
| Id | 83206 |
"Services/Logging/Log Server/Log Server Shadow/Address" defines the adress of a second server.
"Services/Logging/Log Server/Log Server Shadow/Enable" starts/stops logging to the second server.
Except the address the configuration for the second server is copied from the first server.
V9 Hotfix 15 (9061078)
Changes included in Version 9 hotfix15 Definition
IP-DECT: Trap during subscribing handsets
| Status | Closed |
| Id | 83690 |
A trap occurs during subscribing handsets on the IP1200. This is a fix for the previous fix #80424 and fixed again.
Trap identification:
XCPT: no 2 (TLB load) pc 942e23d8 ra 942e23cc va 00000000
PBX Waiting: Trap on leak-check if dtmf maps are configured
| Status | Closed |
| Id | 83691 |
Only happened if debug.xml leak check was used
IP222 IP232 IP241: Adjust gains for better echo canceller performance
| Status | Closed |
| Id | 83703 |
..
myPBX: Remove unimportant notifications
| Status | Closed |
| Id | 83707 |
The following events are not so important that the user has to be notified using a windows bubble notification.
- Visibility requests
- Missed calls
IP222 IP232 IP241: Enable noise reduction
| Status | Closed |
| Id | 83715 |
Enable noise reduction on all handset/handsfree/DHSG headset micro.
Use DSP code 660.27.pa.03
Noise reduction parameter can be tuned at
http://addr/AC-DSP0/mod_cmd.xml?xsl=phone-dsp.xsl
Voicemail: Prevent <prompt>, <record> on incoming control calls
| Status | Closed |
| Id | 83826 |
Turned out to cause idle-reset requests never being processed.
An administrative solution also exists: Watch out for the URL variable "$_noctl" in this article:
http://wiki.innovaphone.com/index.php?title=Howto:Configure_the_innovaphone_Voicemail#URL_Query_String_Variables
HTTPCLIENT: Allow configuration of a http authenticated URL even if the server needs no authentication
| Status | Closed |
| Id | 83900 |
If a password was configured and the server didnt need it, a HTTP put created a file of zero length.
PBX Waiting: Trap on collision of operator connect and two-stage dialing
| Status | Closed |
| Id | 83926 |
If two-stage dialing (Maps) is used to call a Trunk or Gateway object, the call is sent after a blockdial timeout. If an operator connected the call before this timeout happened, a trap occured.
For this to happen DTMF maps and operators have to be configured on the same Waiting Queue object, with is kind of unusual
PBX: Call to a Trunk/Gateway was not marked correctly as external, if no connected number was received
| Status | Closed |
| Id | 83940 |
Calls from a Trunk must be explicitly marked as internal with respective connected number, otherwise they should be treated as external
HTTP: Chunked transfer fails if the last 2 bytes of the chunk header are in the next tcp packet
| Status | Closed |
| Id | 83986 |
Seen with the application platform as broken pipe
PBX: Registration with Name/Number did not work correctly if default device not first
| Status | Closed |
| Id | 83989 |
For a registration with name or number, the information if the PBX password shall be used was always taken from the first device regardless if this was the default device (hw-id identical to name) or not
PBX: Partial Rerouting was prohibited in Alerting State (CFNR)
| Status | Closed |
| Id | 83993 |
This was done under the assumption the partial rerouting is not supported as CFNR by public networks.
PBX Waiting: Call to operator with Twin Phone Checkmark did not work anymore
| Status | Closed |
| Id | 84084 |
Collateral damage of
fix: #82858: PBX Waiting: Evaluate Busy on ... Calls for calls to an operator
SoftwarePhone: Product string
| Status | Closed |
| Id | 84119 |
The product string is changed, used e.g with the PBX registration.
Webfax: Vertical resolution
| Status | Closed |
| Id | 84171 |
The vertical resolutions 96 lpi, 196 lpi and 400 lpi are correctly saved in the SFF file with a proprietary definition known by the tool sfftobmp.
phone: provide complete dialog info to a phoneapp
| Status | Closed |
| Id | 84192 |
Both group indications and dialog infos are signaled via a group indication facility. For dialog infos the parked_to_alerting member was overloaded to provide the info as expected by the existing phoneapp.
Now the parked_to_alerting member is passed to a phoneapp as received.
IP222 IP232 IP241: Force same speed of the switch ports for 1000M/100M scenarios (configuration option added)
| Status | Closed |
| Id | 84200 |
In scenarios with frequent transistions of the attached PC to sleep renegotiating the link speed may be undesired. For this case the force same speed mechanism can be disabled.
Other changes:
1000M is only changed to 100M if the other port runs at 100M. The previous version changed from 1000M to 100M if the other port runs at 100M or 10M.
The statistics can be collected from the PC port or from the LAN prot or from both.
Packet forwarding on the PC port is disabled if the port is down to avoid misleading collision counter behaviour.
CDR fixes for external call detection
| Status | Closed |
| Id | 84211 |
The type="ext" attribute was not set reliably. Additionaly an attribute pseudo was added to the <user/> tag to indicate the type of object the CDRis created for.
http client : authentication was not retried after a failure when the offending request was repeated in the same session
| Status | Closed |
| Id | 84217 |
When a httpclient user repeated a failing request in the same session the authentication was not tried again. Thus a change of the client side URL password or a change of the server side password had no effect until a new session was started.
IP6000: Prevent blinking error LED on old IP6000 with HW-Build <110
| Status | Closed |
| Id | 84227 |
Conference DSP driver was started on old hardware that doesnt support the conference DSP
IP222 IP232 IP241: pressing speaker key when phone is in handset mode switches to handsfree mode instead to toggle monitor mode
| Status | Closed |
| Id | 84297 |
handset/headset plus speaker is not supported
myPBX: Inconsistent display of group monitoring rights
| Status | Closed |
| Id | 84350 |
In the visibility settings groups were not displayed if the membership is "dynamic out". But members of that group still have monitoring rights.
PBX: Import did not work with some data
| Status | Closed |
| Id | 84356 |
The data is processed in chunks. If the chunk border was right behind the closing </user> tag, the decoding of the next chunk failed.
phone: when scrolling directory search results sometimes one of the numbers of a contact was not displayed
| Status | Closed |
| Id | 84362 |
the tag characters assigned to the different numbers were not included in sort order.
phone: status messages for outbound external calls were sometimes garbled on display
| Status | Closed |
| Id | 84365 |
happened when for a preceeding outbound internal call in alerting state presence info was displayed
IP0010: DSP didnt start with build 9061044
| Status | Closed |
| Id | 84388 |
Page Ldap/Replicator/Status didn't display in WebKit
| Status | Closed |
| Id | 84400 |
Page Ldap/Replicator/Status didn't display in WebKit
phone_orchid: displaying both dialed and connected number may be misleading, it's better to omit dialed number
| Status | Closed |
| Id | 84422 |
when for example 022222222 was dialed and the network reported a connected number 03022222222 the display info "022222222 -> 03022222222" looked like a transfer.
Hide LDAP Server Password For Viewer Accounts
| Status | Closed |
| Id | 84557 |
Was accessible for viewers
phone: enable directory search function key also in connected state
| Status | Closed |
| Id | 84559 |
it's sometimes useful to browse the directories for a number while talking.
myPBX: Only send one command at a time
| Status | Closed |
| Id | 84568 |
Queue commands instead of sending overlapping commands. This limits the number of open AJAX connections to two.
H.323: Accidential fallback to slowstart if faststart response received in PROGRESS
| Status | Closed |
| Id | 84601 |
Only happened in some H.323/SIP interop scenarios
IP222 IP232 IP241: Updated gain and equalizer setting
| Status | Closed |
| Id | 84605 |
Tuned for high MOS values according to ETSI ES202737 ES202738 ES202739 ES202740
IP0010 IP1060 IP3010 IP6010 IP22 IP24 IP28 IP302 IP305 IP800 IP6000: Minifirmware not shown on LED
| Status | Closed |
| Id | 84616 |
Minifirmware should blink long green short red. This happened only after a firmware/bootcode update.
phone: ip222, ip232: status stage added to all USB control transfers
| Status | Closed |
| Id | 84617 |
the status stage is mandatory as well for IN as for OUT control tranfers
H.323: More information on "Unexpected Message" event
| Status | Closed |
| Id | 84699 |
The message type and the state for which this message was unexpected is needed to find out what the problem is.
SIP: Trap during channel handling
| Status | Closed |
| Id | 84800 |
Rare trap when re-assigning channels.
IP232,IP222,IP241: Rendering errors when trying to use backround image with indexed colors
| Status | Closed |
| Id | 84814 |
PNG mode "indexed colors" is not supported.
Trying to use an indexed color png leaves phone screen in bad shape.
NAT: Don't forward DNS requests from public network
| Status | Closed |
| Id | 84842 |
As kind of denial of service attack, bursts of incoming DNS requests were seen. The nat process was forwarding these requests to the public DNS. This is a useful function for DNS requests from the private network, but not for requests from the public network.
These DNS requests are now discarded
H.323: Media Negotiation did not work for call with reverse media and media response in CALL-PROC
| Status | Closed |
| Id | 84848 |
Caused SIP interop problems with CUCM
Maximum LDAP PDU Size Too Small
| Status | Closed |
| Id | 84851 |
Maximum LDAP PDU Size Too Small. Now internal maximum allocation unit plus a bit for encoding overhead.
PBX Broadcast: Call to group members was not cleared when CFNR was executed
| Status | Closed |
| Id | 84857 |
This was a collateral damage from
fix: #79549: PBX Waiting/Broadcast: Incomplete CDRs if CFNR configured on object
IPVA: V10 Code Merge Aiming To Stabilize FW Upload To DRAM
| Status | Closed |
| Id | 84862 |
Addresses a problem in conjunction with innovaphone's automated test environment
PBX: Filters should only be applied to calls with media channels
| Status | Closed |
| Id | 84892 |
A join group operation could be prohibited with filters. This was unexpected.
phone: ip222, ip232: raise alarm if an USB Headset does not respond on USB bus anymore
| Status | Closed |
| Id | 84893 |
SIP: Fix for media negotiation on calls re-routed from TONE interface to outbound SIP
| Status | Closed |
| Id | 84932 |
Helps on gateway interfaces configured for media-relay with exclusive codec.
IP-DECT: Call transfer timer
| Status | Closed |
| Id | 84951 |
The call transfer timer is stopped with the call proceeding event now. This fixes a call transfer to e.g. a mobile user with a delayed alert.
IP-DECT: Reverse phone book
| Status | Closed |
| Id | 84953 |
The reverse phone book does not work till hotfix 11. This is fixed again now.
| Status | Closed |
| Id | 84991 |
Loosing remote IP addresses when DNS becomes temorarily unavailable.
Results in interfaces without remote ip addresses.
PBX Routing: A CFNR on a slave gateway object redirecting the call to the master failed
| Status | Closed |
| Id | 85017 |
This was a collateral damage of
fix: #77874: PBX: Routing problem with nodes/escapes/slaves with calls to object in same node but different PBX
SIP: Switch to T.38 did not work when interworking with H.323 slowstart (XCAPI)
| Status | Closed |
| Id | 85047 |
Switch to T.38 did not work when interworking with H.323 slowstart (XCAPI).
IP222 IP232 IP241: Headet volume adjust added, re-enable noise reduction in headset mode
| Status | Closed |
| Id | 85058 |
Headset volume adjust can be done with the webinterface at
http://addr/dsp.xml
This feature can be used to adapt to different DHSG headsets.
The noise reduction in headset mode is re-enabled to avoid sporadic noise with
SIP: Memory leak
| Status | Closed |
| Id | 85083 |
Memory leak.
SIP: Fix for STUN problems
| Status | Closed |
| Id | 85118 |
Try alternative STUN server address if first fails and another was gathered by DNS.
IP232,IP222,IP241: Do not hide configured fkey if neither text nor icon is displayed
| Status | Closed |
| Id | 85211 |
Do not hide configured fkey if neither text nor icon is displayed.
E.g. Call Forwarding fkey with CF destination but no label text.
PBX: Slave with non-ASCII PBX name did not register at master
| Status | Closed |
| Id | 85235 |
Error in utf-8 to unicode convertion in this case
New remote control codes to be used for phone tests by soap applications
| Status | Closed |
| Id | 83468 |
The new UserRc codes are executed only when the addressed phone is either in handset, headset or handsfree mode, i.e when calling, connected or disconnected but not when alerting:
16 - change to handset mode
17 - change to headset mode
18 - change to handsfree mode
19 - monitor mode on (add speaker to handset or headset mode)
20 - monitor mode off (back to plain handset or headset mode)
Alarm/Event handling: Authentication for received remote Alarms/Events
| Status | Closed |
| Id | 83603 |
Allow by configuration to only accept authenticated alarms or events
PBX: Preparations for objects visible only if appropriate license installed
| Status | Closed |
| Id | 83615 |
for v10
PBX: New Feature to allow registration with password to devices regardless of address filter
| Status | Closed |
| Id | 83794 |
It is now possible to configure a flag at a device to allow a registration for this device even if there is an IP Filter which does not match. This is useful if registrations from the public internet to the PBX shall be possible. Without this feature this could be opened only for the complete PBX. Now it can be restricted to a few devices.
SoftwarePhone: Signature added
| Status | Closed |
| Id | 83915 |
The SoftwarePhone installer and the install package is signed now.
IP-DECT: Phone book 'Use TLS' option
| Status | Closed |
| Id | 84001 |
'Use TLS' option added for the central phone book search. This changes the standard port from 389 to 636 if no port is configured. The central phone book search is only available with the IP1202.
simple static logging interface
| Status | Closed |
| Id | 84204 |
to simplify sending of log messages, alarms and errors a simple static interface to the logging module was added. log_if::log(class serial src, const class event & event) passes the given event to the primary logging module (aka LOG0). This works also with 'src' = 0.
myPBX: Pass selected device to launcher
| Status | Closed |
| Id | 84556 |
Needed for version 10 remote video
SIP: Announcing "a=T38MaxBitRate:14400" in T.38 offer
| Status | Closed |
| Id | 84770 |
Announcing "a=T38MaxBitRate:14400" in SDP offer for T.38
since all current devices support that modem speed.
SIP: Set numbering plan to "ISDN/telephony" in case of "Bellcore-dr2" as Alert-Info in INVITE
| Status | Closed |
| Id | 84939 |
Set numbering plan to "ISDN/telephony" in case of "Bellcore-dr2" as Alert-Info in INVITE.
IP-DECT: Phone book error events
| Status | Closed |
| Id | 85001 |
The phone book module sends an error event now if the LDAP directory search fails.
V9 Hotfix 16 (9.061101)
Changes included in Version 9 hotfix16 Definition
phone: ip222, ip232: USB Headset could not be disabled via Menu or by Headset Function in Enable mode
| Status | Closed |
| Id | 84444 |
sometimes the user want's to use the phone as if no headset is connected, i.e. when for example redial key is pressed after a number has been entered or a list entry has been selected the call should be started in handsfree mode and not in headset mode.
If now the headset is disabled via menu or the headset(enable) function key all headset functions are completely disabled and no calls are directed to the headset, the status bar displays an icon indicating the disabled state.
H.323: Interop Problem with CUCM SIP for Transfer
| Status | Closed |
| Id | 85120 |
For the transfer the CUCM first sets the call on hold and then requests a new media proposal from this call, which we cannot deliver. The request is just ignored, there should be an answer.
IP152: Call replacement (blind transfer) did not work
| Status | Closed |
| Id | 85313 |
Call replacement (blind transfer) ends up in no audio.
Trap when disabling Gateway interface
| Status | Closed |
| Id | 85377 |
0:0204:814:3 - SIP_UDP.17 -> SIP_TRANSPORT.1 : SOCKET_RECVFROM_RESULT(87.173.157.2:5060,10.2.2.1:5060;95fd:75c0:9485:bc94:9485:bc30:9406:630c:5060,948a:2198:9476:5924:19:72:0:1:5060)
data(913),SIP_TRANSPORT
\tREGISTER sip:gwdl_ip800 SIP/2.0
\tVia: SIP/2.0/UDP 87.173.157.2:5060;branch=z9hG4bKA072411261DD0135
\tRoute: <sip:87.139.89.223;lr>
\tFrom: <sip:93@gwdl_ip800>;tag=4110839322
\tTo: <sip:93@gwdl_ip800>
\tCall-ID: DECE171E57A1898F@192.168.178.1
\tCSeq: 7998 REGISTER
\tContact: <sip:93@87.173.157.2;uniq=D526C620EFAB7D1ACBBC469D33242>
\tAuthorization: Digest username="_!(schuetz)", realm="gwdl_ip800", nonce="7977d820e909d311", uri="sip:gwdl_ip800", response="6999f52d0e7d4ddbd18b23b59936aaa6", algorithm=MD5, cnonce="F56053DD72D3E1B3", qop=auth, nc=00000401
\tExpires: 1800
\tMax-Forwards: 70
\tUser-Agent: AVM FRITZ!Box Fon WLAN 7112 (UI) 87.04.87 (Jun 7 2011)
\tSupported: 100rel,replaces,timer
\tAllow-Events: telephone-event,refer,reg
\tAllow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
\tAccept: application/sdp, multipart/mixed
\tAccept-Encoding: identity
\tContent-Length: 0
0:0204:815:5 - SIP-Client(SIP-CLIENT.0) <sip:93@gwdl_ip800;user=phone> ...
0:0204:816:5 - SIP-Client(SIP-CLIENT.0) Idle->Registering
0:0204:816:6 - SIP_RAS_APP.3 -> GK.0 : RAS_DISCOVERY, ip=87.173.157.2
data(6),GK
00 00 00 02 39 33 ....93
0:0204:816:7 - GK.0 -> SIP_RAS_APP.3 : RAS_DISCOVERY_CONFIRM
0:0204:817:0 - SIP_RAS_APP.3 -> GK.0 : RAS_REGISTRATION(87.173.157.2:5060)
data(6),GK
00 00 00 02 39 33 ....93
0:0204:817:1 - GK.0 -> SIP_RAS_APP.3 : RAS_REGISTRATION_CONFIRM
data(32),SIP_RAS_APP
00 00 00 02 39 33 00 01 00 16 00 5f 00 21 00 28 ....93....._.!.(
00 73 00 63 00 68 00 75 00 65 00 74 00 7a 00 29 .s.c.h.u.e.t.z.)
0:0204:817:1 - DEBUG this=948b325c event.sig=0
0:0204:817:2 - DEBUG this->reg_reference=0 event.reference=948b3ebc
Presence note got lost
| Status | Closed |
| Id | 85469 |
When configuring a presence on the phone (Main menu/User setup/Presence)
the note is not save along with the selected activity
PBX: Potential trap on registration of a slave PBX, with Master GK-ID configuration changes
| Status | Closed |
| Id | 85477 |
The trap happened if on the Slave a Master GK-ID was configured, then the slave registered, and afterwards the Master GK-ID was removed again and the slave registered again and then was restarted once more.
Presence: Note moves into 'tel' presence
| Status | Closed |
| Id | 85482 |
Presence note moves into 'tel' presence when changing activity by use of 'presence' fkey.
IP232,IP222: Automatically enter input mode 'alpha' when entering directory search screen through fkey
| Status | Closed |
| Id | 85493 |
Automatically enter input mode 'alpha' when entering directory search screen through fkey.
SIP: Interop with Genband SBC
| Status | Closed |
| Id | 85534 |
Handling this kind of SDP offer:
\tv=0
\to=IOTMSX1-0 17 2 IN IP4 206.165.51.38
\ts=sip call
\tc=IN IP4 0.0.0.0
\tt=0 0
\tm=audio 42076 RTP/AVP 0 8 4 18 101 13
\ta=sendonly
\ta=rtpmap:101 telephone-event/8000
\ta=fmtp:18 annexa=yes
\ta=fmtp:18 annexb=no
\ta=fmtp:101 0-15
\ta=ptime:20
\ta=silenceSupp:off - - - -
PBX-Exec: Call at secretary for executive had wrong destination name/number
| Status | Closed |
| Id | 85535 |
Call should be displayed as call to the scretary, diverted by the executive
Media: RTP did not work on switch from slowstart T.38 to EFC G.711
| Status | Closed |
| Id | 85548 |
T.38 retransmission packets were sent, which caused problems with the RTP
IP222 IP232 IP241: DSP Update to Version 680.05
| Status | Closed |
| Id | 85552 |
Sporadic problems with noise reduction fixed.
Noise reduction can be enabled for Handset/Headset/Handsfree mode
SIP: Providing private RTP address although public RTP address is available
| Status | Closed |
| Id | 85751 |
During re-negotiation:
Providing private RTP address to external endpoint although public RTP address is available.
SIP: Do not send "504 Server Time-out" after "200 OK"
| Status | Closed |
| Id | 85788 |
Do not send "504 Server Time-out" after "200 OK".
IP-DECT: Call transfer compatibility with SIP
| Status | Closed |
| Id | 85795 |
For compatibility reasons with the SIP protocol no hold signal should be sent to the call transfer destination. This is fixed again.
SIP: Handling of "sendonly" offer after "inactive" offer
| Status | Closed |
| Id | 85857 |
Switching from "inactive" into "recvonly".
Giving REMOTE_RETRIEVE and HOLD_NOTIFY to app.
SIP: Ignore From-URI in re-INVITE and UPDATE when "from-change" not supported
| Status | Closed |
| Id | 85873 |
Ignore From-URI in re-INVITE and UPDATE when "from-change" not supported by remote side.
Phones: Touching should start dialing of selected directory entry
| Status | Closed |
| Id | 85959 |
Touching should start dialing of selected directory entry
instead of opening directory entry for editing.
At least in Context of Wahlvorbereitung.
SNMP-Traps: agent-addr wasn't reflecting altered IP adress
| Status | Closed |
| Id | 85970 |
SNMP-Traps: The SNMP trap's agent-addr member carried an outdated ip adress if the DHCP leased ip address changed.
SIP: Optimization when validating local media address
| Status | Closed |
| Id | 86029 |
Optimization when validating local media address.
Keep number of SOCKET_GET_LOCAL_ADDR low.
DHCP: client IP connectivity lost when a renew/rebind request for the currently assigned address was refused by the server
| Status | Closed |
| Id | 86030 |
When a client renew/rebind request is refused by the server providing the current address the client starts a new discovery. But in case of success the new address was not set and the client could not be reached anymore.
ENUM: Port in SIP-URI was not honored
| Status | Closed |
| Id | 86037 |
Port in SIP-URI was not honored in regex of DNS result.
IP-DECT: Transferred/rerouted call display
| Status | Closed |
| Id | 86041 |
The remote party number of transferred and rerouted calls are not correctly shown in the handset's display. This fixes the display of CTI initiated calls.
IP-DECT: MAC-alias change of OEM device
| Status | Closed |
| Id | 86047 |
The MAC-alias of an OEM device was changed and this results in conflicts within several DECT modules. Different product short names of the same device are correctly accepted now.
ip1202: Send Inband DTMF did not work
| Status | Closed |
| Id | 86052 |
the "DECT/Config/Master/Send Inband DTMF" checkmark had no effect.
SIP: Support for multiple audio media descriptions
| Status | Closed |
| Id | 86083 |
One without encryption and one with encryption:
v=0
o=OpenStage-Line_0 968610650 446118927 IN IP4 10.30.1.144
s=SIP Call
c=IN IP4 10.30.1.144
t=0 0
m=audio 5012 RTP/AVP 9 8 0 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv
m=audio 5010 RTP/SAVP 9 8 0 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ujVU8G6kgknZnPflRwx8tadNskkp9glas/DFCbC3
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:+uoFBNVVhDY5OOGvMOdAlvpxvc98hX/VeProhlwH
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=sendrecv
Change order of links on debug page
| Status | Closed |
| Id | 86121 |
Make "Tracing" the first item so that no leak check is triggered when opening the page.
Phones: Using config option "CGPN" as CGPN on outbound SIP calls
| Status | Closed |
| Id | 86122 |
Using config option "CGPN" as CGPN on outbound SIP calls.
phone: ip222, ip232: recover from USB port failure probably caused by electric sparks
| Status | Closed |
| Id | 86125 |
An USB port failure is indicated when a Plantronics DA45 headset adapter is plugged and a certain kind of table lamp (halogen) is switched on or off.
It happens independent of current state of the headset (idle or in call)
but only with the abovementioned adapter.
The exact reason is not known yet, may be it's an electric spark from the switch of the lamp or some pulse.
The fix is to reset the port and to restart the plugin process, a possibly active call is terminated.
phone: coder settings of a "Create Registration" function key were not applied to the created registration
| Status | Closed |
| Id | 86164 |
Disable leak check if debug flag is not set
| Status | Closed |
| Id | 86165 |
The leak check is only allowed if the config flag CPU /debug is set.
AD Replication: Configuration Buffer Increased
| Status | Closed |
| Id | 86211 |
Was too small for many maps
whistling tone in all ip28 a/b ports on incoming call
| Status | Closed |
| Id | 86212 |
This phenomena occured after few day uptime after sending CLIP
ip72 firmware did not boot anymore since V9hotfix11
| Status | Closed |
| Id | 86246 |
SIP: Avoid re-configuration of DSP channel when processing re-INVITE
| Status | Closed |
| Id | 86316 |
Fix is required for interop with SIP devices sending re-INVITE for session-refresh,
but incrementing version field in SDP body, altough there is no change in SDP.
phone: headset function key mode 'control' could be configured via WEB interface only
| Status | Closed |
| Id | 86327 |
must be possible locally at the phone too
phone: Partner state provided via dialog info was not reset when the subscription call was released because of network errors
| Status | Closed |
| Id | 86330 |
Gateway: Routing on blind transfer call starts from wrong IF
| Status | Closed |
| Id | 86386 |
Transfered endpoint was used as source interface on routing.
Better use transfering endpoint as source on routing of (blind) transfer call.
Also transfer-to endpoint missed ctSetup.
Also transfered endpoint missed ctComplete.
Linux: Start-up failures
| Status | Closed |
| Id | 86399 |
Linux start-up is improved. This fixes hanging Linux start-ups caused by compact-flash failures.
IP-DECT: GUI Master Configuration
| Status | Closed |
| Id | 86442 |
Disabling LDAP Directory Search fixed for the IP1202.
Linux: GUI in viewer mode
| Status | Closed |
| Id | 86455 |
The Linux General page is fixed for the viewer only mode.
SIP: PRACK after CANCEL contains bad RAck header value
| Status | Closed |
| Id | 86469 |
CSeq or original INVITE transaction was damaged.
But only if CANCEL has been sent right before PRACK.
CANCEL is sent before PRACK only if SDP answer of provisional response is invalid.
PBX: Conference trap
| Status | Closed |
| Id | 86473 |
Potential trap in the PBX BC conference object is fixed.
SNMP: Obsoleted Enterprise-Specific Trap "innoIsdnFailure"
| Status | Closed |
| Id | 86513 |
This SNMP trap is no longer necessary. Meanwhile it is covered more consistently by "innoDiagAlarm" and "innoDiagAlarmClear".
Linux: GUI removed from IP1060
| Status | Closed |
| Id | 86547 |
The IP1060 does not support the Linux Application Platform. The GUI is removed now. Also a trap is fixed, if somebody tries to enable Linux.
Voicemail: <pbx-disc> failed sometimes
| Status | Closed |
| Id | 86569 |
In case of unconnected calls that were subject of a prior <pbx-fwd>
phone: ip222, ip232: handset/handsfree speaker was not switched off when changing over to headset mode
| Status | Closed |
| Id | 86706 |
When the headset talk-key or the phone headset-control-key is pressed while the phone is in handset or handsfree mode the phone changes to headset mode, i.e. headset micro and speaker are activated. The handset or handsfree speaker should be switched off then.
phone_orchid: pressing the speaker key while in headset mode did not switch over to handsfree mode
| Status | Closed |
| Id | 86707 |
When the speaker key is pressed while the phone is in headset mode the headset should be deactivated and the phone should enter handsfree mode.
Media: Preparing for G.722.1
| Status | Closed |
| Id | 85316 |
Decoding SDP containing G.722.1
\tv=0
\to=- 5140 5141 IN IP4 10.138.6.91
\ts=-
\tc=IN IP4 10.138.6.91
\tt=0 0
\tm=audio 50000 RTP/AVP 122 8 0 18 121 101
\ta=rtpmap:122 G7221/16000
\ta=fmtp:122 bitrate=32000
\ta=rtpmap:121 L16/16000
\ta=rtpmap:18 G729/8000
\ta=fmtp:18 annexb=no
\ta=rtpmap:101 telephone-event/8000
\ta=fmtp:101 0-15
SIP: New config file option /tls-unchecked
| Status | Closed |
| Id | 85449 |
New config file option /tls-unchecked disables validation of remote server certificate.
IP-DECT: Configuration changes for OEM device
| Status | Closed |
| Id | 85522 |
For an OEM device the configuration update behaviour is changed.
SIP: New config file option /fixed-contact-addr
| Status | Closed |
| Id | 85529 |
New config file option /fixed-contact-addr to keep SIP client from changing it's Contact address into public address of NAT mapping after registration.
(RFC-3581 Symmetric Response Routing)
IP-DECT: DECT security (3)
| Status | Closed |
| Id | 85554 |
Some changes in preparation for the new feature "DECT security". This feature will be available later (IP1202).
Phones: New config file option /recording-without-remote-party-info
| Status | Closed |
| Id | 85833 |
For interop of recording feature with VOXTRON application.
VOXTRON application gets confused by Diversion header in INVITE.
config change PHONE SIG /recording-without-remote-party-info
phone_orchid: configuration prameters to adjust microphone and speaker volume
| Status | Closed |
| Id | 85848 |
These parameters set a volume correction factor which is applied at any volume level. Parameter changes are applied immediately even in an active call.
config add AC-DSP0 HEADSET /mic-volume <mic-adjust> /spk-volume <spk-adjust>
config write
config activate
-20 <= <*-adjust> <= +20
IP-DECT: Radio reconnect handling with OEM PBX
| Status | Closed |
| Id | 85952 |
The handling for the endpoint location update in combination of a radio reconnect is changed. This fixes the base station behaviour with an OEM PBX.
ip1202: support capture of raw ethernet packets exchanged between MSP (aka DSP) and the firmware
| Status | Closed |
| Id | 86049 |
All exchange between the firmware running on the ACP (Application Command Processor) and the firmware running on the MSP (Media Strem Processor) is in ethernet packet format. The Mindspeed support prefers this trace format.
The capture is enabled via
config add MSP0 /mtrace
It includes as well command and RTP data packets and thus duplicates the RTP packets traced by the general "All TCP/UDP Traffic" option.
HTTP-Client: Requests with specified credentials
| Status | Closed |
| Id | 86133 |
Additional function for OEM httpclient::auth_request
It uses the specified credentials and ignores the username and password from the configuration.
Phones: New config option "Allow User Settings at Phone"
| Status | Closed |
| Id | 86243 |
Allow changing of User Settings even if 'Protect Configuration at Phone' is activated.
| Status | Closed |
| Id | 86527 |
The new hidden option 'Max RTP streams' is added to the IP-DECT Radio module. The option is only visible for an OEM device, but can be used with config change command ("/max-rtp-streams <count>"). The feature is useful to limit the RTP streams for radios connected to the IP-DECT Master with a low data bandwidth. Conference calls are not limited with this feature.
Gateway: New config option "No blind transfer"
| Status | Closed |
| Id | 86689 |
New config option "No blind transfer" to keep Gateway from handling blind transfer requests.
If set blind transfer requests are passed through.
Handling is performed at the next signaling hop.
V9 Hotfix 17 (9061152)
Changes included in Version 9 hotfix17 Definition
make update script parsing more tolerant to suspicious line ends
| Status | Closed |
| Id | 84349 |
Some WEBDAV tools garble line end when a text file is stored after editing. Last seen \\r\\r
instead of \\r
. Any sequence consisting only of \\r
chars should be read as one line end because empty lines have no meaning in an update script.
SIP: Do not interwork holdNotific and retrieveNotific while on hold
| Status | Closed |
| Id | 86736 |
Interworking problem with VOXTRON/XCAPI
Do not interwork holdNotific and retrieveNotific after remoteHold.
Disable LDAP Server When Erasing Flash Directory
| Status | Closed |
| Id | 86763 |
During an upload of a complete configuration the command "mod cmd FLASHDIR0 erase-all" will erase all flash directory content. Replication clients are going to receive nil-responses making them assume a certain entry does no longer exist.
Replication clients are now barred from accessing the LDAP server as long as the box didn't process the post-upload reset.
IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: fax bad signal quality events are sent on good fax connections
| Status | Closed |
| Id | 86788 |
Sometimes during fax transfer a bad signal quality (e.g.50) is reported, even if the connection is good. This happens during the TCF phase, in the image phase the signal quality is fine ( e.g. 3)
Mis-typing on Maintenance/Diagnostics/Counters
| Status | Closed |
| Id | 86812 |
Show "kbit/s" instead of "kBbit/s".
Gateway: #11 could not be dialed on analog interfaces with feature codes enabled
| Status | Closed |
| Id | 86819 |
This is a featiure code used on DECT systems and it was not disabled on analog interfaces
Gateway: Potential trap when recording is configured and a transfer happens
| Status | Closed |
| Id | 86837 |
The problem is with a transfer, which is executed in the Gateway, not the PBX
PBX: XML Export/Import did not work, if DTMF Feature objects present
| Status | Closed |
| Id | 86870 |
The automatically generated user objects caused a problem. This could result in a config that caused the PBX to restart in a loop. The export/import was fixed and the PBX does not restart because of the corrupt config any more.
PBX: Trap if a Hold was attempted for a call without media
| Status | Closed |
| Id | 86874 |
Could be caused by a misbehaving application or voip device
PBX: Name beginning with '*' caused problems with SOAP applications
| Status | Closed |
| Id | 86882 |
A search for such a name was treated as wildcard search for all users. The name '*' is now a reserved name, which cannot be used for an object.
Linux: IP0010 available again
| Status | Closed |
| Id | 86903 |
With V9 hotfix 16 Linux can not be started on the IP0010. This is fixed now.
phone: call diversion override via indirect dialing could not be disabled
| Status | Closed |
| Id | 86944 |
In the indirect dialing screen the right arrow key opens a menu with different options how to place the call.
"Dial - No Diversion" ssets up a call which will ignore the diversions active on the target phone. This menu item can be supressed via the "Fine grained function locking" bit PHONE_LOCK_DIVERSION_OVERRIDE 0x04000000
PBX Waiting: CFNR with number filter did not work
| Status | Closed |
| Id | 86959 |
The CFNR was not executed
PBX Conference: Calling Party was missing in the CDRs for calls to conference members
| Status | Closed |
| Id | 86977 |
Only the conference object itself was present as forwarding party
SIP: Rare trap when cancelling call
| Status | Closed |
| Id | 86994 |
When call abort interferes with re-routing in gateway application.
PBX: Replication from a dyn PBX was not possible
| Status | Closed |
| Id | 87065 |
Configuration of dyn PBX id now possible for replication
TLS: Ignore incoming HelloRequest messages
| Status | Closed |
| Id | 87091 |
When receiving a HelloRequest the TLS client dopped the connection and sent an "Unexpected message" alert.
Now the HelloRequest is ignored and an "No renegotiation" warning is sent.
PBX: Call, which was rejected with busy, because of CF loop, could not be forwarded by Trunk Busy destination
| Status | Closed |
| Id | 87150 |
Such a call was always disconnected
IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: DSP Update to Version 680.05
| Status | Closed |
| Id | 87179 |
The jitter buffer performance has been improved.
The fax/modem bypass performance has been improved.
IP28 firmware size was reduced ( unused code is not linked ).
IP28 G279 didnt work with more than 4 channels.
IP222 IP232 IP241: Disable PC port didnt work
| Status | Closed |
| Id | 87250 |
IP-DECT: Login feature with disabled local R-key handling
| Status | Closed |
| Id | 87269 |
The login feature is fixed now if the local R-key handling is disabled.
Webdav: DELETE may fail if file has been opened for reading before
| Status | Closed |
| Id | 87288 |
If reading (GET) stops but HTTP session remains open, the file remains in state 'open'
and subsequent DELETE request fails with "500 Internal Server Error".
Close file and re-try to delete.
HTTP: Chunked transfer fails if the chunk header is not in a single packet
| Status | Closed |
| Id | 87292 |
PBX: Conference and hanging VM script
| Status | Closed |
| Id | 87312 |
Wrong media initializing in the BC Conference object causes a hanging VM script if a file should be played. This is fixed now.
IP-DECT: No voice with early handover
| Status | Closed |
| Id | 87504 |
This fixes outgoing calls without voice if early handover is done before the media channel is initialized.
phone: ip222, ip232: multiple reports indicating offhook were misinterpreted and could drop a call
| Status | Closed |
| Id | 87506 |
Wireless USB headsets may send more than one report indicating headset offhook state in conjunction with different wireless link states. If the interval between the first and the second indication was very short (8 ms) the second indication was misinterpreted and the just setup call was dropped (observed with a Jabra PRO 930 after plugin).
SIP: Fix for Contact-URI in 200/OK for SUBSCRIBE
| Status | Closed |
| Id | 87507 |
Fix for Contact-URI in 200/OK for SUBSCRIBE and NOTIFY requests.
May contain double port attribue:
SUBSCRIBE sip:8011@172.20.11.53:2053 SIP/2.0
Contact: <sip:8011@172.20.11.53:2053:2053;maddr=172.20.11.53;transport=UDP>
Status:
Fixed in 10.00 and 9.00
IP-DECT: Disturbances GUI info change
| Status | Closed |
| Id | 87522 |
The text of the GUI page disturbances is changed.
IP-DECT: Web UI info page, version and release state
| Status | Closed |
| Id | 87538 |
Now the version and the release state are shown on the web UI info page of the IP1202.
PBX: Blind transfer to WQ by IP-DECT caused hanging calls
| Status | Closed |
| Id | 87542 |
The call-leg to the transfering phone was not cleared by the PBX, so if the phone did not clear this call, it was hanging for ever. Other phones clear such a call after a timeout, but this is only a workaround, the call must be cleared by the PBX
H.323: Problem sending real big signaling messages
| Status | Closed |
| Id | 87543 |
Happened with a configuration with more the 60 Join Group function keys on the phone. The message was not sent and the operation failed.
PBX-SOAP: UserCall on WQ did not take "Send Number" into account and call was not marked as internal
| Status | Closed |
| Id | 87559 |
Calls should be send with "Send Number" as source if configured
PBX: CFB configured at PBX object did not patch number correctly
| Status | Closed |
| Id | 87581 |
In case the called endpoint was in different node then PBX object, the number did not contain all the needed prefixes
SIP: Trap when logging out dect user
| Status | Closed |
| Id | 87606 |
Trap when logging out dect user.
0:0287:977:1 - MASTER_EP-SIG.0 default(948f3e18): serial_event(100) src=DECTMASTER-CALL.0 mod=SIP
0:0287:977:1 - Assertion failed line 790 in common/os/os.cpp, object deleted
0:0287:977:2 - assert-ep 94004474 called from 94055498
PBX: Called number was missing in CDRs for calls to busy endpoint
| Status | Closed |
| Id | 87620 |
In case there was no other event then rel-to/from and the call was dialed with overlap sending
SIP: Handset type information also in Subscribe User-Agent
| Status | Closed |
| Id | 87621 |
Handset type information also in Subscribe User-Agent as sent in REGISTER request.
H.323: Media Negotiation did not work for Hold/Retrieve after a DECT conference
| Status | Closed |
| Id | 87635 |
No media after retrieve
PBX: Conference trap
| Status | Closed |
| Id | 87662 |
A trap in the BC conference PBX object is fixed.
PBX Waiting: Missing ringback on call forward after announcement
| Status | Closed |
| Id | 87674 |
This was a collateral damage of
fix: #81370: PBX Waiting: Call state shows "Disconnecting" after switch from announcement 1 to announcement 2
Gateway: Routing on blind transfer call starts from wrong IF (undo of #86386)
| Status | Closed |
| Id | 87678 |
Rollback of #86386: Gateway: Routing on blind transfer call starts from wrong IF
There are other installations where the old behaviour is the expected behaviour.
PBX Waiting: DTMF overlap dialing or blind transfer to same Waiting object was rejected with busy
| Status | Closed |
| Id | 87681 |
Even if this was caused by a CFB or CFU on the dialed destination
IP-DECT: No idle display update with own message
| Status | Closed |
| Id | 87715 |
A message with the same sender and receiver name overwrites the handset's idle display. This feature is removed now.
Voicemail: <pbx-fwd>, Forward Timer Didn't Fire For Connected Calls
| Status | Closed |
| Id | 87717 |
Voicemail: <pbx-fwd>, Forward Timer Didn't Fire For Connected Calls
<!-
pbx_vm.cpp
-->
IP-DECT: Trap with rejected handovers
| Status | Closed |
| Id | 87730 |
A trap occurs in the rare situation if the handover is rejected by the radio.
H.323: Q.931 User Info could not be sent in incoming call proceeding state
| Status | Closed |
| Id | 87740 |
This is used by the Fax interface
SIP: Parsing of multiple Allow lines was not implemented
| Status | Closed |
| Id | 87753 |
Now all Allow lines are parsed (not only first one).
syslog packets were sent with the initially assigned source address even after a new address had been assigned
| Status | Closed |
| Id | 87754 |
When a device runs as DHCP client the IP-adress assigned to an interface may change either because the DHCP-Server rejects a renew request and provides a new lease or because a WLAN device enters another network.
An IP-address change may also happen when the DHCP mode of a device is changed from 'disabled' to 'client' without reboot.
The source address of syslog packets does reflect such changes now.
Linux: Trap of IP810 if Linux is started
| Status | Closed |
| Id | 87763 |
The IP810 with the V9 hotfix 16 firmware traps if Linux is started. Please do not use this firmware. It is fixed again now.
PBX: CFNR to same node but different PBX failed on Slave
| Status | Closed |
| Id | 87772 |
The PBX routing did not work correctly in this case
PBX administration: Viewer could change phone configuration of users
| Status | Closed |
| Id | 87774 |
SIP: Don't try to dns-resolve IP adresses
| Status | Closed |
| Id | 87781 |
Don't try to dns-resolve IP adresses.
phone: Function key options which need to be enabled in Phone/Userx/Preferences could be edited on phone although disabled
| Status | Closed |
| Id | 87792 |
Partner-Intrude: hide when "Phone/Userx/Preferences/Enable Call Intrusion" is not checked
Dial-Announce: hide when "Phone/Userx/Preferences/Announcement Calls/Outgoing/Allow" is not checked
SIP: Error handling huge INVITE requests
| Status | Closed |
| Id | 87811 |
Failed to compose provisional and final response.
Lots of error messages in trace and truncated responses.
ERROR: SIP message buffer (2049) exceeded! (40,35,941b0c90)
Gateway: MOH from call on hold in CONF interface calls
| Status | Closed |
| Id | 87823 |
If a new call joins the conference or a call on hold retrieves the conference, and there are calls on hold in the conference, the music on hold can be heard in the new or retrieving calls. Now this is fixed.
This affects all devices with a CONF interface, but not the IP800 and the IP305.
IP241,IP222,IP232: Presence note may is not enough truncated on 'presence' fkey
| Status | Closed |
| Id | 87825 |
Presence note may is not enough truncated on 'presence' fkey.
Esp. when there's also an icon to be displayed.
PBX Trunk: If a call was forked to a trunk with no-name option, name was removed from the original call as well
| Status | Closed |
| Id | 87843 |
A global flag was set on the incoming call, which was wrong.
IP230 IP240: Handset receiver volume increased
| Status | Closed |
| Id | 87878 |
receiver volume adapted to changed equalizer settings from fix #80403
phone: ip222, ip232: better handling of spurios headset disconnects
| Status | Closed |
| Id | 87908 |
Sometimes USB headsets get disconnected from USB port because of certain electric pulses. To overcome this problem the headset port is reset and the media stream routed to the handset. If the headset comes up (logical plugged) again in a reasonable time the media stream is routed to the headset again. Otherwise the media stream remains on the handset and the call can be continued by taking off the handset.
PBX Trunk: Diverting as Calling Feature should replace the name as well
| Status | Closed |
| Id | 87913 |
With this feature the Trunk object uses a Diverting Number as calling party number. But not only the number, but also the Name and Name Id should be replaced
Phones: Allow lcd_dump.bmp to be retrieved with viewer credentials
| Status | Closed |
| Id | 87945 |
Allow lcd_dump.bmp to be retrieved with viewer credentials.
IP-DECT: Wrong radio list after MAC-alias change
| Status | Closed |
| Id | 87950 |
The radio list can be wrong after the MAC-alias change. This fixes a bug of the feature "MAC-alias change of OEM device" (#86047). This is only relevant for OEM devices.
SIP: Keep Contact-URI when registering via TCP
| Status | Closed |
| Id | 87969 |
Keep Contact-URI when registering via TCP, even if rport is present in 200/OK for REGISTER
PBX-SOAP: UserCall with 'cn' as destination to a user in different node did not work
| Status | Closed |
| Id | 88008 |
The call is done using the number, but the number needs to be adjusted according to the nodes
OS: Quota mechanism did not work for CPU time consumed by timer handling
| Status | Closed |
| Id | 88034 |
This could result in a MAX_BUSY_TICK restart if many timer expired at the same time
HTTP: Chunked transfer sporadic fails with webdav
| Status | Closed |
| Id | 88078 |
IP6000 IP2000: Webinterface for ETH1 Link setting didnt work
| Status | Closed |
| Id | 88137 |
..
SIP: Decoding problem on application/simple-message-summary
| Status | Closed |
| Id | 88188 |
application/simple-message-summary may get decoded wrongly.
Voicemail: Possibility To Switch Off An Internal Automatism
| Status | Closed |
| Id | 88246 |
An internal automatism could lead to VM-calls getting disconnected after 15s.
The new URI variable "$_divconn=false" turns off auto-connection for diverted/transferred calls:
http://wiki.innovaphone.com/index.php?title=Howto:Configure_the_innovaphone_Voicemail#URL_Query_String_Variables
SIP: DNS priority value not honored always
| Status | Closed |
| Id | 88248 |
DNS priority value not honored, if SRV query returns names without address in additional records.
SIP: Do not take "9564+4631559300" as E.164 number
| Status | Closed |
| Id | 88254 |
Do not take "9564+4631559300" as E.164 number.
PBX Boolean: Unpredictable behaviour if more the 16 times are entered
| Status | Closed |
| Id | 88263 |
There was a limit of 16 times, which was not checked everywhere. The limit is now 32 and checked, so that no more of 32 times can be entered.
IP222 IP232 IP241: Restart on Jitter buffer overrun
| Status | Closed |
| Id | 88312 |
..
SIP: Fix for auto-answer handling
| Status | Closed |
| Id | 88334 |
Fix for auto-answer handling.
IP-DECT: SIP/Intop - rejected call transfer by target
| Status | Closed |
| Id | 88356 |
If the call transfer target rejects the call in ringing state, no fall-back to the initiator call is done and it is not released. This is fixed now. It is only important for a third party PBX.
H.323: Problem with Media Re-Negotiation on a DECT handover call
| Status | Closed |
| Id | 88379 |
The DECT handover call works a little special concerning media renegotiation in a way that local preferences are never honored (the real media negotiation takes place between the original radio and the remote endpoint, the handover radio is just told the result. This special mode did not work correctly
PBX Waiting: User Information Message from announcement interface accidentally forwarded to caller
| Status | Closed |
| Id | 88426 |
The announcement interface uses User Information signaling messages to send status information for example at the end of the announcement. This was forwarded to the caller by accident. Usually this does not do any harm, but on some ISDN networks it could result in clearing of the call because of unexpected message.
H.323: Potential Max-Busy-Ticks restart
| Status | Closed |
| Id | 88441 |
The H.323 state machine could enter a endless loop
IP-DECT: No media after conference and toggled to held call
| Status | Closed |
| Id | 88442 |
If the DECT user leaves the conference mode and toggles to an held call, there is no voice. This is fixed now.
Admin IP: static routes configured at an ETH interface disappear when NAT or VLAN config of this interface is updated
| Status | Closed |
| Id | 88451 |
Routes configured under "IP4/ETHn/IP/Static IP Routes" were cleared when the "IP4/ETHn/NAT" page or the "ETHn/VLAN" page was left by pressing "OK".
IP232,IP222,IP241: Reduce memory requirements of display rendering
| Status | Closed |
| Id | 88485 |
Reduce memory requirements of display rendering.
PBX: Call Completion monitoring call should be terminated only after the CC ringout call is alerting
| Status | Closed |
| Id | 88489 |
This is needed for QSIG interoperability
Viewer was able to download config with standard password
| Status | Closed |
| Id | 88503 |
This was a security hole
Gateway: Hanging calls in state "Clearing"
| Status | Closed |
| Id | 88527 |
Hanging calls in state "Clearing".
Caused by failed call replacement.
IP-DECT: Potential buffer overrun
| Status | Closed |
| Id | 88535 |
Buffer overrun check added in dectusers module for command 'show'.
SIP: Missing UPDATE on call pickup
| Status | Closed |
| Id | 88550 |
The caller does not received UPDATE containing the connected party information.
SIP: Handle one way of DTMF only
| Status | Closed |
| Id | 88572 |
Handle DTMF received via signaling message (INFO)
or DTMF received via RTP (tlephone-event).
Not both at the same call.
PBX Conference: No Media in case media offer only received with alert/connect from called member
| Status | Closed |
| Id | 88590 |
This could happen if a broadcast object was called by a conference
IP222 IP232 IP241: Adjust equalizer above 6.3Khz for lower noise level
| Status | Closed |
| Id | 88658 |
to reduce noise with G722 codec
SIP: Memory leak
| Status | Closed |
| Id | 88688 |
Memory leak on postponed RETRIEVE_NOTIFY.
SIP: Invalid SDP answer if SDP offer contains RED
| Status | Closed |
| Id | 88700 |
Invalid SDP answer if SDP offer contains RED.
Only on local media.
Voicemail: <pbx-query-obj type="filter"/> couldn't read template distributed filter
| Status | Closed |
| Id | 86459 |
Switched to a different internal api method
AD Replication: Refactoring Poll-Timer-based Replication
| Status | Closed |
| Id | 86476 |
-Ridded usnChanged filter term
-Added two status messages
PBX Mobility: Take precaution against hanging calls because of misbehaving mobile phone or client
| Status | Closed |
| Id | 86813 |
Use a 20s timer to terminate any call to the mobility object if no internal call is initiated
IPVA: Query Physical Link Status.
| Status | Closed |
| Id | 86937 |
-The physical link status wasn't evaluated so far. Now it is.
-Added statistics page 'Interfaces/EthX/Statistics'
IP TOS value is now traced with Wireshark
| Status | Closed |
| Id | 87025 |
This value is now correctly traced within IP4 UDP/TCP traces.
PBX Trunk: New destination for rejected calls
| Status | Closed |
| Id | 87151 |
Calls which are rejected (busy after alert), can be redirected to a configurable destination.
SIP: Offer media encryption as separate media description
| Status | Closed |
| Id | 87152 |
Interop issue.
New config file option "config change SIP /separate-encryption".
Offer:
\tv=0
\to=- 10 1 IN IP4 172.16.16.156
\ts=-
\tc=IN IP4 172.16.16.156
\tt=0 0
\tm=audio 16390 RTP/AVP 9 8 0 18 101 13
\tc=IN IP4 172.16.16.156
\ta=rtpmap:101 telephone-event/8000
\ta=fmtp:18 annexa=yes
\ta=fmtp:18 annexb=no
\ta=fmtp:101 0-15
\ta=ptime:30
\ta=silenceSupp:off - - - -
\ta=sendrecv
\tm=audio 16390 RTP/SAVP 9 8 0 18 101 13
\tc=IN IP4 172.16.16.156
\ta=rtpmap:101 telephone-event/8000
\ta=fmtp:18 annexa=yes
\ta=fmtp:18 annexb=no
\ta=fmtp:101 0-15
\ta=ptime:30
\ta=silenceSupp:off - - - -
\ta=sendrecv
\ta=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:VoGZgKwImVwTkJg4jBeYFBafl/CyJpfMX66WqDMZ
Instead of:
\tv=0
\to=- 9 1 IN IP4 172.16.16.156
\ts=-
\tc=IN IP4 172.16.16.156
\tt=0 0
\tm=audio 16386 RTP/SAVP 9 8 0 18 101 13
\tc=IN IP4 172.16.16.156
\ta=rtpmap:101 telephone-event/8000
\ta=fmtp:18 annexa=yes
\ta=fmtp:18 annexb=no
\ta=fmtp:101 0-15
\ta=ptime:30
\ta=silenceSupp:off - - - -
\ta=sendrecv
\ta=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:mDPFKhNdhm6fhkeyPbAI9uzZyTmtA4t9fy1MwXiD
IP-DECT: DECT security (4)
| Status | Closed |
| Id | 87157 |
Some changes in preparation for the new feature "DECT security". This feature will be available later (IP1202).
SIP: Do registration refresh more ahead of expiration
| Status | Closed |
| Id | 87381 |
Instead of refreshing 2 secs before registration expiration,
do the refresh after 98,33% of the registration ttl.
TTL=120secs -> Refresh 2secs before expiration
TTL=3600secs -> Refresh 60secs before expiration
phone: Modification of function key on phone can be disabled for any single key
| Status | Closed |
| Id | 87405 |
A "Disable Modification on Phone" checkmark will be provided in the edit menu for each key. If checked the key cannot be edited on the phone anymore.
This mechanism works in addition to the phone local key type mask set via
"Phone/Protect/Function keys not modifiable on the phone"
A key of a type NOT marked as ``not editableïï in this mask can be made ``not editableïï by setting the above mentioned checkmark
A key of a type marked as ``not editableïï remmains not editable, independent of the checkmark setting.
phone: "Spare" function key to reserve key positions for administrative purposes
| Status | Closed |
| Id | 87406 |
This new key permits to reserve key positions for later definition by administrator.
IP-DECT: Remote hold in conference mode
| Status | Closed |
| Id | 87487 |
If a hold notify message is received from a remote party and the conference mode is active, now the message is forwarded to the conference unit. This prevents the music on hold in conference calls. The state is also shown in the radio call list.
Webdav: Write information into trace if DELETE fails because file is in open state
| Status | Closed |
| Id | 87488 |
Write information about Webdav session holding open file handle.
H.323: Better error handling in case of incompatible SRTP parameters
| Status | Closed |
| Id | 87624 |
Was needed for DECT OEM.
PBX: No CDR was generated for enblock call, which was rejected because of filter config
| Status | Closed |
| Id | 87869 |
A CDR with cause 52 (outgoing call barred) is generated
Voicemail: Add <pbx-getcallinfo out-leg2-orig="...">
| Status | Closed |
| Id | 87880 |
Allows to query the divertingLegInformation2.orignalCalled number
IP-DECT: Fault logging for Master module
| Status | Closed |
| Id | 88005 |
Call and channel fault logging for the IP-DECT Master module are added.
Voicemail: <pbx-prepcallinfo leg2=".." leg2-name=".."/>
| Status | Closed |
| Id | 88351 |
New attributes allow to prepare <pbx-fwd> in such, that the supplementary service divertingLegInformation2 is going to be sent.
PBX: Allow configuration of default presence/dialog-info visibility for group members
| Status | Closed |
| Id | 88352 |
Active group members got full presence/dialog-info because this matched the visibility be group-indications. However this is not desired always, so it can now be configured to restrict this.
IP-DECT: Unused OEM modules removed
| Status | Closed |
| Id | 88413 |
Unused OEM modules has been removed now.
Gateway: Make interop flag "Ack incoming call" configurable on UI
| Status | Closed |
| Id | 88499 |
This was a hidden interop flag, but was now needed multiple times, so it is more efficient to have it configurable on the UI
IP-DECT: Logging for handover calls
| Status | Closed |
| Id | 88536 |
Logging events for IP-DECT handover calls added.
V9 Hotfix 18 (9061158)
Changes included in Version 9 hotfix18 Definition
TLS: Do not ignore early SOCKET_RECV
| Status | Closed |
| Id | 88668 |
Allow SOCKET_RECV from application between SOCKET_CONNECT and SOCKET_CONNECT_COMPLETE. This event flow is used by the HTTP client.
Kerberos: Admin UI trap when having too many Kerberos hosts
| Status | Closed |
| Id | 88698 |
The problem occured if many Kerberos hosts (~1000) were registered on the server. In this case the box trapped due to an XML encoding problem when opening the page General/Kerberos or PBX/Config/Security.
SIP: Provide display name in 200/OK for SUBSCRIBE
| Status | Closed |
| Id | 88764 |
Add display name to To header in 200/OK for SUBSCRIBE.
IP22 IP24 IP28 IP305: Sometimes the DSP stops after sending CLIP
| Status | Closed |
| Id | 89130 |
..
PBX: Configuration UI broken for checkmarks on PBX/Config/General and PBX/Config/myPBX
| Status | Closed |
| Id | 89140 |
The checkmarks on these two pages where all cleared when the other page was edited.
Collateral damage of
fix: #88352: PBX: Allow configuration of default presence/dialog-info visibility for group members
phone: ip222,ip232: the destination of an acticve diversion was not saved when the diversion was deactivated at phone
| Status | Closed |
| Id | 89177 |
The destination (name or number) disappered when the diversion was deactivated. It was also not visible via the WEB interface.
Gateway: Missed FAX pages received with the FAX interface
| Status | Closed |
| Id | 89587 |
If the FAX interface is used to receive a FAX document with ECM mode and the transmitting terminal appends additional EOLs, the page counter is wrong and document pages are not written. This is fixed now.
IPVA: Enhancement for innovaphone testbed: Indicate whether Upload to DRAM took place
| Status | Closed |
| Id | 88223 |
Required for innovaphones' automated testbed
Announcement Calls with DTMF feature code object
| Status | Closed |
| Id | 88643 |
A new DTMF feature code allows to make announcement calls to a dialed number.
Alarm and Event forwarding to a SYSLOG server
| Status | Closed |
| Id | 88659 |
If "Services/Logging/Alarm and Event Forward Server/Type" is set to SYSLOG the xml-formatted alarm and event info is sent to the Server(s) specified under "Services/Logging/Alarm and Event Forward Server/Address".
V9 Hotfix 19 (9061180)
Changes included in Version 9 hotfix19 Definition
SIP/TLS: Rejecting server certificate
| Status | Closed |
| Id | 88444 |
Validating server certificate against configured domain name.
Must be validated against configured proxy domsina name.
SIP: Wrong branch value in Via header in ACK request
| Status | Closed |
| Id | 89317 |
Branch value in Via header in ACK request must be new after 200 response.
Branch value in Via header in ACK request must be same after non-200 response.
H.323: RTP-DTMF did not work on exclusive coder/media relay configurations
| Status | Closed |
| Id | 89328 |
Problem for DTMF on SIP trunks
Voicemail: <pbx-getcallinfo out-confid="...">, pass conference guid into a script
| Status | Closed |
| Id | 89332 |
In order to allow correlation of CDRs to voicemail-recorded files
AD Replication: A Buffer for Processing The Paged Result Cookie Was Too Small
| Status | Closed |
| Id | 89385 |
1KB wasn't enough, now 2KB.
LDAP Expert: "Next"-Browsing through DB failed
| Status | Closed |
| Id | 89391 |
URI encoding error
phone: ip222, ip232,ip241: the notification tone indicating a new message when a call is active was sent to remote
| Status | Closed |
| Id | 89392 |
instead to notify the receiver of the message the remot party did hear the tone
SIP: "Supported: timer" missing in UPDATE message
| Status | Closed |
| Id | 89429 |
"Supported: timer" missing in UPDATE message.
phone: ip222,ip232: humming noise in USB headset speaker in outbound call setup phase, disappears once connected
| Status | Closed |
| Id | 89432 |
sometimes a humming noise was heard in the USB headset speaker in the setup phase of an outbound call. it disappeared as soon as the call was connected.
phone: an intrusion call set up via Partner function key could not be cleared at the intruding phone via TAPI
| Status | Closed |
| Id | 89443 |
The TAPI interpreted the recording state as a conference
log message forwarding to another innovaphone device did not work since V9hotfix15 (on the receiving device)
| Status | Closed |
| Id | 89492 |
this problem is located on the reciving device, it does not depend on the the version of the forwarding device.
IP22 IP24 IP28 IP305: DSP debug code added
| Status | Closed |
| Id | 89493 |
..
PBX Mobility: Conference ID not set for outgoing calls initiated by myPBX
| Status | Closed |
| Id | 89502 |
This could cause several problems:
- When the call was sent to a local user with multiple registrations, the call to each registration had a different conferenceID, so myPBX could not match these calls to actually being only a single call, so multiple calls were dissplayed
- The CDRs created for this call could not be matched
SIP: Don't tell application that registration is down when handling redirect response
| Status | Closed |
| Id | 89582 |
Don't tell application that registration is down when handling redirect response for REGISTER.
SIP: No T.38 parameter when indicating capabilitity only
| Status | Closed |
| Id | 89617 |
No T.38 parameter when indicating capabilitity only.
Offer
\tv=0
\to=- 2 1 IN IP4 172.16.16.124
\ts=-
\tt=0 0
\tm=audio 16386 RTP/SAVP 8 101 13
\tc=IN IP4 172.16.16.124
\ta=rtpmap:101 telephone-event/8000
\ta=fmtp:101 0-15
\ta=ptime:20
\ta=silenceSupp:off - - - -
\ta=sendrecv
\ta=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:WJl714I+mpSr47ld1YjYMf8t9xQo0xYHUng1CnDi
\tm=image 0 udptl t38
\tc=IN IP4 172.16.16.124
Instead of
\tv=0
\to=- 2 1 IN IP4 172.16.16.124
\ts=-
\tt=0 0
\tm=audio 16394 RTP/SAVP 8 101 13
\tc=IN IP4 172.16.16.124
\ta=rtpmap:101 telephone-event/8000
\ta=fmtp:101 0-15
\ta=ptime:20
\ta=silenceSupp:off - - - -
\ta=sendrecv
\ta=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:sPngj1zoO9wI1pc1tMTymWCzTgwMoamPuJAFIhga
\tm=image 0 udptl t38
\tc=IN IP4 172.16.16.124
\ta=T38FaxVersion:0
\ta=T38MaxBitRate:14400
\ta=T38FaxFillBitRemoval:0
\ta=T38FaxTranscodingMMR:0
\ta=T38FaxTranscodingJBIG:0
\ta=T38FaxRateManagement:transferredTCF
\ta=T38FaxUdpEC:t38UDPRedundancy
OS: Potential trap when starting a timer, which is already deleted
| Status | Closed |
| Id | 89628 |
Could result in a strange trap with PBX transfer/recall
phone: while recording was running for a call this call ould not be cleared via SOAP/TAPI
| Status | Closed |
| Id | 89629 |
The TAPI assumed the intruding phone to be in a conference
PBX Admin UI: Reconfiguring user visibility (Access) did not work correctly
| Status | Closed |
| Id | 89636 |
If a setting for a user/group was deleted, it could happen that some of the settings (Group, Online, Presence, Dialog, Ids) where copied to the next entry.
Voicemail: Trap
| Status | Closed |
| Id | 89648 |
If Trace-Checkmark activated, a diverted call carrying an mwi interrogate was processed
Possible Trap While Getting Post-Mortem Log
| Status | Closed |
| Id | 89652 |
Possible Trap While Getting Post-Mortem Log
Media: Redirecting SRTP streams for NAT clients only after successful SRTP authentication
| Status | Closed |
| Id | 89661 |
Media endpoints support NAT.
If receiving RTP/SRTP from an address other than negotiated one
media endpoints redirecting their media stream towards source of incoming media stream.
In case of SRTP, this NAT workaround is only executed if incoming media stream has passed authentication.
For securitiy reasons.
AD Replication: Merge v10 code into v9 code
| Status | Closed |
| Id | 89671 |
Some recent refactoring works weren't in sync. A state machine didn't reach "Completed".
phone: if a call ringing while the handset was offhook was accepted via SOAP/TAPI the call was not cleared when going onhook
| Status | Closed |
| Id | 89680 |
This could happen when the handset was lifted and kept lifted after the disconnect key was prressed. an inbound call arriving in this state could be accepted via SOAP/TAPI and was connected to the handset but the call could not be cleared by going onhook. Only the disconnect key did clear the call.
Now the call is cleared as expected when going onhook.
phone_orchid: pressing speaker key in handset/headset mode switches to handsfree mode, pressing again returns to previous mode
| Status | Closed |
| Id | 89730 |
handset/headset plus speaker is not supported on phone_orchid, the previous solution where the connection was dropped when the speaker key was pressed again (see #84297) was perceived as irritating.
IP22 IP24 IP28 IP305: Sometimes the DSP stops after sending CLIP (2)
| Status | Closed |
| Id | 89760 |
..
SIP: Bug in handling of INVITE with Replaces
| Status | Closed |
| Id | 89777 |
Bug in handling of INVITE with Replaces.
Results in hanging call.
IP-DECT: Trap with call transfer
| Status | Closed |
| Id | 89786 |
A trap occurs if a call transfer is received in the IP-DECT radio module. This is fixed now.
Fix for MIPS counter
| Status | Closed |
| Id | 89804 |
MIPS counter was incorrect on IP1201 and IP4001
SIP: Media negotiation fails on calls into Waiting Queue
| Status | Closed |
| Id | 89838 |
Media negotiation fails on calls into Waiting Queue if
caller put call on hold before WQ agent accepts the call.
'power-off loop' relay switching function failure
| Status | Closed |
| Id | 89942 |
Especially POE-switches with higher supply voltages than 48V lead to a decreased timespan of powering the build-in relays of a ip6010/ip810 gateway. The detection of a power-fail condition is therefore derived from the POE ICs which react earlier and thus increases powering time of the relays.
myPBX: Default group visibility was not displayed correctly
| Status | Closed |
| Id | 89954 |
The default group visibility can be configured from v9hotfix17. Regardless of that configuration myPBX showed full visibility in the visibilty settings.
IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: DSP Update to Version 680.07
| Status | Closed |
| Id | 89956 |
Fixed modem bypass with slow modems
PBX-SOAP: Present normalized number of peer also
| Status | Closed |
| Id | 88521 |
The SOAP API presents the adjusted number of the peer (called/calling), which is the shortest possible number which can be dialed to call this. It is the same number as displayed on the phone. Sometimes an application needs to know the normalized number of the peer, which is the number in the context of the root node. This number is sent as additional number with the identifier "norm"
IP232,IP222,IP241: Config option to adjust LCD brightness in idle state
| Status | Closed |
| Id | 89261 |
-> Main Menu -> Phone Setup -> LCD light (idle state)
Can be tuned down to zero.
case independence for the characters of the Basic Russian Alphabet added
| Status | Closed |
| Id | 89367 |
mappings added
H.323: Automatically connect signaling TCP if NAT router is detected
| Status | Closed |
| Id | 89497 |
When regestering an endpoint from a private network to a PBX within the public network, the signaling TCP connection must be established and maintained by the endpoint. Otherwise calls to the endpoint are not possible.
PRI-QSIG: Interop config for channel numbering
| Status | Closed |
| Id | 89578 |
The QSIG standard defines to use Channel numbers (1-30) instead of timeslot (1-15, 16-31) as it is defined for EDSS1. There are many 'old' QSIG implementations around, which do it wrong. The QSIG-ECMA1 protocol setting is used for these 'old' implementations and the QSIG-ECMA2 setting for standard conform inplementations.
With the QSIG-ECMA1 also 'old' facility coding is used. There is also the combination of standard facility coding and timeslots for channels around so an independent mechanism to configure the channel numbering is needed.
V9 Hotfix 20 (9061198)
Changes included in Version 9 hotfix20 Definition
SIP: Auto answer with SDP in ACK
| Status | Closed |
| Id | 89539 |
Auto answer results into no-audio when INVITE comes w/o SDP offer.
SIP: Fix for overlap dialing with KPML
| Status | Closed |
| Id | 89581 |
Dialing digits entered before KPML subscription is established
need to be queued until KPML subscription is established.
SIP: Re-try INVITE after 407 even if no password configured
| Status | Closed |
| Id | 90024 |
Calculate Digest with zero-length password and re-try INVITE.
IP232,IP222,IP241: Truncate directory entry information
| Status | Closed |
| Id | 90027 |
Truncate directory entry information to keep from overlapping with number type indication.
IP-DECT: Don't show DTMF in radio call list
| Status | Closed |
| Id | 90034 |
User dialled digits during calls which are sent as DTMF should not be shown in the radio call list. This is fixed now.
H.323: Allow media offers with 0.0.0.0 as address
| Status | Closed |
| Id | 90053 |
Needed for SIP interoperability. Some third party SIP PBXs use addresses of 0.0.0.0 to indicate that they don't receive media. This may happens if an endpoint is put on hold. We did not forward such an offer and thus no Music on Hold was heard.
PBX: Busy On ... Calls at PBX objects did not take into acccount that a call may be routed back to Slave
| Status | Closed |
| Id | 90054 |
The busy on ... calls on PBX objects can be used to limit bandwidth usage between a master and a slave to a certain number of calls. Some calls are sent from a slave to the master and back to the slave if the routing decision cannot be done on the slave alone. This happens if escapes are used which overlap other obects (e.g. the local trunk). It is a common configuration the the E.164 routing scheme.
With this fix, these calls are not counted for this purpose.
Voicemail: Send silence RTP during recording
| Status | Closed |
| Id | 90095 |
Send silence RTP during recording for some SIP carriers that do not send RTP without receiving RTP.
IP2000: Prevent blinking error LED IP2000
| Status | Closed |
| Id | 90102 |
The firmware tried to load the conference DSP, which is not available on the IP2000
SIP: Memory leak when receiving NOTIFY(message-summary)
| Status | Closed |
| Id | 90106 |
Memory leak when receiving NOTIFY(message-summary)
H.323: Support for registration from a private network thru NAT
| Status | Closed |
| Id | 90306 |
In case an endpoint registers to a PBX from within a private network thru a NAT router, the signaling TCP connection must be maintained in order to be able to receive calls. When the registration is up a dummy call is sent to the PBX to establish the signaling TCP. This TCP connection is maintained after the dummy call is cleared. If this TCP connection is lost (e.g. NAT Router reset), the Registration is cleared and restarted, so that after the re-registration another dummy call is sent.
This is a fix for the previous fix
fix: #89497: H.323: Automatically connect signaling TCP if NAT router is detected
which did not work well
SIP: SUBSCRIBE using old IP address in Contact field
| Status | Closed |
| Id | 90320 |
If the IP address is changed at DHCP renew (or network change) the endpoint will immediately do a re-register to update the SIP Proxy with the new IP address.
All SIP messages but SUBSCRIBE uses the new IP address in the Contact field.
PBX: CC Requests were sent with wrong number if a SendNumber was configured at the user
| Status | Closed |
| Id | 90432 |
Usually this is no problem only when interworking with some QSIG PBX's this causes the call-completion to fail.
phone_orchid: dialtone missing when recording is active and the active call is held to open a consultation call
| Status | Closed |
| Id | 90433 |
the consultation call could be established but there was no dialtone after pressing the R-key and no ringback tone after the number had been entered.
Possible trap when doing a leak check
| Status | Closed |
| Id | 90451 |
When many leaks exist or leak check is done when much tracing is turned on. The leak check itself could cause a watchdog trap, because the collecting of the leaks is done on highest priority so not even the timer interrupt could trigger the watchdog.
IP-DECT: Busy state on maximum call count
| Status | Closed |
| Id | 90461 |
The base station does not go to the busy state if the maximum call count is reached and the last call is an incoming call. This is fixed now.
PBX: Support for Opticaller Data Callthru did not work
| Status | Closed |
| Id | 90480 |
Support for Opticaller data callthru was added, but did not work
PBX: Forking a call to a Trunk with "Outgoing call restricted", causes the original call to be restricted as well
| Status | Closed |
| Id | 90487 |
The "Outgoing call restricted" flag on the trunk object to which the call was forked caused the call as a whole to be marked as Calling Line Presentation Restricted.
SIP: Locally configured DNS entries were not used if no DNS server configured
| Status | Closed |
| Id | 90508 |
If no DNS server was configured, but DNS names are to be resolved,
local DNS entries can be added (Services/DNS/Hosts).
SIP stack fails with SRV query and does not try A query which would deliver IP address.
IP-DECT: RTP stream
| Status | Closed |
| Id | 90539 |
If a remote hold event is received, no RTP data should be sent by the IP-DECT device.
A CTI initiated call is established with a call transfer and a "No Media data received" error event can occur.
This is fixed now.
SIP: Trap on IP-DECT
| Status | Closed |
| Id | 90569 |
Trap in GK-CHANNEL when Dectmaster application sends DTMF before call if created.
SIP: Disabled IP-DECT interface tries to register
| Status | Closed |
| Id | 90597 |
Disabled IP-DECT interface tries to register.
H.323: Timer to monitor response to setup too short for some traffic cases
| Status | Closed |
| Id | 90696 |
This created unnecessary event in IP-DECT systems when calling powered off or out of range handsets
PBX Waiting: Potential Trap if editing while a call is initiated with SOAP
| Status | Closed |
| Id | 90766 |
The Waiting object can be used as outgoing dialing object with SOAP. If this is done and the configuration is changed while an outgoing call was pending, a trap could happen
H.323: Potential Trap in special case which could only happen in version 10
| Status | Closed |
| Id | 90768 |
This fixed is merged to version 9 only of consistency reasons
SIP: Respect changes in PAI/PPI header when receiving UPDATE with SDP offer
| Status | Closed |
| Id | 90778 |
PAI/PPI was processed when receiving UPDATE without SDP offer.
PAI/PPI was ignored when receiving UPDATE with SDP offer.
Now PAI/PPI is processed when receiving UPDATE with SDP offer.
SIP: Possible buffer overrun
| Status | Closed |
| Id | 90780 |
Fix for possible buffer overrun.
SIP: Wrong error log "Timeout during media negotiation for call"
| Status | Closed |
| Id | 90821 |
Error log "Timeout during media negotiation for call" may occur after re-negotioation.
Re-negotioation occurs during hold/retrieve/transfer.
PBX Trunk: Name to Number Feature did not work with calls to extern
| Status | Closed |
| Id | 90858 |
If an endpoint cannot be found by name, the call should be forwarded to extern
PBX CSV Import: Corrupted objects at buffer boundaries
| Status | Closed |
| Id | 90942 |
The upload is processed in chunks of 2K. At boundaries of these chunks data could be corrupted. This was fixed and the chunk size increased to 10K
Phone: Trap when selecting registration for a directory entry
| Status | Closed |
| Id | 90999 |
Trap when selecting registration for a directory entry,
but only if the registration has either no name or no number.
H.323: No event should be generated in State 11 and 25
| Status | Closed |
| Id | 91020 |
State 25 is incoming overlap sending. This means a call was received with incomplete dialing information and the caller failed to dial more digits within the timeout of 2min. This is no indication of any malfunction but only a usage problem, so no event should be generated.
State 11 is disconnecting with inband announcement. A timeout happens if a user listens to the announcement for more then 30s. This could be normal.
SIP: Trap when cancelling call
| Status | Closed |
| Id | 91023 |
Double delete of a call entity.
SIP: Heavy TLS retry load when server certificate was rejected
| Status | Closed |
| Id | 91033 |
Collateral damage from earlier fixed for DNS refreshing.
H.323: Incoming faststart call was sometimes not accepted as faststart
| Status | Closed |
| Id | 91136 |
Only happens if non EFC is used on the incoming call, so this only happens in interop cases with other H.323 equipment
SNMP Get-Next Requests Carrying an Octet-String Value Caused Memory Leak
| Status | Closed |
| Id | 91215 |
SIP: Media negotiation for video fails if called through waiting queue or multi reg
| Status | Closed |
| Id | 91235 |
Media negotiation for video fails if called through waiting queue or multi reg.
In this case the PBX has to handle offer/offer-collision.
In this case the PBX must select audio and video codec.
In this case the PBX must send SDP answers to both endpoints.
SIP: Secondary target (hostname) is not resolved
| Status | Closed |
| Id | 91287 |
Usually a response to a SRV query delivers additional records containing the ip address of any target (hostname).
Some DNS servers do not.
Additional A querys are required.
An A query was issued for the primnary target (most preferred hostname).
No A query was issued for the secondary target (less preferred hostname).
Fixed now.
IP-DECT: Trap in Radio module (IP1202)
| Status | Closed |
| Id | 91315 |
A trap in the IP-DECT Radio module occurs if the Mobility Master is used and a duplicate IPEI command is sent to the Master. The Master handles it with a location cancel and an endpoint delete command sent to the radio. If the two commands arrives with no delay, the Radio module traps. This is fixed now.
Ldap Replication from NDS
| Status | Closed |
| Id | 91347 |
Skip isDeleted attribute with content others than 'true'. Occurred when replicating from an NDS running in AD compatibility mode.
SIP: Wrong call was disconnected after successful transfer
| Status | Closed |
| Id | 91349 |
Wrong call was disconnected after successful transfer.
SIP: SDP answer for T.38 switch-over must contain multiple media descriptions
| Status | Closed |
| Id | 91377 |
... if SDP offer contains multiple media descriptions.
PBX: Not possible to login as user with non-full admin rights if 'Password protect all Pages'
| Status | Closed |
| Id | 91414 |
Some pages needed for the UI, which are normally not password protected, could not accessed with the reduced rights
SIP: Follow offers ptime proposal
| Status | Closed |
| Id | 91421 |
Better follow offers ptime proposal.
Otherwise SAMwin operator does not stop sending re-INVITE.
IP0010 IP1060 IP3010 IP6010 IP22 IP24 IP28 IP302 IP305: switch from modem bypass to voice did not work
| Status | Closed |
| Id | 91429 |
In some cases announcments with music trigger modembypass, in this case at least the fallback to voice should work.
H.323: Allow update of Registration password
| Status | Closed |
| Id | 91440 |
needed in case a password is changed, which is replicated to a DECT system
H.323: PROGRESS in connected state was treated as 'unexpected'
| Status | Closed |
| Id | 91483 |
This happened with H.323 connections without registration when disconnecting a call with inband information (e.g. a call to an ISDN interface). Unnecessary events were generated.
Linux: Memory allocation changed for IP810
| Status | Closed |
| Id | 86420 |
The memory allocation for the IP810 is changed to 128MB/384MB for innovaphone/Linux.
Important:
Linux must be used with the kernel version 3.4.10 or later. This kernel is included in the Linux Application Platform V9.00 hotfix12 and later. The kernel is automatically updated with the Linux Application Platform V9.00 hotfix12.
phone: new "Do Not Disturb" action "ring once"
| Status | Closed |
| Id | 89960 |
If "Phone/User-x/Preferences/Do Not Disturb/Action: ring once" is selected a new inbound call is indicated with a short tone only.
Both the the tone and the duration of the tone can be configured under ""Phone/User-x/Preferences/Ring Tones/Do Not Disturb".
If not configured the default ring tone is played for one and a half second.
PBX: Support of long user-user-informations by SOAP
| Status | Closed |
| Id | 90029 |
Support of long user-user-informations (UUI) for SOAP sessions added. A long UUI is split into multiple short UUIs supported by Q.931.
It is required by the FAX interface.
Relay: Support of long user-user-informations by FAX
| Status | Closed |
| Id | 90030 |
Support of long user-user-informations (UUI) for the FAX interface added. A long UUI is split into multiple short UUIs supported by Q.931.
Gateway: Support of a header line for FAX documents
| Status | Closed |
| Id | 90374 |
Support of a header line for FAX documents is added in the FAX interface.
SIP: Support for MESSAGE inside voice call
| Status | Closed |
| Id | 90408 |
Support for text messages inside voice call.
PBX-SOAP: Support for 'rc' and 'srce164' on UserCall on Waiting Queue
| Status | Closed |
| Id | 90537 |
If a Waiting Queue is used for outgoing calls, these features can be usefull for some applications
PBX: New configuration option 'Hide connected Number' at object
| Status | Closed |
| Id | 90693 |
In some cases it is desireable not to reveal the final destination of a call to a caller. For example a call center agent should not be called directly by the customer.
IP-DECT: Configuration option 'Registration with system password'
| Status | Closed |
| Id | 91460 |
The configuration option 'Registration with system password' is added. If ticked, all users are registered with the system password. This is useful, if the PBX users are only allowed to register with the PBX password.
Voicemail: URL-En-/Decoding
| Status | Closed |
| Id | 91609 |
new statement allows to URL-encode or URL-decode a string
<lib-enc string=".." string_out="$var" type="url"/>
<lib-dec string=".." string_out="$var" type="url"/>
V9 Hotfix 21 (9061222)
Changes included in Version 9 hotfix21 Definition
PBX Mobility: Connected number from mobile phone was forwarded to caller
| Status | Closed |
| Id | 90722 |
If a call is answered on the mobile phone, it should look identical to the caller to the case that the call was answered locally. This means a connected number from the mobile phone must not be forwarded.
Potential Trap when rapidly switching local Media connections (Conferencing)
| Status | Closed |
| Id | 90933 |
There was a race condition when switching local media channels (e.g. ISDN channels to conference interfaces), which could cause media not functioning or even a trap
SIP: Better handling of incoming calls
| Status | Closed |
| Id | 91153 |
Msg sequence INVITE,CANCEL,INVITE may result into second call rejected.
IP0010,3010,6010,1060,810: ethernet link down not detected when a cable was unplugged after boot
| Status | Closed |
| Id | 91600 |
The link state interrupt was triggered only once after boot, further link state changes were not indicated.
myPBX: Rejecting an incoming broadcast call disconnected the call for all alerting endpoints
| Status | Closed |
| Id | 91719 |
The call sould continue alerting on the other endpoints.
phone: ip222, ip232: reset/restart USB headset when a hang condition or a port disconnect condition is detected
| Status | Closed |
| Id | 91732 |
For unknown reasons some types of wireless headsets stop working after some hours or days. Either the port state changes to disabled or the device rejects control commands with a stall response. In both cases the device is reset and restarted now. If even this fails the complete USB host controller is reset and in most cases the device returns to operational state thereafter
phone: ip222, ip232: Some USB headsets were not detected after a soft reset
| Status | Closed |
| Id | 91734 |
This was observed with Jabra BIZ 2400 USB at IP222/232 with hardware build 800 and newer.
IP-DECT: Trap with data calls
| Status | Closed |
| Id | 91807 |
The IP-DECT Radio traps if a data call is released and the release includes a facility.
Kerberos administration: Increase maximum number of Kerberos users from 20 to 50
| Status | Closed |
| Id | 91847 |
Increase limit on page General/Kerberos.
Use POST for submitting form, instead of GET.
Voicemail: <pbx-getcallinfo out-leg2-name=".." out-leg2-orig-name="..">
| Status | Closed |
| Id | 91874 |
Passing name info of divertingLegInformation2 facility into the script.
PBX-SOAP: Potential Trap with UserClear
| Status | Closed |
| Id | 91886 |
A SOAP applicatoin (e.g. TAPI) uses the method UserClear to clear a call. This could cause a trap on some platforms when doing this for a mobile endpoint.
SRTP: Avoid one-way media with high start sequence numbers
| Status | Closed |
| Id | 91892 |
In some cases SRTP calls had one-way media because the RTP sequence number wrapped from 65535 to 0 at be beginning of the call before the receiver started receiving and processing packets.
The scope of start sequence numbers for RTP streams is changed from [0;65535] to [0;32767] to make sure that the receiver can always receive packets before the overflow happens.
The calculation of the roll-over counter (ROC) is also improved to be more reliable.
H.323: Unnecessary re-initializing of rtp-channel on incoming calls to phone
| Status | Closed |
| Id | 91898 |
This did not create any problems except CPU load and together with another problem in RTP it caused no media on incoming SRTP calls approximately every 1000th call.
IP22,IP24,IP28,IP302,IP305: RTP-DTMF not offered when using a/b interface
| Status | Closed |
| Id | 91905 |
For example:
\tv=0
\to=- 14 1 IN IP4 10.17.1.91
\ts=-
\tt=0 0
\tm=audio 16414 RTP/AVP 8 0 18 4 97
\tc=IN IP4 10.17.1.91
\ta=rtpmap:97 CLEARMODE/8000
\ta=fmtp:18 annexa=yes
\ta=fmtp:18 annexb=yes
\ta=ptime:20
\ta=silenceSupp:off - - - -
\ta=sendrecv
PBX: MOH URL parameters for parked calls should refer to the parked-to and not parking user
| Status | Closed |
| Id | 91931 |
The MOH URL Paramter (%l, %h, %n, ...) can be used to use different MOH Files based on the User who is holding the call. In case of a parked call this should refer to the object where the call is parked, not to the user who has initiated the parking.
IP-DECT: Wrong XML data in Radio endpoint
| Status | Closed |
| Id | 91933 |
The XML data in the Radio endpoint is wrong and fixed now. The data is only used for the command show-endpoints, not for the GUI.
PBX Mobility: Call from mobile endpoint to a user with mobility, but without PBX phone was disconnected
| Status | Closed |
| Id | 91976 |
This was a collateral damage of
fix: #86813: PBX Mobility: Take precaution against hanging calls because of misbehaving mobile phone or client
SIP/DNS: Wrong port used on secondary SIP server
| Status | Closed |
| Id | 91995 |
If SRV query returns 2 hosts with different port, but no IP address in additional records,
SIP starts two A queries for the two host names.
Both resolved IP addresses are combined with the port of the most preferred host of the SRV answer.
IP-DECT: Hold/Retrieve could result in no media for incoming SIP calls with SRTP
| Status | Closed |
| Id | 92014 |
This was in fact a H.323 media negotiation problem between master and slave
SIP/WLAN: Keep local Contact-URI up-to-date on subscriptions
| Status | Closed |
| Id | 92134 |
Keep local Contact-URI up-to-date on subscriptions (e.g. message summary) when local ip address changes at runtime.
IP-DECT: Hold and Retrieve with SIP and SRTP could result in no media
| Status | Closed |
| Id | 92189 |
Problem with forwarding changed SRTP Keys from master to radio
phone: ip241: do headset volume control in digital domain
| Status | Closed |
| Id | 92203 |
IP232,IP222,IP241: Reduce flicker when opening details of call list entry
| Status | Closed |
| Id | 92242 |
Reduce flicker when opening details of call list entry.
IP-DECT: Potential trap when switching to an from 3pty conference
| Status | Closed |
| Id | 92262 |
In case of a very unlikely collision of events, a message was sent to an already deleted object, which cause a restart
IP232,IP222,IP241: Fix for display of international numbers on phone UI
| Status | Closed |
| Id | 92335 |
Show international numbers as +4930123456 instead of I4930123456.
PBX: Call Completion to a user without registration, but forking, caused immediate Callback
| Status | Closed |
| Id | 92341 |
Instead to call completion request should be rejected in this case
SIP: Huge SIP messages causes out-of-sync on TCP stream
| Status | Closed |
| Id | 92373 |
Reading SIP messages from TCP stream gets confused by huge SIP messages.
Presence exchange with external UC was disordered.
Increased size limit from 100KByte to 200KByte.
Gateway: Transmitting FAX documents to receiver with polling mode
| Status | Closed |
| Id | 92388 |
It is not possible to send fax documents to a receiver with polling capability. This is fixed now.
IP-DECT: Cipher key index table update function
| Status | Closed |
| Id | 92467 |
The cipher key index table is wrongly updated in the Crypto Master if a entry line yet exists. This is fixed now.
The Crypto Master is needed for DECT Security Early Encryption.
IP222 IP232: Handset gains changed to avoid low microphone volume
| Status | Closed |
| Id | 92577 |
Low microphones levels were squelched.
Voicemail: Emailing file names with '#'-characters failed
| Status | Closed |
| Id | 92757 |
An Escape mechanism wasn't applied
Webmedia: New URL parameter "fallback=true"
| Status | Closed |
| Id | 92957 |
New URL parameter "fallback=true".
If given and the specified media file cannot be retrieved,
webmedia starts playing builtin MOH.
Status:
Implemented in 10.00 und 9.00
Trap On Ldap DOS Attack
| Status | Closed |
| Id | 92978 |
Malign asn.1 content wasn't rejected
IP800 IP6000 IP2000: duplicated DTMF digits in transcoding scenarios
| Status | Closed |
| Id | 92981 |
When tandeming VOIP links for trancoding or other purposes DTFM digits were sometimes duplicated.
The RTP carried up to 25ms DTMF remaining DTMF, now its only 16ms.
PBX: Trap if user object is deleted, which is used by other applications (e.g. myPBX)
| Status | Closed |
| Id | 92985 |
The applications need to cleanup in this case
IP-DECT: CSV user export
| Status | Closed |
| Id | 92991 |
There are some missed users in the CSV user export file. It occurs if there are users with login rights. This is fixed now.
IP0010,3010,6010,1060,810: limit number of ethernet packets processed per receive interrupt
| Status | Closed |
| Id | 93130 |
this prevents that too much time is spent in ethernet driver in case of broadcast storms or DOS attacks.
IP-DECT: Call transfer with enbloc dailing
| Status | Closed |
| Id | 93185 |
Call transfer with enbloc dailing fails. This is fixed now.
This changes also the R-key handling: after dialling a digit for a consultation call the call must disconnect with R-1 like in ring-back state.
SNMP Walk udpTable, tcpConnTable Could fail
| Status | Closed |
| Id | 93225 |
The table index returned wasn't always lexically ascending, causing a walk to stop.
phone: ip222,ip232: sporadic boot time trap when a Jabra LINK 280 adapter is plugged
| Status | Closed |
| Id | 93261 |
the initialisation fails with CC=5 on first device descriptor read. after restart of host controller serial_irq() traps in reading the done list.
Happens mostly with upload DRAM.
| Status | Closed |
| Id | 93272 |
Outgoing calls with beginning number *5 or *7 are blocked by the feature codes module because of hidden new service codes for an OEM device (#79028). This is fixed now.
PBX: Master Slave license update period 10s instead of 10min
| Status | Closed |
| Id | 93330 |
For test purposes the period was reduced to 10s, but by accident this change ended up in version 9 as well
Gateway: CGPN-Maps executed even if the Route did not match in case of enbloc calls
| Status | Closed |
| Id | 93400 |
For enbloc calls the CGPN of routes were executed even if the dialed number was incomplete.
PBX: Port License counting wrong, when moving users
| Status | Closed |
| Id | 93477 |
When moving a user from a master to a slave, the license count on the master was reduced only after reboot.
PBX: XML error on User Interface if slave name with non-ascii characters used and registration was redirected from such a slave
| Status | Closed |
| Id | 93483 |
Conversion of UCS2 as received from the registration to UTF-8 on Web UI was wrong
PBX: Hide connected Endpoint did nor work for forward to other PBX or Gateway object
| Status | Closed |
| Id | 93513 |
Diverting Leg Information facilities or Name Identification facilities were still forwarded in this case.
LDAP Client: SearchRequest.derefAliases Changed To neverDerefAliases(0)
| Status | Closed |
| Id | 94812 |
Was derefInSearching(1)
Gateway: Header line of FAX documents with big endian devices
| Status | Closed |
| Id | 94824 |
The header line of FAX documents with big endian devices was not correctly printed. This is fixed now.
SIP: Port mapping with STUN failed since DNS resolvin of STUN server failed
| Status | Closed |
| Id | 94876 |
Port mapping with STUN failed since DNS resolvin of STUN server failed.
Wrong STUN server port was used (5060 instead of 3478).
H.323: No media after blind transfer to Waiting Queue on other PBX, when operator connects
| Status | Closed |
| Id | 94961 |
Happened when different coders where used on caller and called side.
SIP: Display names need escaping of " and \\ according to RFC-3261
| Status | Closed |
| Id | 94966 |
Acc. to RFC-3261 characters " and \\ (%x22 and %x5C) are to be escaped as "quoted-pair".
PBX Waiting: Presence set for operator was not cleared, on delete or editing of Waiting Queue object
| Status | Closed |
| Id | 95016 |
When configuration of the Waitinng object is now changed, any presence set by the Waiting object is cleared.
Gateway: FAX interface interop with non-conforming Fax devices improved
| Status | Closed |
| Id | 95029 |
There are Fax devices sending wrong (too long) message initially after being called. In this case it is best handled by ignoring these message and wait for the retry instead of disconnecting the call.
SIP/UDP: Sending response to wrong address and port
| Status | Closed |
| Id | 95065 |
Sending response to wrong address and port.
But only if Via header of incoming request contains domain name.
IP241: Headset receiver muffled sound
| Status | Closed |
| Id | 95164 |
..
PBX Exec: Secretary availability monitoring did not work with multiple scretaries, with names starting identically
| Status | Closed |
| Id | 95197 |
The availability state (secretary booked into the exec primary group) was not associated with the correct secretary. A compare of the names only covered the first half of the name.
Click sounds at caller side when calling another port of same gateway
| Status | Closed |
| Id | 95436 |
occured since V9hotfix5
PBX Broadcast: Memory leak when calling busy broadcast object with round-robin config
| Status | Closed |
| Id | 95439 |
If all destinations of a broadcast object are busy a name-id facility generates a memory leak.
IP222 IP232 IP241: Use fifo for DSP control channel
| Status | Closed |
| Id | 95763 |
try to fix a trap with USB headset
IP232,IP222,IP241: Fix for call status display
| Status | Closed |
| Id | 95769 |
If a call was remotely disconneced during HOLD,
the held phone was constantly displaying "held" instead of "disconnected".
SIP: Interworking issue with "LifeSize Passport/LS_PP1_4.11.9 (8)"
| Status | Closed |
| Id | 95785 |
Problems decoding large and complex SDP offer from LifeSize.
802.1x (EAPOL) did not work on interfaces configured for VLAN
| Status | Closed |
| Id | 95931 |
802.1x (EAPOL) frames received without a VLAN tag must always be passed to the protocol module, even if the interface is confiured to use VLAN
PBX: Allow Name (instead of Long Name) to identify user for mobility data call thru
| Status | Closed |
| Id | 91660 |
This should simplify Opticaller configuration
IP-DECT: Static ports between Master and Radio
| Status | Closed |
| Id | 91815 |
Now the VOIP connections between the Master and the Radio use static ports instead of dynamic ones. This is useful if only a few ports should be opened through a firewall. For calls from the Radio to the Master the ports 1716 and 1717 (TLS) are used. For the default Master connection for calls from the Master to the Radio the ports 1718 and 1719 (TLS) are used. For dynamic Radio-Master connections the ports from 1722 are used. Every connection needs two ports.
IP-DECT: Cipher key index request for security test devices
| Status | Closed |
| Id | 92223 |
Cipher key index request procedure is changed to pass the test with security test devices. The cipher key index is used for DECT "Early Encryption"(EE).
Voicemail: <pbx-getcallinfo out-calling-name="..."/>
| Status | Closed |
| Id | 92286 |
Pass H.450 callingName into the script
PHONE_SIG_MODE_KEEP_NUMBER_TYPE can be set at phonesig startup to keep type of number in all q931 numbers
| Status | Closed |
| Id | 92385 |
PBX Trunk: Flag to block presence/dialog-info subscriptions
| Status | Closed |
| Id | 92824 |
Some networks e.g. sip carriers behave badly when receiving subscribes for presence/dialog-info, which cannot be handled, so there is an option added to block these.
SIP: Debug information for problems with STUN
| Status | Closed |
| Id | 93233 |
Added debug output to trace problems with STUN.
Gateway: FAX interface User-User-Info error response in disconnect event
| Status | Closed |
| Id | 93455 |
User-User-Info response of the FAX interface is not forwarded in the alerting state. The problem exists when call to Fax interface was routed through multiple PBX. Now the UUI response is sent in the disconnect event if the response is a error notification.
IP222 IP232 IP241: DSP code update
| Status | Closed |
| Id | 94916 |
DSP code update to version 680
SIP: New interop tweak "No Remote Hold Signaling"
| Status | Closed |
| Id | 94954 |
New config option on gateway interfaces "No Remote Hold Signaling".
Disables interworking of "inactive" into remoteHold.
IP6010: DSP trace options improved
| Status | Closed |
| Id | 95078 |
T38 trace flag worked only in one direction.
Changing trace options needed a reboot.
IPv6: Disable checkmark on ethernet interfaces added
| Status | Closed |
| Id | 95452 |
Allows to disable sending/receiving of IPv6 packets on this interface. This may be desireable for security reasons
IP222 IP232 IP241: DSP code update
| Status | Closed |
| Id | 95764 |
Channel was muted during silent periods ( no CNG during NLP operation ).
V9 Hotfix 22 (9061240)
Changes included in Version 9 hotfix22 Definition
PBX: URI dialing, should not be case sensitive and numbers should be possible
| Status | Closed |
| Id | 89326 |
needed for federation
IP222 IP232 IP241: LCD display is after softreset sometimes out of sync
| Status | Closed |
| Id | 95866 |
Displaycontroller needs to be stopped before restart.
Httpclient: Problems with HTTPS URLs
| Status | Closed |
| Id | 96099 |
Shortcut to local file I/O did not work fot HTTPS URLs.
phone: do not report "No Media Data received" errors for connections to a recording device
| Status | Closed |
| Id | 96102 |
some recoding devices, for example ASC never send data on a recording connection
IP-DECT: Possible no media in case of media renegotiation after handover
| Status | Closed |
| Id | 96124 |
The problem happened in about 50% of the cases of a media renegotiation which results in a different coder after handover.
IP-DECT: Show release state in IP1202
| Status | Closed |
| Id | 96157 |
The release state is not shown in the IP1202. This is fixed now.
SIP: Must not answer "refresher=uac" if request contains "refresher=uas"
| Status | Closed |
| Id | 96167 |
Interop issue with Genband C20 PBX.
Must not answer
Session-Expires: 400;refresher=uac
in 200/OK, if INVITE contains
Session-Expires: 400;refresher=uas
PBX Trunk: "Outgoing Calls restricted" did not work correctly, Presentation restricted was set, but number could be wrong
| Status | Closed |
| Id | 96200 |
For example if an analog Gateway was registered to a PBX user, and this Gateway did not send a Calling Party Number with the call, the call was sent with Presentation restricted, but without digits. This could affect Billing Applications which are based on CDRs from the Gateway.
phone: ip222,ip232: inbound calls automatically connected to Plantronics Savi W440/740/745 headsets with new firmware Versions
| Status | Closed |
| Id | 96276 |
reported for:
- Savi W440 with firmware 0118 on USB/DECT Dongle D100
- Savi W740/745 with firmware 0115
reason:
the newer firmware versions reject truncated output reports (no trailing 0 bytes) with STALL. The error handling for this case was wrong and caused an autoconnect.
IP-DECT: Wrong GK id of standby Master to Mobility Master
| Status | Closed |
| Id | 96302 |
The standby Master uses a wrong gatekeeper id to register to the Mobility Master. This is fixed now.
PBX-CDRs: Conference ID missing in CDRs created by Mobility data callback/callthru
| Status | Closed |
| Id | 96399 |
This caused these calls not to show up in the reporting
PBX: Unexpected behaviour if too many filter were configured
| Status | Closed |
| Id | 96416 |
No new filters were accepted without error message
IP22 IP24 IP28 IP302 IP305: Sporadic DSP host interface overruns
| Status | Closed |
| Id | 96606 |
On the small gateways the DSP hangs if control packet on host interface arrive too fast. Now the rate is limited.
License download not working
| Status | Closed |
| Id | 96622 |
License download stops.
Licenses are not downloaded.
PBX: Changing of PBX Object Name did not change the name to be used for registration
| Status | Closed |
| Id | 96634 |
So if the name of the PBX registering as slave was changed as well, it did not register anymore. The PBX object had to be deleted and created with new name.
Gateway: FAX interface on IPVA
| Status | Closed |
| Id | 96660 |
The FAX interface on IPVA can not connect to a remote device because of wrong protocol events. This is fixed now.
SRTP: One way audio after some minutes on IP6000 IP2000 IP6010 IP0010
| Status | Closed |
| Id | 96673 |
Applies to v9hotfix21 on IP6000, IP2000, IP6010, and IP0010.
Collateral damage from fix #91892: SRTP: Avoid one-way media with high start sequence numbers.
Gateway: CGPN-Maps executed even if the Routing was already completed
| Status | Closed |
| Id | 96685 |
A CGPN map, which was configured in a route following the route, which was actually executed was executed as well.
This was a collateral damage from fix: #93400: Gateway: CGPN-Maps executed even if the Route did not match in case of enbloc calls
IP22 IP24 IP28 IP302 IP305: Sporadic DSP host interface overruns - CLIP disabled, new trace option
| Status | Closed |
| Id | 96716 |
On the small gateways the DSP hangs in some conditions.
Now the trace-stop is replaced with an Assert to recover from this situation.
To get a trace of this condition a new trace option is added at dsp.xsl, called txt-trace. This traces the DSP message as text, so that they can be read out after a trap.
Typical usage is to enable DSP-trace, DSP control messages DSP data messages and DSP txt trace.
DSP pcm trace and DSP T38 trace should be off to avoid excessive debug load.
Also, the CLIP messages are disabled since they caused problems in the past.
H.323: One-way-voice if SRTP call to a Waiting queue is forwarded via Waiting Queue Maps to a phone
| Status | Closed |
| Id | 96721 |
This is a problem with SRTP key exchange, which could happen in other traffic scenarios as well.
IP-DECT: Web UI administrator user list removed on IP1202
| Status | Closed |
| Id | 96726 |
The Web UI administrator user list is removed on the IP1202 now.
IP28: Click sounds at caller side when calling ip28 gateway
| Status | Closed |
| Id | 96782 |
IP28: Sometime Ringing stopped working on an anlog port. Worked again only after reset.
| Status | Closed |
| Id | 96787 |
This happened due to low ringing volatge, the default value of "low" on the interface configuration for the ringing voltage caused this to happen.
IP28: Ring voltage failure (RING_FAIL)
| Status | Closed |
| Id | 96826 |
IP22 IP24 IP28 IP302 IP305: Sporadic DSP host interface overruns - Updated DSP code
| Status | Closed |
| Id | 96852 |
Clip enabled on all channel.
phone: ip222,ip232,ip241: Local Network Coder default for User-2..6 was G711 instead of G722 as for User-1
| Status | Closed |
| Id | 96890 |
WEB-Interface "Phone/User-2..6/General/Options/Local Network Coder" was preset to G711 instead to G722 as for User-1.
PBX: New option for RTP Proxy - proxy only if different registration address
| Status | Closed |
| Id | 88439 |
To avaoid RTP Proxy for two endpoints located within same private network behind NAT
PBX Executive: Allow monitoring of availability of secondary secretary, don't treat Exec as secretary
| Status | Closed |
| Id | 95497 |
With these two additions a configuration with two executives and two secretaries, each secretary being primary to one executive an secondary to other can be configured with a single group for each secretary and both executives can monitor the availability of both secretaries.
SIP: Workaround for buggy registrar
| Status | Closed |
| Id | 96313 |
Workaround for buggy registrar.
Different expirtes values in Contact header and Expires header.
\tSIP/2.0 200 OK
\tVia: SIP/2.0/UDP x.x.x.x:2069;rport=2069;branch=z9hG4bK-CDDD130C
\tTo: ;tag=5b8729d5-6f6353c4-cbsxz
\tFrom: <sip:38795988@193.90.37.3>;tag=2109370043;epid=0090331e0bef
\tCall-ID: 7ef63c56e909d311b3890090331e0bef@95.130.221.205
\tCSeq: 1004 REGISTER
\tExpires: 60
\tContact: <sip:38795988@x.x.x.x:2069;transport=UDP>;expires=300
\tUser-Agent: ZTE-SBC
\tX-ZTE-Cause: "SBC-4721-2002"
\tContent-Length: 0
Better apply the smaller expires value.
PBX: Send forking calls as diverted calls
| Status | Closed |
| Id | 96370 |
So that on the called side, it will be displayed who forked the call the same way as a diverted call is displayed.
PBX: pbx_makecall.txt not only for mobility, but for Waiting Queue also
| Status | Closed |
| Id | 96384 |
Allows to initiate call with simple web request from a Waiting Queue
IP-DECT: Physical location was wrong after logout/login on handset
| Status | Closed |
| Id | 96392 |
The physical location information is based on the redirection of the registration from the PBX at the physical location to the registration PBX. Some information was not cleared with the logout, so re-registration startet with the registration PBX right away.
phone: if a number to be dialled contains a comma, the digits following the comma are sent as DTMF tones after connect
| Status | Closed |
| Id | 96402 |
This applies to all numbers dialed en bloc, i.e. numbers dialed via indirect dialing, a phone directory or a function key. The comma must not be the first character of the number.
IP-DECT: OEM PBX type info in GUI data
| Status | Closed |
| Id | 96723 |
Now the GUI data includes the type info of an OEM PBX.
IP22 IP24 Ip28 IP302: Don't complete media negotiation for ab-interfaces if no media can be sent
| Status | Closed |
| Id | 96773 |
For incoming calls to a phone media negotiation was already completed during ringing, so that when going off hook the media channel was already established. This causes interop problems, because there are endpoints which asssume there is inband info (e.g. ringback) if media negotiation is complete so local tones (e.g. ringback) were turned off.
In the past with slowstart this premature media negotiation was usefull to avoid delayed media after off-hook. With SIP or H.323 faststart there is no use anymore.
Needed to avoid that the DSP send CLIP and tones at the same time, which can cause sporadic DSP failures.
SIP: New interop tweak /register-interval
| Status | Closed |
| Id | 97834 |
New config file option /register-interval 60
Problem is too weired to explain.
This option can be used to set the REGISTER interval to a fixed value regardless of the negotiation.
V9 Hotfix 23 (9061252)
Changes included in Version 9 hotfix23 Definition
SUBSCRIBE for MWI not correctly handled after change of IP address
| Status | Closed |
| Id | 96898 |
SUBSCRIBE for MWI not correctly handled after change of IP address
phone: ip222,ip232: USB headset echo effects in call setup phase when a G722 call is started by a CTI application
| Status | Closed |
| Id | 96974 |
The caller hears the calling tones and it's own speech from the headset microphone but the callers speech is not transmitted to the called party.
H.323: Offered packetization should be honored for SIP interoperability
| Status | Closed |
| Id | 96983 |
Problem happend with calls from Samwin CBC
RTP-DTMF: Must increase duration field when sending RTP-Event with END marker
| Status | Closed |
| Id | 97001 |
Must increase duration field when sending RTP-Event with END marker to comply with RFC.
ASN.1 BER: Decoding of Sequence Member with indefinite length failed
| Status | Closed |
| Id | 97072 |
An Avaya supplementary service couldn't be decoded
H.323: Channel Close sometimes not sent on hold
| Status | Closed |
| Id | 97080 |
Esspecially on the second hold within a call the Channel Close was not sent to the party, which put the other on hold. This caused the channel not beeing turned off on this side (the other side receives music on hold in this case)
IP4 did not work anymore when IP6 was disabled via WEB interface
| Status | Closed |
| Id | 97111 |
When the "IP6/ETNx/IP6/Options/Disabled" was checked the IP4 operations did stop after a while.
phone: ip222,ip232: Plantronics Savi W440 dosn't report Talk-Key events in a call established at phone or by a CTI application
| Status | Closed |
| Id | 97115 |
When a call via this headset was initiated/accepted by the Redial-Key, the Headset(Mode:Control) function key or a CTI application, the call could not be disconnected by pressing the Talk-Key at the headset because the Headset did not report this action.
phone: DTMF digits following a comma in a number to be dialed were not handled correctly in some cases
| Status | Closed |
| Id | 97150 |
- in the "Destination Number" configured under "Phone/Direct Dialing" in conjunction with a nonzero "Autodial Timeout": the DTMF digits were sent as dial digits
- with a nonzereo "Enblock Dialing Timeout" configured under "Phone/User x/General/Options": sending of DTMF digits was delayed by the configured timeout\t
Gateway: FAX interface on IP800/IP305/IP302
| Status | Closed |
| Id | 97571 |
Fix for the last fix #96660.
The FAX interface on the IP800/IP305/IP302 can not connect to a remote device because of wrong protocol events. This is fixed now.
H.323: Fast Unregister/Register operations could lead to failed registrations, in case of fixed signaling ports
| Status | Closed |
| Id | 97637 |
A listening socket could still be in use. Only happened with IP-DECT and multi-master.
H.323: Trap when Name-Id of more the 128 characters is to be forwarded
| Status | Closed |
| Id | 97639 |
buffer overrun happened
PBX: Name-Id of busy destination was not forwarded to other PBX
| Status | Closed |
| Id | 97646 |
Display was different when calling a busy phone on local PBX or on another PBX
IP22 IP24 IP28 IP302 IP305: ASSERT on DSP queue overrun added
| Status | Closed |
| Id | 97653 |
phone: "Prepare Override" function key did not work since V9hotfix21, the overriding source address was ignored
| Status | Closed |
| Id | 97665 |
SIP: Trap when terminating a call while re-negotiation is ongoing
| Status | Closed |
| Id | 97675 |
Trap when terminating a call while re-INVITE is pending.
phone: ip230,ip240,ip241 : unefined codes received from a DHSG Headset basestation were misinterpreted as Hookswitch indication
| Status | Closed |
| Id | 97732 |
sometimes DHSG Headset basestations send codes not defined for DHSG which are silently discarded now.
H.323: Potential Trap when reconfiguring an H.323 registration
| Status | Closed |
| Id | 97820 |
This is a collateral damage from
91815: IP-DECT: Static ports between Master and Radio
PBX Executive: Calls with calling id restriction and without calling id, were sent to executive, even if secretary available
| Status | Closed |
| Id | 97897 |
This happend for calls coming in from public ISDN with calling id presentation restriction thru a trunk object without number
SIP: Trap when using STUN
| Status | Closed |
| Id | 97898 |
Trap when using STUN.
SIP: Dialog-Info was encoded with wrong state attribute
| Status | Closed |
| Id | 97926 |
Dialog-Info was encoded with wrong state attribute "full".
Must be "partial".
SIP: Logging was wrong
| Status | Closed |
| Id | 98037 |
Logging was wrong.
PBX: Potential trap when receiving unknown presence activity
| Status | Closed |
| Id | 98043 |
In the respective version unknown activities are mapped to "busy"
PBX Waiting: timeout argument for pbx_makecall.txt URL
| Status | Closed |
| Id | 97010 |
A timeout argument was added to the PBX0/ADMIN/pbx_makecall.txt URL to cancel an outgoing call initiated with this URL.
ISDN: Send legacy Redirecting Number for 'old' Fax Servers
| Status | Closed |
| Id | 97615 |
The redirecting number is an old style information element, which contains part of the information as the diverting leg2 facility. Some Fax Servers do not understand the leg2 facility.
PBX-CDRs: Better CDR for pickup
| Status | Closed |
| Id | 97681 |
A pickup was not indicated in the CDRs
PBX-SOAP: Status of Boolean object indicated as local number
| Status | Closed |
| Id | 97738 |
When monitoring a Boolean object with SOAP a call is indicated. The local number of this call is set based on the status of the boolean object (00 automatic-off, 01, automatic-on, 10 - manual-off, 11 - manual-on)
PBX-SOAP: UserPark allows to park to another object
| Status | Closed |
| Id | 97741 |
The argument 'cn' was ignored in v9 and earlier, now it can be used to identify a destination for the park.
V9 Hotfix 24 (9061271)
Changes included in Version 9 hotfix24 Definition
IPVA: Unused ETH1 Could Cause Out-Of-Memory Situation
| Status | Closed |
| Id | 98179 |
Outgoing packets could queue up without ever getting purged.
SIP: Trap - not checking array boiundaries
| Status | Closed |
| Id | 98219 |
Trap - not checking array boiundaries when processing User-Agent header.
SIP: STUN not working
| Status | Closed |
| Id | 98223 |
STUN not working if STUN server IP address is configured.
No problem if if STUN domain name is configured.
phone: ip222,ip232: USB connection sometimes lost until reboot
| Status | Closed |
| Id | 98290 |
Observerved with with headsets from various manufacturers. Most probably caused by strong electrostatic discharges to the USB connection cable. In such case also unplugging/plugging of the headset was not noticed by the driver anymore.
Logging: "Alarm and Event Forward Server" address could not be changed anymore once configured
| Status | Closed |
| Id | 98457 |
LDAP Replication: ASN.1 Decoding Failed For More Than 100 Attributes Per Object
| Status | Closed |
| Id | 98468 |
Happend with 100 attributes in a PBX stored phone configuration.
Corrupt trap buffer when tracing UART messages
| Status | Closed |
| Id | 98533 |
IP232,IP222: Confirming blind transfer with redial key did not work
| Status | Closed |
| Id | 98591 |
Short user guide:
Press 'redial' before accepting the call, enter telephone
number for the call diversion and confirm by pressing 'redial' again.
phone: ip222,ip232: USB headset media connection lost after a release received from a remote conference peer
| Status | Closed |
| Id | 98600 |
Happened only on a release of the call which was the active call when the conference was established. The remaining VOIP connection was OK but the media stream was not passed from/to headset anymore.
IP241: DHSG Headset messages are sometimes wrong
| Status | Closed |
| Id | 98642 |
Beim Neuladen des Sequencers bei nderung der LCD-Helligkeit kommt der UART-Takt gelegentlich zu schnell. Besser noch wre 2 sequencerprogramme zu definiere, aber wie das geht ist nicht im Orchid/Titan Usermanual nicht offensichtlich -->
SIP: SDP version not increased when answering an offer where only media-mode has changed
| Status | Closed |
| Id | 98739 |
If remote side changes from 'sendrecv' to 'inactive'
the SDP answer follows this change of media-mode,
but SDP version was not increased.
Gatway: Configuration of Blockdial Timeout at Routes did not work
| Status | Closed |
| Id | 98748 |
Wrong value was calculated, if multiple maps were used in a single route blockdial timeout configuration of a map was lost, when another map was configured.
enabled state of an external directory configured via a PBX config template was lost in some cases,
| Status | Closed |
| Id | 98816 |
This did happen for example when
- a second phone was registered to the same PBX user (twin phone)
- a "Phone/Reset/Reset User Specific Configuration" was done via the phones WEB GUI (but not when this was done via the PBX GUI)
SIP: Trap when sending <dialog-info>
| Status | Closed |
| Id | 98902 |
Trap when interworking group-indications into dialog-info.
Memory Leak when deleting voicemails
| Status | Closed |
| Id | 98929 |
A list wasn't cleaned
IP222 IP232: Noise in the microphone at 6400Hz
| Status | Closed |
| Id | 98941 |
.
IP222 IP232 IP241: Codec register debugs added
| Status | Closed |
| Id | 98972 |
Codec register debugs are enabled with dsp trace.
Codec register are dumped after changes to analyse Manits 97903
PBX: No config updates were sent to the phones, when selection of templates was changed
| Status | Closed |
| Id | 98983 |
When the config of the template itself was changed an update was sent, but if it was changed which templates were used on a user object, no update was sent.
Phone: Could not configure fkey labels containing single quotation mark
| Status | Closed |
| Id | 98986 |
Could not configure fkey labels containing single quotation mark.
myPBX: Missed calls for mobility calls accepted somewhere else
| Status | Closed |
| Id | 98995 |
A cause code was missing in CDRs generated by mobility
PBX Mobility: No DTMF R-Key Features possible after data callback
| Status | Closed |
| Id | 99020 |
For a mobility call, which was established with data callback no DTMF R-Key Features (e.g. put the call on hold) were possible.
SIP: Switch-over to t38 did not work in one configuration scenario
| Status | Closed |
| Id | 99130 |
Interworking of SIP and H.323.
Switch-over to t38 did not work if both Gateway interfaces were configured to media-relay with exclusive audio codec.
phone: ip222,ip232: ignore HID function of USB headset charging cables
| Status | Closed |
| Id | 99133 |
Some USB headset charging cables present a HID function as long as the headset is connected to the cable (probably used for headset firmware updates).
Dependent on the enumeration sequence the HID function of the cable could hide the HID function of the headset when the headset base station (or the bluetooth or DECT dongle) is plugged in paralll to the phone.
IP22 IP24 IP28 IP302 IP305: ASSERT on DSP queue overrun added (2)
| Status | Closed |
| Id | 99155 |
H.323: Alternate Registration to IP address did not work, if primary used Discovery
| Status | Closed |
| Id | 99228 |
If the registration to the primary destination, which used discovery failed, the alternate registration to IP address was sent to the discovery port (1718) instead of the registration port (1719).
phone: ip222,ip232: support USB headsets with two audio input channels from microphone
| Status | Closed |
| Id | 99290 |
First seen with a Plantronics Blackwire C320 (one of the cheapest wired models)
PBX Mobility: Trap if calling from a slave to an object at the master, with mobility destination on the same slave
| Status | Closed |
| Id | 99400 |
Null pointer access happend
IP22 IP24 IP28 IP302 IP305: Sporadic DSP host interface overruns - Tonegeneration fixed
| Status | Closed |
| Id | 99403 |
Simultaneous tone generation and caller id generation caused DSP problem on analog gateways.
Tone generation with undefined coder disabled to avoid this problem.
phone: ip222,ip232: "Phone/Preferences/Use Handset like a Headset" mode did not work
| Status | Closed |
| Id | 99414 |
on an ip2x2 only the headset function key (Mode:Control) can be used to to start an outbound call or to accept an inbound call in this mode but the headset function key was ignored when no headset was plugged/enabled.
SIP: Handling of 180 with SDP answer is required after 180 without SDP
| Status | Closed |
| Id | 99428 |
A 180 with SDP answer must be processed after 180 without SDP has already been received.
IP232,IP222,IP241: Rendering error on fkeys during hotdesking
| Status | Closed |
| Id | 99446 |
Bad pixels right of fkey during hotdesking.
But only at fkeys with icon.
SIP: Must re-create message-summary subscription after re-connecting to server after local address change
| Status | Closed |
| Id | 99488 |
Must re-create message-summary subscription after re-connecting to server after local address change.
Must not re-use call-id and tags for re-subscription.
phone: ip222,ip232: prevent unintentional autoconnect of an inbound call arriving while the headset radio link runs down
| Status | Closed |
| Id | 99497 |
This happens when another call was just released and the new call arrives while the phone is in idle state but the headset base is running down the radio link.
Phone: NOTIFY(sipfrag) was missing after transfer complete
| Status | Closed |
| Id | 99535 |
When call replacement (REFER) is completed a NOTIFY(sipfrag:200/OK) must be sent to the sender of REFER (transfering party).
Phones: Allow changes to language, ringtones and fkeys if "Allow User Settings at Phone" is activated
| Status | Closed |
| Id | 99548 |
Allow changes to language, ringtones and fkeys if "Allow User Settings at Phone" is activated.
PBX: Trap with operator searches and objects with 'Hide from LDAP' and multi-level nodes
| Status | Closed |
| Id | 99557 |
An endless loop happened, which caused the PBX to restart.
PBX Mobility: Call Waiting Facility was missing for waiting calls
| Status | Closed |
| Id | 99572 |
A caller could not see that his call was waiting
PBX: No Registration redirect back to original PBX
| Status | Closed |
| Id | 99579 |
This could happen with a config error or during startup when not all objects are read.
H.323: Fallback to Slowstart after CFNR did not work
| Status | Closed |
| Id | 99600 |
If a CFNR to a Slowstart endpoint (e.g. XCAPI) was performed after a call to an EFC endpoint, the fallback to slowstart did not work in a szenario with multiple PBXs.
phone: ip222,ip232: delay ringing to USB headset when a previous call was released immediately before
| Status | Closed |
| Id | 99616 |
Some USB headsets (even wired ones) need a surprisingly long time to disconnect (up to 500 millisecons). To play the ring tone the headset must be connected again and this may fail before the disconnect is completed.
The default delay is one second from start of last disconnect, it can be set by
config add PHONE APP /usb-calm <ticks>
where <ticks> means 20 ms timer ticks.
phone: ip222,ip232: minimize delays in audio stream connect/disconnect operations
| Status | Closed |
| Id | 99625 |
PBX: CFNR/CFB on PBX object did not work in some cases
| Status | Closed |
| Id | 99674 |
The number appended to the call when the forwarding was executed was not correct sometimes
phone: ip222,ip232,ip241: send RTP data to network after hold/retrieve even if remote party does not send
| Status | Closed |
| Id | 99682 |
IP-DECT: Default boot code file name of the IP1202
| Status | Closed |
| Id | 99688 |
Now the default boot code file name used by the update script is correct.
phone: ip222,ip232: Plantronics DA45 with new firmware version (0090) did not work
| Status | Closed |
| Id | 99724 |
The new firmware rejects commands with STALL which were accepted by the older firmware (the commands were sent with trailing zeros which were silently ignored). This may also apply to Blackwire C420 / C435 / C620 which use the same firmware.
SIP: Display name of "original called party" was missing
| Status | Closed |
| Id | 99739 |
Display name of "original called party" (first diverting party) was missing in case there were multiple diversions.
When processing INVITE with (multiple) History-Info header.
phone: ip222, ip232: support Jabra UC Voice 750 and Jabra Speak 510
| Status | Closed |
| Id | 98092 |
IPVA: ETHx Transmit Queue Size Limited To 1MB
| Status | Closed |
| Id | 98187 |
Was unlimited
Linux: Shutdown warning message
| Status | Closed |
| Id | 98438 |
If Linux is running, a shutdown warning message is shown at the Linux General page now.
SIP/TLS: Using domain name as fall back to proxy name when comparing to the certificat "subject"
| Status | Closed |
| Id | 98643 |
Using domain name as fall back to proxy name when comparing to the certificat "subject".
Voicemail: Allow SMTP Email Port Different Than tcp/25
| Status | Closed |
| Id | 98683 |
eg in email.xml: ..smtp.foo.bar:888..
phone: directory search highlights first matching entry if the search expression contains any non numeric digit
| Status | Closed |
| Id | 98806 |
To prevent unintended dialing of a directory entry starting with numeric digits the search expression was checked if it consists of dialable digits (0-9*#,) only.
In this case the first matching entry was not automatically highlighted (activated) so that the input (number) could be dialled by going off-hook.
Now the check includes the numeric digits (0-9) only.
SIP: New interop tweak /accept_any_reg_interval
| Status | Closed |
| Id | 98887 |
For endpoints that do not read the expires value from 200/OK response to REGISTER.
E.g. "User-Agent: TRBOnet.Enterprise"
phone: ip222,ip232: support Plantronics Blackwire C320 / C520 / C720, Voyager Legend UC, Calisto 620
| Status | Closed |
| Id | 99112 |
Gateway: Only transparent (clearmode) coder in offer if data call
| Status | Closed |
| Id | 99234 |
This is a SIP interop issue. Some equipment cannot ignore clearmode coder offers, if not supported.
SIP: New interop tweak /c-line-at-session-level
| Status | Closed |
| Id | 99237 |
New interop tweak /c-line-at-session-level for clients that do not read RTP address from media description.
For clients not compliant to RFC-4566.
phone: a park function key with both 'Number' and 'Name' left empty implies to use 'Number' or 'Name' of the Registration
| Status | Closed |
| Id | 99387 |
The key works the same way as a key with an explicitely configured 'Number' or 'Name'.
phone: volume of pickup notification tone is set according to the volume configured for internal ring tone
| Status | Closed |
| Id | 99424 |
If the user prefers a different setting the automatically derived volume can be overridden with a fixed volume as before via
config add PHONE SIG /notify-pickup-gain <gain>
( see http://wiki.innovaphone.com/index.php?title=Howto:Change_the_volume_of_the_pickup_key_audio_notification )
phone: flag to disable speaker key to prevent conversations in handsfree mode
| Status | Closed |
| Id | 99555 |
set via
config add PHONE APP /no-speaker-key
PBX Mobility: Dial thru
| Status | Closed |
| Id | 99780 |
When calling the Mobility object from the mobile phone, additional dialed digits are used to call the destination. This is an alternative to using DTMF for dialing. How many digits may be dialed depends on what the network of the mobile phone supports
V9 Hotfix 25 (9061282)
Changes included in Version 9 hotfix25 Definition
PBX: Admin rights restriction was not displayed after Apply anymore
| Status | Closed |
| Id | 99987 |
If an admin had restrictions for PBX oder Node, this retrcition was not shown after pressing Apply when editing an object.
SIP: Problems on media negotiation
| Status | Closed |
| Id | 100107 |
Problems on media negotiation when CFNR is executed by PBX
on an incoming SIP call without SDP offer.
IP22 IP24 IP28 IP302 IP305: Sporadic DSP host interface overruns - Updated DSP code
| Status | Closed |
| Id | 100128 |
DSP code 680.12.pf.01. Assert if DSP is not responding added
PBX: Case insensitive search for PBX objects did not work for non-ascii characters
| Status | Closed |
| Id | 100195 |
No result was found when searching with lower case and Name contained upper case
PBX: When counting calls to slave PBXs, subscriptions were counted as well
| Status | Closed |
| Id | 100203 |
Only calls with media should be counted
AD Replication: Mapping yielding empty CN caused 100% cpu load
| Status | Closed |
| Id | 100209 |
A detection for data differences was by-passed.
SIP: Don't use uri scheme "sips"
| Status | Closed |
| Id | 100210 |
Don't use uri scheme "sips" even when using TLS as transport.
"sips" is not accepted by MS LYNC.
Media: NAT workaround was activated during transfer in some cases
| Status | Closed |
| Id | 100338 |
NAT workaround was activated during transfer in some cases.
Late packets from transferring endpoint may trigger the NAT workaround at the 2 transferred endpoints, making the transferred endpoints redirecting to the obsolete RTP addr/port of the transferring endpoint.
Replication: Distribution Of Trace Flag To Existing Connections
| Status | Closed |
| Id | 100383 |
Prior to this fix a reboot was necessary to see all trace-output.
Phone: Distinctive ringing of emergency calls
| Status | Closed |
| Id | 100490 |
Distinctive ringing of emergency calls.
I6000 IP2000 Allow changing SRTP key while data is queued for encryption
| Status | Closed |
| Id | 100549 |
Bug in the crypto crypto driver. When the SRTP key is changed while a packet is being encrypted the SRTP socket hung up.
IP-DECT: Call transfer timeout
| Status | Closed |
| Id | 100603 |
The call transfer timeout is changed to 25s for transfer to mobile targets. Used by OEM PBXs.
myPBX: Add node escape prefix to number from PBX LDAP directory
| Status | Closed |
| Id | 100698 |
Internal numbers might differ depending on the caller, due to node membership. So the internal number has to be adapted for the individual users.
SIP: No INVITE is sent if client has registered with a local domain name
| Status | Closed |
| Id | 100728 |
No INVITE is sent if client has registered with a local domain name instead of ip address.
E.g.
\tREGISTER sip:172.16.16.124 SIP/2.0
\tVia: SIP/2.0/UDP kws6000.noexist.local:5060;branch=z9hG4bK40ce4a47e899
\tFrom: ;tag=40b10bdf37c5
\tTo: <sip:sga@172.16.16.124>
\tCall-ID: 4040c07c656162185bf6d53451f1ecae
\tCSeq: 387119859 REGISTER
\tContact: <sip:sga@kws6000.noexist.local>
\tAuthorization: Digest username="sga", realm="172.16.16.124",...
\tAllow: OPTIONS, INVITE, ACK, CANCEL, BYE, SUBSCRIBE, NOTIFY,...
\tMax-Forwards: 70
\tUser-Agent: KIRK Wireless Server 6000 PCS13__ r40453
\tSupported: replaces
\tSupported: 100rel
\tExpires: 60
\tContent-Length: 0
H.323: No media after Hold/Retrieve on call with multiple media-relay
| Status | Closed |
| Id | 100823 |
This could be considered a mis-configuration, because a single media-relay should be good enough, but should work anyway.
LDAP/Expert: Search Page-Size Now 50
| Status | Closed |
| Id | 100847 |
was 100
NTP-Server: use destination address from client request as source address in response to client
| Status | Closed |
| Id | 100940 |
a response to a client request received via ETH1 was sent with the ETH0 address as source address when routed through default gateway on ETH0.
a response with a source address not matching the adressed server is discarded on client side.
Correctly distinct Quick Dial and Directory Search object
| Status | Closed |
| Id | 101020 |
Sometimes die Quick Dial object became a Directory Search object, if the submit has been rejected.
Escape Mechanism for Flash Directory Objects
| Status | Closed |
| Id | 99578 |
The mechanism's motivation is to avoid unnecessary binary encodings and to spare some length per configuration line.
http://wiki.innovaphone.com/index.php?title=Concept_Flash_Directory#Escape_Rule_For_Object_Values
phone: support transparent recording of calls to/from other registrations than the active registration
| Status | Closed |
| Id | 100120 |
If a registration is configured for 'transparent' recording a call via this registration is always recorded now even if the registration is not the 'active' registration and no recording is configured for the 'active' registration.
SIP: Workaround for SIP client giving wrong ip address in Contact-URI
| Status | Closed |
| Id | 100822 |
Workaround for SIP client giving wrong ip address in Contact-URI.
3CXPhoneSystem gives ip address of PBX in Contact-URI.
PBX sends upcoming SIP requests to its own ip address.
V9 Hotfix 26 (9061288)
Changes included in Version 9 hotfix26 Definition
phone: ip110/150/230:- start completion of a prepared dial string in 'numeric' keyboard mode if last char of string is numeric
| Status | Closed |
| Id | 101082 |
Pressing a 'Dial' function key whith 'Prepare' checked permits further editing of the string configured under 'Number' or 'Name'. The keyboard mode was set to 'alpha' if the string contained any character not permitted in an e164 number. Now the mode is set to 'numeric' when the last character of the string is a decimal digit.
myPBX: Possible trap on configuring visibility and call diversions
| Status | Closed |
| Id | 101325 |
SIP: STUN server could not be removed from Gateway interface config without reboot
| Status | Closed |
| Id | 101384 |
STUN server could not be removed from Gateway interface config without reboot.
It disappeard from config, but was still used by SIP interface.
PBX: The checkmark "RTP Proxy - Except Addresses are identical or private" could not be cleared
| Status | Closed |
| Id | 101421 |
once set, this checkmark could not be unset
Gateway: Disabled routes were barely distinguishable from normal routes
| Status | Closed |
| Id | 101422 |
Light gray background helps
PBX: Routing from one slave to another with multiple nodes and overlap dialing could fail
| Status | Closed |
| Id | 101491 |
Only under more special conditions
Linux: Network configuration reset message
| Status | Closed |
| Id | 101492 |
The reset needed message is sometimes missed if the network configuration for Linux is changed. This is fixed now.
PBX: Blind Transfer to a different Slave could fail under special circumstances
| Status | Closed |
| Id | 101509 |
The blind transfer failed when
- e.164 style nodes are used
- transfer destination is thru a trunk on a different PBX
- transfering endpoint is configured to a node, which is another PBX
- transfered call is from trunk
- ...
IP-DECT: Radio call list with OEM PBX
| Status | Closed |
| Id | 101606 |
The radio call list can't be shown with Mozilla if an OEM PBX is used and a 'R'-key has been pressed. This is fixed now.
PBX: Boot-Loop when replicating objects with wrong password
| Status | Closed |
| Id | 101879 |
When a replication was configured, replicating all objects from a different PBX, with different PBX password then configured already on the box, the box entered a boot loop, which only could be stopped with a long reset.
phone: ip222,ip232: Unintended autoconnect to handset when headset USB port connection is lost while ringing
| Status | Closed |
| Id | 102028 |
If a Jabra PRO 930 has "Auto sleep Mode" enabled it enters sleep (low power) state when it was not used for 8 hours.
When a call arrives this is signaled to the headset. When the Jabra PRO 930 is in sleep state this works for about one second but then the USB port connection gets lost.
The handling of such presumably temporary failures was not correct in case of a ringing call.
IP6000: Prevent blinking error LED on old IP6000 with HW-Build 200
| Status | Closed |
| Id | 102151 |
Conference DSP driver was started on old hardware that doesnt support the conference DSP
PBX: ISDN Partial Reroute did not work with e164 configurations
| Status | Closed |
| Id | 102226 |
The destination of the reroute was wrong (digits missing) if the reroute was done on a trunk object not in the root node with escapes
IP222 IP232 IP241: Prevent simultaneous ec and pcm trace
| Status | Closed |
| Id | 102240 |
Simulatenous ec and pcm trace cause packet loss at the DSP host interface.
phone: Permit Call Intrusion in Silent Monitoring Mode via Recall-Menu if configured at Phone
| Status | Closed |
| Id | 100266 |
Preconditions:
- "Phone/User-x/Preferences/Enable Call Intrusion" checked
- "config add PHONE APP /recall-ci-monitor" performed
In the Recall-Menu opened when the Menu-key is pressed after dialling a busy user the "Monitor" option is offered in addition to the "Intrude" option.
SIP: New option "Filter incoming calls" on gateway interfaces
| Status | Closed |
| Id | 101622 |
New option "Filter incoming calls" on gateway interfaces.
IP222 IP232 IP241: DSP code update with improved Echocanceller
| Status | Closed |
| Id | 102061 |
New DSP code with improved EC
Gain settings changed for new DSP code.
V9 Hotfix 27 (9061294)
Changes included in Version 9 hotfix27 Definition
IP232,IP222: Wrong keyboard input
| Status | Closed |
| Id | 102369 |
Wrong keyboard input.
Key toggle runs even after key was released.
PBX: Restart when submitting an object (under very unlikely timing conditions)
| Status | Closed |
| Id | 102482 |
This only happend when during the submitting of the object the HTTP session was terminated extremly quickly.
SIP: Config option /take-zero-addr-for-hold did not work anymore
| Status | Closed |
| Id | 102984 |
Config option /take-zero-addr-for-hold did not work anymore.
Since v9hf16 (#85534: SIP: Interop with Genband SBC).
SIP: Memory leak when handling SUBSCRIBE(dialog-info)
| Status | Closed |
| Id | 102997 |
Memory leak when handling SUBSCRIBE(dialog-info),
but only if a endpoint sends overlapping SUBSCRIBE requests
to same destination with different Call-ID's.
Gateway: Trap on collision of call-complation termination collision
| Status | Closed |
| Id | 103125 |
A restart could happen if a call-completion monitoring was terminated at the same time from the network and from the user/pbx.
NAT: Do keepalive on TCP sessions
| Status | Closed |
| Id | 103133 |
Otherwise no cleanup of TCP sessions would happen if the remote endpoints are restarted.
myPBX: Add node escape prefix to number from PBX LDAP directory did not work
| Status | Closed |
| Id | 103344 |
Previous fix #100698 did not work.
ip24, phone_orchid, ip6010...: too much padding of short ethernet frames
| Status | Closed |
| Id | 103410 |
some firewalls complain if short ehternet frames are padded to more than 60 bytes which was mistakenly done in some ethernet drivers.
PBX: Master/Slave Max Calls take subscriptions into account
| Status | Closed |
| Id | 103415 |
Functionkeys with dialog or presence subscription accross PBXs could block calls.
PBX: Call Completion was executed on termination of multicast call
| Status | Closed |
| Id | 103628 |
The fact that the call was accepted was treated as a user action, but the accept was automatic, so no user has touched the phone
PBX-SOAP: Memory leak when terminating a SOAP session in an unusual way
| Status | Closed |
| Id | 103668 |
For example if just the network connection is lost while a Poll command was pending a cmd_exec object was leaking.
phone: ip222,ip232: call waiting not signaled in USB headset when "Call Waiting: beep once" was configured
| Status | Closed |
| Id | 103792 |
SIP: Handling of reject for UPDATE for session refresh was not correct
| Status | Closed |
| Id | 103996 |
Receiving a reject for UPDATE used to refresh a call (session refresh)
must be handled like receiving BYE.
IP-DECT: Trap with login feature
| Status | Closed |
| Id | 104609 |
There is a trap in DECT radio if the user login feature is used. This is fixed now.
DHCP: A 'Coder' manufacturer option longer than 31 characters could not be configured at server and not evaluated by client
| Status | Closed |
| Id | 105071 |
A coder config longer than 31 characters could not be entered in the field
"IP4/ETXn/DHCP Server/Offer Parameters/Coder" and the DHCP Client silently discarded a longer coder config possibly provided by a non innovaphone DHCP server.
phone: ip222,ip232: handle additional product id for 'Jabra BIZ 2400 Mono USB'
| Status | Closed |
| Id | 103560 |
the versions tested so far had product id 0x2401, newer ones come with 0x2401
IP241 IP222 IP232: Change back to previous DSP code
| Status | Closed |
| Id | 104862 |
Previous DSP has a better echocanceller.
Also the IP241 Handset micrphone parameters are updated.
The IP241 handset receiver equalizer is unchanged.
V9 Hotfix 28 (9061309)
Changes included in Version 9 hotfix28 Definition
PBX: In e.164 configuration a CFNR on slave PBX for call from local trunk back to local trunk did not work
| Status | Closed |
| Id | 103430 |
The object initiating the CFNR was lost when the call was sent to the master.
Collateral damage of
fix: #99674: PBX: CFNR/CFB on PBX object did not work in some cases
Media: Do not write error log if RTP is received before media negotiation is complete
| Status | Closed |
| Id | 104390 |
Do not write error log if RTP packets are received before media negotiation is complete.
Error 0x00050003 (Wrong Payload Type received) was generated before.
Increasing memory usage when viewing PBX pages with Kerberos login
| Status | Closed |
| Id | 104506 |
When the the PBX pages are displayed using a Kerberos login, some command_exec objects are never deleted. This causes increasing memory usage.
PBX E.164 Configuration: Call forward to remote Trunk, should call internal loopback destination
| Status | Closed |
| Id | 104520 |
If a call forward is configured to the switchboard of a remote location (typically -0, same as trunk prefix) the call should not be sent out to the trunk, but the internal loopback destination should be called.
EDSS1 Interworking: Interworking Of Incoming Partial Rerouting Failed
| Status | Closed |
| Id | 104610 |
A number field wasn't initialised, leading to an interworking fault at the boundary between EDSS1 and H.450.
SIP: Cannot change a password on DECT systems without restart
| Status | Closed |
| Id | 104614 |
Cannot change a password on DECT systems without restart.
Event RAS_UPDATE_KEY was not handled by SIP stack.
SIP: Memory leak when receiving BYE for a dialog in early state
| Status | Closed |
| Id | 104628 |
Memory leak when receiving BYE for a dialog in early state.
On a call which is not connected yet.
On a call where a INVITE server transaction is pending.
HTTP client: Update of nonce is ignored in digest authentication
| Status | Closed |
| Id | 104733 |
Once digest authentication is chosen the HTTP client does not accept any more changes to the digest parameters in the same session.
IP232,IP222, IP241: Fkeys may overlap call control
| Status | Closed |
| Id | 104886 |
Fkeys overlap call control in case of two inbound ringing calls.
SIP: Memory leak when receiving 403 after 401 for REGISTER
| Status | Closed |
| Id | 105022 |
Memory leak when receiving 403 after 401 for REGISTER:
REGISTER
401 Unauthorized
REGISTER with Authentication
403 Forbidden
SIP: "Spiral" was handled like "Loop"
| Status | Closed |
| Id | 105176 |
Check Request-URI checking for loop error.
Status:
Fixed in 9.00, 10.00, 10.10, 11.00
LDAP Replication: Increased Buffer for Computation of Object Differences
| Status | Closed |
| Id | 105189 |
Was to small
PBX Trunk: List of Facilities Could Get Corrupted
| Status | Closed |
| Id | 105255 |
phone: pickup notification tone too loud and tone blurred on ip110,150,200a,,230,240
| Status | Closed |
| Id | 105424 |
The volume of the pickup notification tone is derived from the volume configured for the internal ring tone.
If this volume is not appropriate it can be set to a fixed value (see http://wiki.innovaphone.com/index.php?title=Howto:Change_the_volume_of_the_pickup_key_audio_notification ).
SIP: New config file option /no-cng-tone-detection
| Status | Closed |
| Id | 105219 |
New config file option /no-cng-tone-detection
To keep calling side from initiating switch-over to T.38.
V9 Hotfix 29 (9061320)
Changes included in Version 9 hotfix29 Definition
IP-DECT: Handover not possible for accepted waiting calls
| Status | Closed |
| Id | 105747 |
Handovers are not possible for accepted waiting calls. This is fixed now.
Phones: SIP-Call was rejected if first offered codec was CLEARMODE
| Status | Closed |
| Id | 105932 |
SIP-Call was rejected if first offered codec was CLEARMODE
PBX: For pickup a wrong picked from number was displayed in case of nodes with escapes
| Status | Closed |
| Id | 106051 |
Number adjustment did not work correctly in this case
Voicemail Objekt: Trap During Reconfiguration
| Status | Closed |
| Id | 106274 |
Wasn't reproducable. Added a counter-measure against a suspected scenario.
IP6000 IP2000: Crypto driver stopped working after receiving bad SRTP packets
| Status | Closed |
| Id | 106681 |
Better protection against receiving non-SRTP packets.
IP222,IP232: Cannot move cursor rightwards in 'indirect dialing' screen
| Status | Closed |
| Id | 106791 |
Cannot move cursor rightwards in 'indirect dialing' screen.
Moving cursor leftwards works, but rightwards doesn't.
SIP: Call was dropped after successful session refresh
| Status | Closed |
| Id | 106886 |
Call was dropped after successful session refresh.
Handling of 200/OK for UPDATE was wrong.
Was wrong since bug fix #103996 (v9hotfix27)
SIP: Memory leak when receiving BYE while re-INVITE server transaction is pending
| Status | Closed |
| Id | 107066 |
Memory leak when receiving BYE right after re-INVITE.
re-INVITE server transaction is not deleted.
SIP: Do not send SDP answer twice (PRACK and ACK)
| Status | Closed |
| Id | 107107 |
Do not send SDN answer in ACK if it already been sent in PRACK.
Regards early media scenarios that starts with INVITE without offer.
INVITE(no sdp)
183(sdp offer)
PRACK(sdp answer)
200(PRACK)
180(no sdp)
PRACK(no sdp)
200(PRACK)
200(no sdp)
ACK(no sdp)
Gateway: Fix for call-replacement
| Status | Closed |
| Id | 107318 |
When handling a call leg replacement the Gateway releases the replaced call before accepting the replacement call.
May confuse the replacing endpoint.
In case of SIP this regards handling of INVITE with Replaces header.
In case of H.323 this regards handling of SETUP with ctSetup facility.
PBX Map: Overlap dial thru a Map Object on Slave with a call via the Master did not work
| Status | Closed |
| Id | 107682 |
If a phone registers from a different location, any call from this phone has to be routed via the master to check for 'local' objects. In this case overlap dialing thru a Map object on the slave did not work.
phone: importing a phonebook may result in memory leaks
| Status | Closed |
| Id | 107760 |
happens when phonebook entries containing non UTF8 characters are deleted
IP-DECT: Trap in Radio
| Status | Closed |
| Id | 108556 |
A rare trap can occur in the IP-DECT Radio with IP1202 and a multi-master solution.
phone: fine grained function locking - PHONE_LOCK_USER_INFO bit supresses display of local user info
| Status | Closed |
| Id | 105697 |
For phones installed in rooms open to the public it's sometimes required to prevent this phones from beeing called by non authorized persons. Adding this bit to the mask defined under "Phone/Protect/Fine grained Function Locking" supresses any info about the local user (number/name/display name).
"DELETE" Assertion traces caller
| Status | Closed |
| Id | 106293 |
For debugging purposes
SIP/SDP: Workaround for illegal codec signaling from Ricoh FAX
| Status | Closed |
| Id | 106513 |
Workaround for illegal codec signaling from Ricoh FAX:
\tv=0
\to=RICOH-SIP-IPFAX 1379412928 1379412928 IN IP4 130.30.3.32
\ts=Session SDP
\tt=0 0
\tm=audio 5004 RTP/AVP 18
\tc=IN IP4 130.30.3.32
\ta=rtpmap:18 G.729/8000
Must be "G729" not "G.729"!
V9 Hotfix 30 (9061325)
Changes included in Version 9 hotfix30 Definition
PBX: Number mapping for calls sent to 'Route Master calls if no Master to' sometimes wrong
| Status | Closed |
| Id | 100299 |
Esspecially in the case that the master was available but the call could not be forwarded to the final slave.
IP232,IP222,IP241: Display information of pickup fkey truncated too much
| Status | Closed |
| Id | 107962 |
Display information of pickup fkey truncated too much
wrong activation of non-existent spread-spectrum clock
| Status | Closed |
| Id | 108014 |
happens for all non-ip28 (ip22/24/302/305) gateways if hardware build >= 402, causes the gateways to stall due lack of clocking
SIP: Different registrations for the same AOR from same ip address and same port were handled as one
| Status | Closed |
| Id | 108199 |
SBC forwards different registrations for the same AOR to the PBX from same SBC ip address and SBC same port.
PBX must take this as individual registrations as long as Contact-URI differs.
Even is REGISTERs are sent from same ip address and port and for same AOR.
IP-DECT: Trap with login feature
| Status | Closed |
| Id | 108236 |
In a rare case a trap can occur if the login feature is used and the master is changed. This is fixed now.
SIP: Bug in media negotiation
| Status | Closed |
| Id | 108538 |
Bug in media negotiation when processing CFNR on an incoming SIP call received without offer.
SIP: Trap when outgoing SIP subscription is canceled while DNS is pending
| Status | Closed |
| Id | 108550 |
Trap when outgoing SIP subscription is canceled while DNS is pending.
PBX: Leak when sending group indications to an not responding endpoint
| Status | Closed |
| Id | 109270 |
Each call only a single group indication was removed from the queue, if the rate of group indications was higher then the rate of failed calls, the memory for group indications accumulated.
IP-DECT: Traces added for login feature
| Status | Closed |
| Id | 108300 |
New traces added for login feature.
PBX: Make Node/PBX at Config Template configurable
| Status | Closed |
| Id | 108506 |
For management of administration rights
V9 Hotfix 31 (9061333)
Changes included in Version 9 hotfix31 Definition
Media channel diagnostics
| Status | Closed |
| Id | 108639 |
Added some traces to support debugging of media channel issues.
SIP: Insufficient buffer space for response construction
| Status | Closed |
| Id | 109624 |
CANCEL response was not sent if received CANCEL request was bigger than expected.
E.g.
CANCEL sip:51409@10.46.17.174:5060;transport=UDP SIP/2.0
Record-Route: <sip:ea6a4b4@10.39.47.182;transport=udp;lr>
CSeq: 1 CANCEL
Call-ID: 80628647ee31e34851f74d5500
From: Surgery ;tag=80628647ee31e24851f74d5500
To: <sip:51409@st-johns.local>
Via: SIP/2.0/UDP 10.39.47.182;rport;branch=z9hG4bK736474346101292-AP;ft=10.39.47.182~13c4
Via: SIP/2.0/UDP 10.39.47.181:15060;rport=15060;ibmsid=local.1368808668750_7353594_7379782;branch=z9hG4bK736474346101292
Via: SIP/2.0/UDP 10.39.47.181:15060;rport;ibmsid=local.1368808668750_7353593_7379781;branch=z9hG4bK980490016415039
Via: SIP/2.0/TLS 10.39.47.182;branch=z9hG4bK80628647ee31e24851f74d55001-AP;ft=84340;received=10.39.47.182;rport=35249
Via: SIP/2.0/TLS 10.39.47.240;branch=z9hG4bK80628647ee31e24851f74d55001;avaya-cm-term-reaction=shortcut
Via: SIP/2.0/TLS 10.39.47.182;branch=z9hG4bK341225591747865-AP;ft=3
Via: SIP/2.0/TLS 10.39.47.181:15061;branch=z9hG4bK341225591747865;rport=36631;ibmsid=local.1368808668750_7353592_7379780
Via: SIP/2.0/TLS 10.39.47.181:15061;branch=z9hG4bK73567447322163;ibmsid=local.1368808668750_7353591_7379779
Via: SIP/2.0/TLS 10.39.47.182;branch=z9hG4bK80628647ee31e44851f74d5500-AP;received=10.39.47.182;rport=35249;ft=84340
Via: SIP/2.0/TLS 10.39.47.240;branch=z9hG4bK80628647ee31e44851f74d5500
Max-Forwards: 69
Content-Length: 0
IP-DECT: Transferred remote initiated calls without voice
| Status | Closed |
| Id | 109998 |
Some transferred remote initiated calls have no voice connection. This are calls which are initiated with myPBX to an external endpoint. This is fixed now.
Linux: DNS configuration changes device DNS
| Status | Closed |
| Id | 110073 |
The Linux DNS configuration changes the local device DNS configuration. This shouldn't be and is fixed now.
Linux: Empty server identifier and no NTP server if only ETH1 is used
| Status | Closed |
| Id | 110695 |
If only ETH1 is used and Linux gets a fixed IP address, the DHCP message doesn't include a valid server identifier and NTP server address. This is fixed now.
IP-DECT: Wrong trace warning
| Status | Closed |
| Id | 111061 |
A wrong trace warning of the last fix is removed.
Trap in webdav client when processing XML directory listing
| Status | Closed |
| Id | 111063 |
Trap in webdav client when processing XML directory listing.
IP-DECT: OEM Configuration read failure
| Status | Closed |
| Id | 111110 |
There is a read failure for an OEM configuration. This is fixed now.
DHCP: A server with "Reserved and same Vendor Clients only" checked did not provide leases to IP62 phones
| Status | Closed |
| Id | 111276 |
phone: ip241,ip222,ip232: sometimes display and USB hadrware did not recover from a reset
| Status | Closed |
| Id | 111309 |
sometimes the display and USB hardware was not working after a reset (firmware update or configuration change) and a power cycle was required to bring them up again.
phone: ip222,ip232: support Jabra UC VOICE 550 / 750 Version A headset models
| Status | Closed |
| Id | 106061 |
Headsets with Version A printed on the package have IDs different to the non-A versions even if the part numbers do not differ. The USB firmware of the Version A headsets differs from the predecessor firmware and requires a special timing.
phone: ip150: changed handset speaker parameters for hardware 102/602
| Status | Closed |
| Id | 110048 |
V9 Hotfix 32 (9061342)
Changes included in Version 9 hotfix32 Definition
PBX: Calls to 'No Master' were sent with wrong Number under special Conditions
| Status | Closed |
| Id | 109000 |
This happened if there was a call-forward, which resulted in a call to the master and the master could not send the call to the destination slave either because this Slave was not registered or there was a busy-out setting preventing it.
The problem only happend with a E.164 config.
Linux: Disable feature
| Status | Closed |
| Id | 111515 |
It can occur that the Linux cannot be disabled. This is fixed now.
phone: ip222,ip232: USB Bluetooth dongle of some "Plantronics Voyager" Headsets not detected anymore since V9hotfix24/V10rc1
| Status | Closed |
| Id | 111590 |
The Plantronics bluetooth headsets Voyager PRO UC, Voyager Legend and Calisto 620
come with an USB bluetooth dongle with one of the product codes 0415, 0416, 0417. Dongles with the product code 0416 were not detected.
phone: under soap control no audio data was sent when a call was retrieved after another call has been transferred
| Status | Closed |
| Id | 111662 |
Phones: Pickup list sometimes contains doublets
| Status | Closed |
| Id | 111725 |
Same call could be is listed more than once.
| Status | Closed |
| Id | 112186 |
REGISTER refresh was rejected "503 Service Unavailable"
if Contact header contains a not-quoted display-name.
Eg:
REGISTER sip:10.88.32.1;transport=udp SIP/2.0
Max-Forwards: 70
Content-Length: 0
Via: SIP/2.0/UDP 10.88.132.139:5060;branch=z9hG4bKd4e0fc46e
Call-ID: f68155fd504d807
From: 4044 ;tag=1a877766617814e;epid=SC2c318c
To: 4044 <sip:4044@10.88.32.1>
CSeq: 1287 REGISTER
Contact: 4044 <sip:4044@10.88.132.139:5060;transport=udp>;expires=3605
User-Agent: optiPoint 410_420/V6 6.0.55
Since fix #97150 a series of comma didn't extend the wait time before DTMF dialing any more.
| Status | Closed |
| Id | 112334 |
Since fix: "#97150: phone: DTMF digits following a comma in a number to be dialed were not handled correctly in some cases." from 21.3.2013 a series of comma didn't extend the wait time before DTMF dialing any more. The wait time was always 1 second because only the last comma was seen.
phone ip222,ip232: Plantronics Savi W440 Headset sometimes mute when controlled by a SOAP-Application or myPBX
| Status | Closed |
| Id | 112578 |
When an outbound call was started by a SOAP-Application after a call started using the headset Talk-button the headset was mute because the radio link was not established.
Gateway/H.323: Trap when canceling an call with Media Relay because out of Resources
| Status | Closed |
| Id | 112690 |
In this case the cleaup of the outgoing call was incorrect and caused a trap. Only happened when the outgoing call was H.323.
PBX: Connected Number missing on calls to some PBX objects
| Status | Closed |
| Id | 112731 |
The connected number is needed to determin if the destination of the call is internal, which is needed for features like not automatic recording of external calls
PBX: Handling of enblock (sending-complete) calls improved
| Status | Closed |
| Id | 112746 |
Respond with CallProceeding, so that if the call is rejected, there is some ack before the reject. Otherwise this would look like an error.
Trap in webdav client when processing XML directory listing
| Status | Closed |
| Id | 112764 |
Trap in webdav client when processing XML directory listing.
phone: PBX directory config page extended by Address, Gatekeeper ID and Attribute field to permit for non default values
| Status | Closed |
| Id | 111980 |
By default address and gatekeeper ID of the PBX where the user is registered are used and the 'Long Name' is searched. Now for example this can be changed to use the master PBX and to search the 'Display Name'.
phone: set up call with "Sending complete" when the number has been provided before the call is initiated
| Status | Closed |
| Id | 112103 |
This applies to calls initiated while browsing a directory or a call list, by pressing a dial function key or via indirect dialing, i.e. when a number is entered before going off-hook.
To permit for incomplete numbers in a phone directory "Sending Complete" is not set when a number is terminated by a '+' character. Then the '+' is stripped off and the number can be completed by typing more digits.
In this case and in case the user goes off-hook before typing any digit the number is assumed to be complete when a '#' character is entered or the "Enblock Dialing Timeout" is reached before the next digit was entered.
The old overlap sending behaviour can be restored by
config add PHONE SIG /overlap-sending
phone: ip222,ip232: support for Jabra BIZ 2300, Sennheiser Presence UC
| Status | Closed |
| Id | 112335 |
V9 Hotfix 33 (9061347)
Changes included in Version 9 hotfix33 Definition
H.323: The efc-features were not forwarded accross PBXs from an endpoint, which was called with slowstart
| Status | Closed |
| Id | 112037 |
If a slowstart endpoint performed a transfer, connecting two efc endpoints on other PBXs, it could happen, that the media negotiation between the new endpoints was slowstart, because the PBX on which the transfer was performed did not receive the efc-featurse
IP222 IP232: Option to disable Energy Efficient Ethernet (EEE) added
| Status | Closed |
| Id | 112979 |
Needed for some PCïs that loose the link with EEE.
EEE status display added to V9 and V10.
phone ip222,ip232: phone keypad locked when digits are entered too fast (can be unlocked by ESC key)
| Status | Closed |
| Id | 113068 |
H.323: unexpected Restart on a very unlikly Hold/Disconnect collision
| Status | Closed |
| Id | 113079 |
If the two events happened during the same couple of microseconds an assertion in the code caused a restart.
PBX: The top level Tag of a CDR should contain the normalized number of the endpoint it was created for
| Status | Closed |
| Id | 113112 |
This was sometimes not the case, but only the extension number without node prefixes was included.
DNS: Services/DNS/Query Caused A Trap
| Status | Closed |
| Id | 113137 |
An internal buffer length check was wrong
phone ip222,ip232: added config flag to prevent ringing via speaker when a headset is plugged and enabled
| Status | Closed |
| Id | 113263 |
config add AC-DSP0 /headset-only
unconditionally disables ringing via speaker when a headset is plugged and enabled. this is done independent of the "Do not Disturb" setting.
Voicemail: <exec> Without "url" Causes Trap
| Status | Closed |
| Id | 113428 |
A check was missing
Flash Directory: Config-Encoding Of Objects Breaking Through 8K Line Length
| Status | Closed |
| Id | 113851 |
This fix just helps where an object's representation within the configuration file expands beyond the 8K barrier.
http://wiki.innovaphone.com/index.php?title=Concept_Flash_Directory#Config-Encoding_Of_Objects_Breaking_Through_8K_Line_Length
ip1202: config flag to force reboot when receive interrupts are missing for a certain time
| Status | Closed |
| Id | 113948 |
By default the the MAC is is reset in case of missing receive interrupt.
config add ETH0 /rx-miss-reboot
forces a reboot instead of a MAC reset.
config add ETH0 /rx-wait-max <seconds>
defines the maximum time to wait after the last receive interrupt before MAC reset or reboot (default is 30 seconds).
config add ETH0 /itrace
activates an interupt backlog which is written to trace buffer before MAC reset or reboot.
TLS: Problem with negotiation of protocol version on server side
| Status | Closed |
| Id | 114046 |
When the client offered TLS 1.2 or higher, the connection was refused instead of downgrading to the highest supported protocol version.
NT ISDN Point to Multipoint Interfaces: Rejecting of a call had delay of 4.5s
| Status | Closed |
| Id | 115118 |
A call was not rejected right away, but SETUP was resent in case another endpoint would respond. This should be done only if the call was rejected because of incompatible destination.
PBX: If an endpoint performs a pickup-req, the resulting call should be sent to the requesting endpoint only
| Status | Closed |
| Id | 115569 |
If on a user two phones were registered and one phone performed a pickup, both phones were ringing for the call to be picked up.
V9 Hotfix 34 (9061355)
Changes included in Version 9 hotfix34 Definition
PBX: Pickup call was not indicated as internal in Connected Number
| Status | Closed |
| Id | 113197 |
This could cause problems were it is important to know if the call is internal or external, for example if only external calls are to be recorded.
phone: In Recording Mode 'transparent' or'optional' internal calls were recorded although 'External Calls Only' was checked
| Status | Closed |
| Id | 114516 |
happened only to outbound calls initiated by some dialing application.
outbound calls initiated directly at the phone and inbound calls were recorded correctly.
Gateway trap with 'Out of Memory' when CF-card stucks
| Status | Closed |
| Id | 114781 |
A CF-card that stucks leads to huge memory allocations of type cf_command containing non-processed CF-requests.
phone: In Recording Mode 'transparent' or'optional' a 2nd call started by a dialing application could terminate the 1st call
| Status | Closed |
| Id | 114789 |
This happened when a 2nd call was started by a dialing application and then terminated again while the call was in alerting state.
H.323: No media if a reverse Media call is sent to a slowstart endpoint and tranfered to a EFC endpoint
| Status | Closed |
| Id | 115018 |
Media negotiation problem which could happen under special conditions when an XCAPI application is performing a call transfer
DHCP: Increase maximum length of "Local Networks" and "IP Routing" option strings from 127 to 252 characters
| Status | Closed |
| Id | 115709 |
Primary Address for "Alarm and Event Forward Server" of type SYSLOG could not be configured
| Status | Closed |
| Id | 115745 |
(clone of #114012) SIP: Trap in federation scenario
| Status | Closed |
| Id | 116005 |
Trap in federation scenario when processing INVITE.
ISDN: Missing Ringback on calls sent out to an NT Mode interface
| Status | Closed |
| Id | 116390 |
If there is no progress indicator indicating inband tones, channels should not switched on for calls sent out to an NT Mode ISDN interface. Otherwise RTP containing silence could switch off any locally generated ringback.
V9 Hotfix 35 (9061367)
Changes included in Version 9 hotfix35 Definition
phone: two way media on a recording connection did not work anymore
| Status | Closed |
| Id | 103956 |
Since v9hotfix22/v10beta6 recording connections are established in sendonly mode because usually recorders do not send any media data and thus bulks of ?No Media Data received? events may be reported.
Now two way media can be explicitely enabled by checking "Phone/User x/Recording/Two Way Media" if required (for example for the Innovaphone Operator "Greeting Function").
Gateway trap with 'Out of Memory' when CF-card stucks
| Status | Closed |
| Id | 114781 |
A CF-card that stucks leads to huge memory allocations of type cf_command containing non-processed CF-requests.
SIP: The 2xx response to the REGISTER request MUST contain, in a Contact header field, a complete list of bindings
| Status | Closed |
| Id | 116835 |
The 2xx response to the REGISTER request MUST contain, in a Contact header field, a complete list of bindings that have been registered for this address-of-record.
Not only the one that has just been added.
SIP: Interop to "Thomson TB30 hw4 fw2.77.0.2 00-26-44-30-5D-A2"
| Status | Closed |
| Id | 117149 |
Thomson TB30 hw4 fw2.77.0.2 00-26-44-30-5D-A2
uses slightly different Contact-URI for registration:
"sip:102-dkkn2jgs10ffe@10.35.0.133:5060;user=phone;transport=udp"
and subscription:
"sip:102-dkkn2jgs10ffe@10.35.0.133:5060;transport=udp"
Subscription is rejected.
H.323: Don' generate "Unexpected Message" event for messages received after sending call clearing
| Status | Closed |
| Id | 117248 |
These messages are not unexpected, but results of a normal collision
ip1202: DTMF tones to be sent to the local DECT phone were sent to the voip channel
| Status | Closed |
| Id | 117254 |
thus DTMF tones sent from a remote peer were not heard by the local peer
IP-DECT: Resent disconnected calls to handsets
| Status | Closed |
| Id | 117376 |
Calls disconnected by the gatekeeper can be wrongly resent to the handsets, if the calls are disconnected with the release code Non-selected-user-clearing (26). This is fixed now.
PBX Gateway: Internal Destination flag did not work for outgoing calls
| Status | Closed |
| Id | 117418 |
If external transfers are not allowed, this flag should allow a transfer to a gateway object for a call coming in from an external source.
H.323: Internal/External information got lost on Endpoint after Transfer
| Status | Closed |
| Id | 118350 |
The information if the endpoint to which a phone is connected after a transfer is internal or external was not available on the phone. The recording of internal or external calls only did not work in this case.
phone: ip222,ip232: support Jabra Pro 935 USB-Bluetooth Headset
| Status | Closed |
| Id | 117060 |
The Pro 935 looks like a Pro 930 but has a bluetooth- instead of a DECT-headset. The bluetooth-headset can be paired with a mobile phone.
V9 Hotfix 36 (9061368)
Changes included in Version 9 hotfix36 Definition
phone: keep remote party name after connect when dialled and connected number differ in first digits only
| Status | Closed |
| Id | 118537 |
PBX Exec: Partner Keys at exec did not work correctly if secretary names matched in the first half
| Status | Closed |
| Id | 118869 |
If two secretaries were configured with names, being identical in the first half and identical length (e.g. 'Hans' and 'Harz'), for some functions like presence status not the correct secretary was found.
phone: ip110/150/200a/230/240: false "Excessive loss of Data" reports when playing Music on Hold (MOH)
| Status | Closed |
| Id | 119055 |
phone: CLIR couldn't be overridden at phone by "Number Presentation: On" when "Hide own Number" was checked in a config template
| Status | Closed |
| Id | 119270 |
Overriding via WEB interface works
Waiting Queue: Switching to next announcement by DTMF "0" did not work
| Status | Closed |
| Id | 119525 |
Switching to next announcement by DTMF "0" did not work.
SIP: Bug in media re-negotiation
| Status | Closed |
| Id | 119635 |
Bug in media re-negotiation on media-relay interfaces.
V9 Hotfix 37 (9061372)
Changes included in Version 9 hotfix37 Definition
ip1202: improved ethernet receive error handling
| Status | Closed |
| Id | 120007 |
- workaround for 10/100Mb/s gemac Rx lockup:
the interface is run in promiscuous mode and the driver filters the packets
- workaround for Rx Queue Overrun problem:
on a Rx Queue Overrun interrupt gemac and phy are rest completely
- for test purposes promiscuous mode can be disabled|enabled by
!config add ETH0 /rx-promiscuous 0|1
or temporaryly by
!mod cmd ETH0 rx-promiscuous 0|1
PBX: Pickup with partner key did not work if visibility was configured by name
| Status | Closed |
| Id | 120070 |
The call was displayed on the partner key, but the pickup did not work. It did work if visibility was configured with a group.
PBX-SOAP: Clearing of call to waiting queue took 30s
| Status | Closed |
| Id | 120231 |
Call was hanging as if there was in-band information
FXS with Feature Codes, possible trap on call-completion
| Status | Closed |
| Id | 120384 |
When call completion was executed, there was a chance of a trap under special conditions
IP-DECT: Rare trap on IP1202
| Status | Closed |
| Id | 120442 |
There is a rare trap in DECT-Master if a new call is sent to the radios and there still exists an old call for the endpoint and this call is assigned to an unregistered radio. The trap only occurs on the IP1202, not the IP1200. This is fixed now.
DHCP-Server: strip leading and trailing spaces from values entered in "IP4/ETHx/DHCP-Server/Offer Parameters"
| Status | Closed |
| Id | 120514 |
PBX: Trap on Park/Pickup
| Status | Closed |
| Id | 120579 |
If a Park function key is used to park a call and pickup it again, a restart happend. This is a collateral damage from
fix 115569: PBX: If an endpoint performs a pickup-req, the resulting call should be sent to the requesting endpoint only
from v10sr8 and v9hf33
PBX: Pickup accross locations from different nodes did not work
| Status | Closed |
| Id | 120638 |
Adjustment of number was missing
SIP: Offered wrong local IP address as RTP address
| Status | Closed |
| Id | 120739 |
Offered wrong local IP address as RTP address.
Collateral damage of #119269: SIP: Offered wrong local IP address as RTP address
phone: ip222,ip232: Jabra UC VOICE 550/750 Version A - Microphone occasionally mute
| Status | Closed |
| Id | 120815 |
V9 Hotfix 38 (9061386)
Changes included in Version 9 hotfix38 Definition
phone: ip222,ip232: audio parameter configuration via command line did not work in some cases
| Status | Closed |
| Id | 121065 |
Happened with command lines containing options without a value, for example a
config change AC-DSP0 HEADSET /spk-volume /mic-volume 5
did not affect the microphone volume.
Further input was not validated so big negative or positive values gave confusing results.
PBX: Transfer with consultation in ringback - no ringback after transfer if performed by analog phone on IP22/.../IP28
| Status | Closed |
| Id | 121197 |
It is not a problem of the analog interface, but the PBX, which does not play ringback if a retrieve is done before the transfer, which is done by the FXS.
PBX: When a dyn PBX was deleted, with an id identical to the start of the id of another dyn PBX, this other dyn PBX was broken
| Status | Closed |
| Id | 121433 |
Some VARS of the wrong dyn PBX were deleted
IP-DECT: Watchdog trap on IP1202 with feature codes and handovers
| Status | Closed |
| Id | 121456 |
There is a watchdog trap on IP1202 with enabled feature codes with firmware V9 hotfix37, if more than one master is used and the handset makes a handover. This is fixed now.
SIP: Wrong expires value in Contact header of 2xx response for REGISTER
| Status | Closed |
| Id | 121641 |
Wrong since v10sr9, v9hotfix35.
Wrong expires value in Contact header during registration refresh.
Correct value in Expires header.
pbx: memory leak when trace is active
| Status | Closed |
| Id | 121897 |
PBX SOAP: Trap if trying to initiate a call for a User with Mobulity configured, without specifiying the device
| Status | Closed |
| Id | 122005 |
With the PBX SOAP API a call can be initiated for a user, without specifying for which device the call should be initiated. In this case a default device is picked. If an application does this for a user with mobility, a restart happens because of a null pointer access.
PBX CDRs: CDR was missing for calls rejected because of busy_on ...
| Status | Closed |
| Id | 122014 |
This was fixed in version 10 already and now merged back to version 9
Admin; The input field for the device name showed the url-decoded name
| Status | Closed |
| Id | 122076 |
If a name with '+' or '%' was configured as device name, these charecters were nocz displayed correctly in the input field.
phone: do not mute microphone in alerting state
| Status | Closed |
| Id | 122205 |
For some some analogue endpoints it is not possible to detect when the media connection is really established, it may hapen before connect is signaled to the phone. To prevent confusion when voice is received from remote but the answer is supressed the microphone is unmuted now already in alerting state by default.
The former behaviour can be restored by
config add PHONE APP /mute-while-dialing 1
phone recording - supress calling tones and call status display for calls to recording device
| Status | Closed |
| Id | 122221 |
Adjusting the Volume level of Local Playback of DTMF Tones - Marcus Mlbsch
| Status | Closed |
| Id | 122706 |
The customer complained that DTMF feedback tones to the user were too loud. There was one obvious reason in the sources: the table of VoiceOutputGain steps was not in sync with the table of SignalLevel steps, i.e. the relation between speech and DTMF level differed depending on the volume setting. The other point is that DTMF tones are perceived louder than the lower frequency call progress tones. Therefore adjusted the SignalLevel steps to correlate with the VoiceOutputGain steps and introduced 6 dB extra attenuation for DTMF tones compared to call progress tones.
SIP: Bug in media negotiation when processing reINVITE without SDP offer
| Status | Closed |
| Id | 122780 |
Bug in media negotiation when processing reINVITE without SDP offer.
Exclusive codec config got lost during call.
SIP: Mobility did not work with SIP
| Status | Closed |
| Id | 122942 |
Mobility did not work with SIP since RTP-DTMF was ot suppressed.
IP-DECT: Trace and variable check added
| Status | Closed |
| Id | 121496 |
Trace and variable check in IP-DECT master added to give some hints of bugs.
V9 Hotfix 39 (9061388)
Changes included in Version 9 hotfix39 Definition
FXS: Trap on very rare race collision of retrieve with call release
| Status | Closed |
| Id | 122980 |
If a retrieve happens at the same time as a call release of the held call, a trap could happen. The propabilty of this to happen was very low.
phone: ip241: 'Ok' key inserts newline characters in number/name input fields
| Status | Closed |
| Id | 123369 |
This way numbers may be misinterpreted as names.
IP222 IP232: Propietary SmartEEE disabled
| Status | Closed |
| Id | 124415 |
Needed for some PCïs that loose the link with EEE.
This is caused by the "propietary smartEEE " feature of the ethernet phy.
When disabled the link is stable
Regular EEE still works.
V9 Hotfix 40 (9061390)
Changes included in Version 9 hotfix40 Definition
QSIG: Progress Indicator was missing in PROGRESS on call to Busy User
| Status | Closed |
| Id | 124957 |
This did not do any harm, but an error log was generated in a Unify PBX.
V9 Hotfix 41 (9061392)
Changes included in Version 9 hotfix41 Definition
phone: ip222,232: support Jabra UC VOICE 750 MS Duo Drk PN 7599-829-409 29/12/14 Version:A
| Status | Closed |
| Id | 126419 |
Just another model of the Jabra UC VOICE 550/750 Series with a new product ID
SIP: No reINVITE with updated identity was sent
| Status | Closed |
| Id | 126722 |
No reINVITE with updated identity was sent on interfaces with media-relay and exclusive codec.
PBX: Node prefixes missing in CDRs from Broadcast object
| Status | Closed |
| Id | 127131 |
The number of the object which generated the CDRs shall contain the normalized number including all prefixes.
phone: ip110,200,230,240: sometimes no DTMF tone was sent for a digit entered in an active call
| Status | Closed |
| Id | 127693 |
-
CONF Interface: Noise on IP800/IP305
| Status | Closed |
| Id | 127695 |
There are noise and peaks in a conference call on the IP800 and IP305 caused by the CONF interface. This is fixed now.
Phone sometimes returns to speaker-mode instead to release the call when going onhook
| Status | Closed |
| Id | 127990 |
This could happen when handset was lifted while holding speaker key pressed or when the speaker was relased at the same time when the handset was lifted.
IP-DECT: Default config change
| Status | Closed |
| Id | 128180 |
There is a change in the default configuration which prevents a wrong configuration if a software factory reset is done with the command config clear. IP-DECT handover fails with the wrong configuration. This is fixed now.
PBX Map: Hide Connected Endpoint, should be evaluated for calls thru Map
| Status | Closed |
| Id | 127834 |
This is a feature introduced with v10, which is now merged back to v9
V9 Hotfix 42 (9061397)
Changes included in Version 9 hotfix42 Definition
PBX: Avoid hanging calls after unsuccessful blind transfer to busy endpoint
| Status | Closed |
| Id | 128908 |
Happen with a Multicast object: Call to multicast, then hold and another call to the same multicast object, which returns busy, then hangup, which initiates a blind transfer to busy endpoint.
Media Relay: Support for unknown audio codecs
| Status | Closed |
| Id | 129907 |
Support for unknown audio codecs (e.g. iLBC) when forwarding RTP audio (RTP-Proxy, Media-Relay).
Trap in PBX during boot after downgrade from v11
| Status | Closed |
| Id | 130015 |
Trap in PBX during boot after downgrade from v11.
phone: with optional recording a 3-party conference could not be established although recording was stopped
| Status | Closed |
| Id | 130286 |
Call flow:
- A calls B
- B answers -> recording is started
- A stops recording with redial-key
- A opens a consultation call to C
- C answers -> recording is restarted
- A stops recording with redial-key
When A presses the Menu-key a 3-party conference should be started.
This did not work anymore since V9hotfix25 and not at all in V10.
V9 Hotfix 43 (9061401)
Changes included in Version 9 hotfix43 Definition
Voicemail failed in Chief+Secretary Scenario
| Status | Closed |
| Id | 114863 |
Audio prompting didn't start
ISDN: Call was rejected without cause on channel collision
| Status | Closed |
| Id | 131351 |
This is a protocol violation and created unwanted log entries on the other side
PBX Routing: A objcet shadowing a node escape should be used as node extern as default
| Status | Closed |
| Id | 131645 |
So that for calls to a local trunk no node extern needs to be configured
IP-DECT: Debug for rare trap
| Status | Closed |
| Id | 131677 |
A debug message in case of a rare bug is added.
IP-DECT: Feature codes trap
| Status | Closed |
| Id | 131806 |
There can be a trap with feature codes caused by an uninitialized variable. This is fixed now.
GUID generation fixed. Could result in duplicate GUIDs
| Status | Closed |
| Id | 132117 |
This could create problems in different places for example in Reporting when two CDRs with same GUID were sent.
phone: ip222/232 - support for new Jabra EVOLVE headset series and for additional Plantronics Blackwire headsets
| Status | Closed |
| Id | 130552 |
-
phone: ip222/232 - Config: Reject Automatically Connected Inbound Call routed to Headset if Headset is not plugged or disabled
| Status | Closed |
| Id | 130575 |
Using the configuration given below an inbound call is automatically connected to the headset if a headset is plugged and enabled, otherwise the call is rejected with cause busy.
"Phone/User-x/Preferences/Announcement Calls/Micro On"
"Phone/User-x/Preferences/Announcement Calls/Treat any Call as Announcement"
"Phone/Preferences/Route Automatically Connected Inbound Calls to Headset (if enabled)"
"Phone/Preferences/Reject Automatically Connected Inbound Call routed to Headset if Headset is not plugged or disabled"
The last checkmark affects only normal inbound calls. Announcement calls via the PBX MCAST-Announce object or via the "Dial/Announce" Function key will be routed to the speakerphone if no headset is plugged or if the headset is disabled.
phone: ip222/232 - added support for Jabra BIZ 2300 USB Duo headset
| Status | Closed |
| Id | 132131 |
V9 Hotfix 44(9061405)
Changes included in Version 9 hotfix44 Definition
SIP: Fix for trap
| Status | Closed |
| Id | 132389 |
Fix for trap due to failed assertion.
SIP: Cannot call from SRTP endpoint to non-SRTP endpoint
| Status | Closed |
| Id | 132448 |
If called non-SRTP endpoint rejects, the call is re-tried as RTP call without encryption.
But no if the caller is a SIP endpoint.
PBX CDRs: Records for calls to objects without registrations were missing
| Status | Closed |
| Id | 132656 |
Old fix from v10 merged to v9
V9 Hotfix 45 (9061408)
Changes included in Version 9 hotfix45 Definition
PBX Waiting: Cause got lost, when disconnecting a waiting calls with SOAP
| Status | Closed |
| Id | 133887 |
The call was disconnected without cause, which typically resulted in a display "call aborted" on the calling endpoint instead of "user busy" which could be desired by the application.
IP305, include interface licenses in firmware
| Status | Closed |
| Id | 133913 |
IP305 is sold with full interface licenses only
myPBX 9: Script error when using popup windows
| Status | Closed |
| Id | 134965 |
After an update, Internet Explorer 11 showed a script error "Access Denied" when using popup windows in myPBX 9. This problem only occured when using the myPBX launcher.
<--
client.js
-->
AD Replication: Oversized AD Objects Deleted Replicated Objects
| Status | Closed |
| Id | 135034 |
An internal error code wasn't set by a handling for the resulting decoding failure. The internal error code is now set to error=86, "LDAP Decoding Error".
The replication will stop consequentially.
Actual cause were the AD objects being member in too many AD groups.
phone: ip222,232: audio connection to remote conference peer sometimes lost after a coder renegotiation on one connection
| Status | Closed |
| Id | 135080 |
This problem occured in the folllowing situation:
- a local call (audio+video) was established via an USB headset and then put on hold
- a consultation call (audio) to an external peer was established
- a 3-pty conference was established but the local connection remained mute
Trap: Flash Directory: LDAP Substring Search Caused MAX_BUSY_TICKS
| Status | Closed |
| Id | 135368 |
Consequtive asterisks weren't skipped.
V9 Hotfix 46 (9061415)
Changes included in Version 9 hotfix46 Definition
PBX/IP6000: Potential restart if there are groups or boolean objects with non-Ascii characters
| Status | Closed |
| Id | 111560 |
This is a general problem that the strcmp from the standard lib does not work correctly under very special conditions.
Incomplete HTTP responses from HTTP server in certain circumstances
| Status | Closed |
| Id | 138895 |
It might have happened, that the HTTP server closed the underlying TCP connection before all data could be sent.
FAX Interface: Hanging calls
| Status | Closed |
| Id | 139602 |
If a call setup to the FAX interface includes a user-user-information element, the call hangs. This is fixed now.
SIP: Remove all bindings did not work
| Status | Closed |
| Id | 140071 |
A REGISTER with "Contact: *" was not handled as it should.
IP-DECT: Reverse phone book search configuration
| Status | Closed |
| Id | 140086 |
The IP-DECT reverse phone book search accepts a configuration with phone number types like e164:H,mobile:M now.
H.323: Call to a Call Broadcast Destination failed under special conditions
| Status | Closed |
| Id | 140491 |
DTMF dial from a Waiting Queue to a Call Broadcast object with many destinations. This caused special timing in H.323, which created the problem.
V9 Hotfix 47 (9061418)
Changes included in Version 9 hotfix47 Definition
SIP: Switch from Media-Relay to No-Media-Relay when handling INVITE with Replaces
| Status | Closed |
| Id | 139046 |
Switch from Media-Relay to No-Media-Relay when handling INVITE with Replaces.
May result into no media after INVITE with Replaces.
PBX: OEM Registration licenses did not work anymore
| Status | Closed |
| Id | 142371 |
New handling of license versions broke the OEM licenses
PBX Waiting: Cause got lost, when disconnecting a waiting calls with SOAP (again)
| Status | Closed |
| Id | 142458 |
Previous fix did not work
IP-DECT: Master trap
| Status | Closed |
| Id | 143206 |
There is a Master trap because of an uninitialized variable within a facility call. This is fixed now.
V9 Hotfix 48 (9061420)
Changes included in Version 9 hotfix48 Definition
PBX: Execute CFB on Trunk/Gateway objects, if the far endpoint rejects call with busy
| Status | Closed |
| Id | 143675 |
This is useful to do re-routing in case of a called service is busy
HTTPCLIENT: trap when an application cancels a request inmidst DNS-name resolution
| Status | Closed |
| Id | 143738 |
Memory leak in the hardware encryption driver of the IP6000
| Status | Closed |
| Id | 143945 |
Under excessive load some packets allocated in memory were sometimes not freed in the hardware encryption driver of the IP6000.
Fixed possible trap on CF card error
| Status | Closed |
| Id | 144019 |
The box might have trapped on CF card errors (card full, invalid data read etc.)
Phones: Partner fkeys with subscriptions or favourites may not work
| Status | Closed |
| Id | 144707 |
Partner fkeys with subscriptions or favourites may not work in some cases.
But only if partner's name is used as destination
and if namesmatch partly.
E.g. "name" and "name.x"
IP6000: Prevent blinking error LED on old IP6000 with HW-Build 201
| Status | Closed |
| Id | 147092 |
Conference DSP driver was started on old hardware that doesnt support the conference DSP
V9 Hotfix 49 (9061421)
Changes included in Version 9 hotfix49 Definition
LDAP: Trap in Flash Directory UI
| Status | Closed |
| Id | 145405 |
A deleted memory region was re-accessed.
IP-DECT: Wrong name with reverse phone book search
| Status | Closed |
| Id | 145482 |
If there is a similar number in the LDAP directory, the number can be resolved in a wrong name. This is fixed now.
IP-DECT: Phone book search filter
| Status | Closed |
| Id | 145556 |
The configured phone book search filter isn't considered in the search string. This is fixed now.
Licenses containing digits (e.g. G729channel) did not work
| Status | Closed |
| Id | 146486 |
Problem parsing the license string
V9 Hotfix 50 (9061427)
Changes included in Version 9 hotfix50 Definition
SIP: Changed handling of History-Info header and stop sending Diversion header
| Status | Closed |
| Id | 147429 |
Trying to comply to RFC-7044 and RFC-7131.
Decoding: Skip top-most entry "History-Info" (highest index value) if this entry reflects the called party itself.
Encoding: Add top-most entry "History-Info" (highest index value) that reflects the called party itself.
SIP header "Diversion" is removed since it is declared as deprecated (RFC-5806 Category Historic now).
OEM Registration licenses did not work anymore
| Status | Closed |
| Id | 150069 |
Collateral damage of fix: #146486: Licenses containing digits (e.g. G729channel) did not work
SHA-2 hash algorithms
| Status | Closed |
| Id | 113239 |
Port the hash algorithm to our platform.
Support for SHA2 certificates
| Status | Closed |
| Id | 113352 |
- encoding and decoding
* verification
* create such certificates on boxes (except sha224)
Signature algorithms:
* sha224WithRSAEncryption { pkcs-1 14 }
* sha256WithRSAEncryption { pkcs-1 11 }
* sha384WithRSAEncryption { pkcs-1 12 }
* sha512WithRSAEncryption { pkcs-1 13 }
V9 Hotfix 51 (9061429)
Changes included in Version 9 hotfix51 Definition
Fax server: Wrong error correction
| Status | Closed |
| Id | 151280 |
The error correction doesn't work if it is necessary. It results in missed document parts or failed connections. This is fixed now.
PBX: CFB on Trunk or Gateway did not work if the call was cleared with DISC
| Status | Closed |
| Id | 151934 |
This happend for example on ISDN interfaces with in-band busy tones
PBX: Twin Phone algorythm did not work for transfer/recall
| Status | Closed |
| Id | 152169 |
A recall after a transfer should also use the twin phone algorythm. For example if one of the phones is busy, the call should be sent to the busy phones only.
SIP: New config option /send-deprecated-diversion-header
| Status | Closed |
| Id | 152337 |
Diversion header is not sent anymore since v11r1sr5 / v11r2sr1 / v10sr24 / v9hotfix50.
For interop reasons this config option is added.
If set the old and deprecated Diversion header is sent.
V9 Hotfix 52 (9061432)
Changes included in Version 9 hotfix52 Definition
Web-UI: Font-family of input, select, textarea, button did not inherit body style
| Status | Closed |
| Id | 153879 |
Font-family of input, select, textarea, button did not inherit body style.
Using now "font-family:inherit" to have same font-familiy all over.
IP-DECT: Fix for "Wrong name with reverse phone book search"
| Status | Closed |
| Id | 154071 |
Since the fix "Wrong name with reverse phone book search" it doesn't work. This is fixed again.
IP2x P30x IPxx10: Tone is sometimes not switched off
| Status | Closed |
| Id | 153129 |
-
V9 Hotfix 53 (9061433)
Changes included in Version 9 hotfix53. Definition
V9 Hotfix 54 (V9 Hotfix 54 (9061435)
Changes included in Version 9 hotfix54. Definition
PBX SOAP: Remote number update missing on blind transfer on another PBX
| Status | Closed |
| Id | 166764 |
The CT-COMPLETE facility used to transmit the new number, was not used to update SOAP call
IP-DECT: Forced logout does not store CKI
| Status | Closed |
| Id | 167108 |
If an users logs in a handset and a previously used handset is logged out, the cipher key index for early encryption isn't saved for this handset. This is fixed now.
CF/SATA driver: Disturbs Linux SATA driver at start-up
| Status | Closed |
| Id | 167567 |
The innovaphone CF/SATA driver can disturb the Linux SATA driver at Linux start-up, Linux recognizes a spurious interrupt and disables wrongly the SATA interrupt. The SATA device doesn't work or works slowly. This is fixed now.
V9 Hotfix 55
Changes included in Version 9 hotfix55. Definition