Reference10:Release Notes Firmware
This is the Firmware V10 Roadmap Document.
Service Releases are planned for the second monday each month.
This article is generated automatically. Do not edit! Please see the disclaimer before using the information presented here!
V10 Service Release 1 (100889)
Changes included in Version 10 Service Release 1 Definition
SIP: Registrations were not refreshed in time on large DECT installations
| Status | Closed |
| Id | 102556 |
Registrations were not refreshed in time on large DECT installations.
Active REGISTER transactions were limited to 100.
Refresh transactions are executed with priority now.
myPBX launcher: Some texts do not have enough space in config dialog
| Status | Closed |
| Id | 102675 |
Some of the additional translations of SR1 need more space in the configuration dialog of the launcher.
Phones: Wrong user info on idle screen of IP240/230/200/110
| Status | Closed |
| Id | 103006 |
Wrong user info on idle screen of IP240/230/200/110.
Local user's display name was displayed instead of local user's name (h323id).
PBX Session Border: Potential Trap when trying to register SIP endpoints
| Status | Closed |
| Id | 103064 |
The trap could happen when the Session Border object was changed or, when an incoming registration was cleaned up because the internal registration failed.
PBX Session Border: Trap when calling a SIP client with Media Relay
| Status | Closed |
| Id | 103091 |
Happened each time.
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.
Video: 3rd party conference not working properly
| Status | Closed |
| Id | 103223 |
Conference socket was not initialized correctly.
myPBX: Do not show "anonymous" for outgoing calls
| Status | Closed |
| Id | 103232 |
"anonymous" was always displayed if there was no number or name. This should only be displayed for incoming calls.
SIP: Honor Retry-After header in 500 response to REGISTER
| Status | Closed |
| Id | 103282 |
Retry after the given time instead of giving up right away and possibly trying the alternate addess.
myPBX: Script error on line 3493
| Status | Closed |
| Id | 103301 |
After logging-out and in again, the search input field throws a script error when pressing enter or cursor keys.
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: 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
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
IP110A: Firmware doesnt start after bootcode update to V10 final
| Status | Closed |
| Id | 103638 |
IP110A: After bootcode was updated to V10 final the bootcode cannot access the flash memory and the phone doesnt start the firmware.
To fix this use gwload to load a special code "fix110.bin" to DRAM. This restores the original bootcode during startup. After the next reset the IP110A firmware starts again from flash.
gwload /setip /i <addr> /gwtype 110 /dram fix110.bin
The binary is at \\\\inno-sifi\\dfs\\build\\10.00\\phone_inca\\100878\\fix110.bin
Bootcode build 10880 is fixed.
IP110 Bootcode has been removed from the V10 final downlod package.
PBX: Forking should not be executed on calls to WQ operators
| Status | Closed |
| Id | 103657 |
The forking was executed, but did not work correctly. The call did not stop ringing on the original phone, when the forked call was connected.
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 |
phone: builtin test result popup message missing
| Status | Closed |
| Id | 103857 |
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").
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.
PBX Broadcast: It could happen that not all phones ring when a Broadcast object was called
| Status | Closed |
| Id | 104002 |
Under special conditions, for example if the call was done thru DTMF two stage dialing on a Waiting Queue with pcap traces runing in parallel.
Gateway: FAX interface training timing
| Status | Closed |
| Id | 104077 |
Outgoing fax calls fail using a SIP carrier because of a wrong training timing. This is fixed now.
SIP: Error while processing NOTIFY(dialog-info)
| Status | Closed |
| Id | 104083 |
Error while processing NOTIFY(dialog-info).
Call state was not decoded properly.
| Status | Closed |
| Id | 104092 |
Gateway: FAX interface transmit timing with ECM
| Status | Closed |
| Id | 104161 |
Outgoing fax calls with error correction mode fail because of a too short send timing. This is fixed now.
Gateway: FAX interface minimum scan line time with ECM
| Status | Closed |
| Id | 104191 |
The minimum scan line time for outgoing fax calls with ECM is not correctly set. This is fixed now.
myPBX: Update translations
| Status | Closed |
| Id | 102796 |
Improved translations.
New language: Hungarian
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
SIP: Re-use inbound TCP connection to send request to client behind NAT
| Status | Closed |
| Id | 104033 |
Re-use inbound TCP connection in case of NAT.
Re-use inbound TCP connection to send request to client.
V10 Service Release 2 (100897)
Changes included in Version 10 Service Release 2 Definition
IP-DECT: Trap with call transfer, overlap dialing and OEM PBX
| Status | Closed |
| Id | 104203 |
There is a potential trap with call transfer, overlap dialing and an OEM PBX. This is fixed now.
SIP: Media-Relay on an outgoing PBX call was missing, when call was initiated by INVITE w/o SDP
| Status | Closed |
| Id | 104211 |
Media-Relay on an outgoing PBX call was missing, when call was initiated by INVITE w/o SDP.
Gateway: FAX interface minimum scan line time with ECM
| Status | Closed |
| Id | 104262 |
The minimum scan line time for outgoing fax calls with ECM is not correctly used. This is fixed now.
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.
SIP: Handling of re-INVITE getting stuck
| Status | Closed |
| Id | 104398 |
A re-INVITE may take too long to get answered.
Start next pending re-INVITE after timeout.
("N510 IP PRO/42.078.00.000.000" took more than 3 seconds to handle re-INVITe(sendonly))
Media: NAT workaround did not start in some cases
| Status | Closed |
| Id | 104399 |
NAT workaround (send outgoing RTP to source of incoming RTP) did not start in some cases.
Esp. after re-negotiation for HOLD and RETRIEVE.
SIP: Signaling instance not always cleaned up
| Status | Closed |
| Id | 104497 |
Signaling instance not always cleaned up when Standby Master goes back to standby.
Big scale applications only.
SIP: Re-use inbound TCP connection to send request to client behind NAT
| Status | Closed |
| Id | 104499 |
Re-use inbound TCP connection in case of NAT.
Re-use inbound TCP connection to send request to client.
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.
PBX-SOAP: UserFindDestination did not take node, but pbx for start of the search
| Status | Closed |
| Id | 104524 |
The Fax Server uses this function to find users
Gateway: Potential trap with transfer of calls to busy destination, sending Name Id
| Status | Closed |
| Id | 104527 |
This happened when the Fax Server tried to send a fax to a busy phone.
SIP: Must add 'received' parameter to topmost Via header when sending responses
| Status | Closed |
| Id | 104531 |
Must add 'received' parameter to topmost Via header when sending responses.
This helps to detect NAT situation on client side.
SIP: Keep TCP connection open if connected to SIP server throuch NAT
| Status | Closed |
| Id | 104534 |
Must keep TCP connection open if connected to SIP server throuch NAT,
other wise SIP server cannot send requests to client.
PBX Multicast: Chat to Multicast object should be rejected
| Status | Closed |
| Id | 104554 |
Instead a call to the members of the multicast group was initiated
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.
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.
myPBX: Numbers decorated with slashes could not be dialed
| Status | Closed |
| Id | 104695 |
Numbers containing slashes were treated as URIs and dialed in a wrong way.
NAT: UDP Port forwarding with port mapping did not work
| Status | Closed |
| Id | 104702 |
Inbound packets were forwarded to the configured host but in the packets sent back by this host the source port was not mapped back.
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.
SIP: Handling of dialog-info marked as 'full' was wrong
| Status | Closed |
| Id | 104759 |
A dialog-info marked as 'full' with no active dialogs
must clear all calls from partner fkey.
softwarephone v10 does not increment license count on V9 pbx
| Status | Closed |
| Id | 104829 |
An attached softwarephone v10 at a v9 pbx did not increase the consumed softwarephones license count
IP232,IP222, IP241: Fkeys may overlap call control
| Status | Closed |
| Id | 104886 |
Fkeys overlap call control in case of two inbound ringing calls.
avoid ip29 switching to spread-spectrum clock
| Status | Closed |
| Id | 104890 |
ip29 has the spread-spectrum clock unwired since HW200
SIP: Do not throuch an error if DNS query fails in case of 'closed federation'
| Status | Closed |
| Id | 104928 |
Do not throuch an error if DNS query fails in case of 'closed federation'.
Because DNS will fail on most queries due to recursion-desired==false.
SIP: Call was dropped after successful session refresh
| Status | Closed |
| Id | 104942 |
Call was dropped after successful session refresh.
Handling of 200/OK for UPDATE was wrong.
Was wrong since bug fix #103996 (v10sr1)
truncated NBNS NODE STATUS RESPONSE sent
| Status | Closed |
| Id | 104949 |
not critical because only irrelevant zeros at end of packet were missing, but wireshark complained
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
myPBX: Updated translations
| Status | Closed |
| Id | 105040 |
- Improved texts
- Fixed placeholders
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.
SIP: Interop problems when interworking t38 capability indication to H.323
| Status | Closed |
| Id | 105097 |
Interop problems when inteworking t38 capability indication to H.323.
E.g. m=image 0 udptl t38
Was taken as an offer with port 0.
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: Return "489 Bad Event" if dialog subscription was rejected by application
| Status | Closed |
| Id | 105557 |
Return "489 Bad Event" if dialog subscription was rejected by application.
Not just "603 Decline".
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.
Phones: Added Hungarian language
| Status | Closed |
| Id | 104244 |
Added Hungarian as another language.
PBX CSV Import, support for passwords and Groups
| Status | Closed |
| Id | 104637 |
.
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.
SIP: New config option "Local Domain" for federation interfaces
| Status | Closed |
| Id | 105186 |
New config option "Local Domain" for federation interfaces.
SIP: New config option "Local Port" for federation interfaces
| Status | Closed |
| Id | 105215 |
New config option "Local Port" for federation interfaces.
To configure multiple federation interfaces on different ports.
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.
SIP: Changed VOIP signaling options at runtime
| Status | Closed |
| Id | 105453 |
Changed VOIP signaling options at runtime
V10 Service Release 3 (100918)
Changes included in Version 10 Service Release 3 Definition
HTTP client: allow chunk definition to cross packet boundary
| Status | Closed |
| Id | 103765 |
HTTP chunked transfer encoding did not work if chunks did not arrive in single SOCKET_RECV_RESULT events.
phone: indicate only not-registered/registered on phone display, don't discriminate between primary and alternate gatekeeper
| Status | Closed |
| Id | 103816 |
Only in rather simple configurations the primary and the alternate gatekeeper can be clearly distinguished. In more complex PBX redundancy configurations it may be not possible to find out the role of the gatekeeper a phone is registered to. Thus it's better to leave off this information because it's not really important for the average phone user.
myPBX Outlook integration uses primary PBX email address too
| Status | Closed |
| Id | 103847 |
Instead of just using the SIP address h323@domain the primary PBX email address is now also used for contact identification.
H.323: Payload Type received with an SDP Answer was not forwarded in H.323
| Status | Closed |
| Id | 104552 |
Instead the payload type originally sent in the offer was used. This caused no or one way media with some third party video endpoints.
myPBX: Avoid hanging presence-requests
| Status | Closed |
| Id | 105712 |
One-time presence requests across PBXes did not terminate if the subscription call was established but the remote party did not send any presence info. Now there is a timeout of 200ms (after the subscription call has been established).
Video: display artifacts due to cropping field in the h264 stream
| Status | Closed |
| Id | 105730 |
Wrong handling of the cropping field in the h264 stream. Visible when doing Video with Polycom and IPAD Mini or Galaxy 7.0.
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.
PBX Waiting: A call to Waiting could not be canceled by myPBX or SOAP
| Status | Closed |
| Id | 105879 |
If the call was alerting at the WQ, nothing happned, when trying to cancel the call. If the call was connected to an announcement, the next announcement was played.
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
myPBX: Remove X-Button from input fields in windows 8
| Status | Closed |
| Id | 106057 |
The Windows 8 specific input button causes problems in the search field of myPBX. Therefore it is removed from the input fields in myPBX.
myPBX: Visibility settings did not show allows from all templates
| Status | Closed |
| Id | 106106 |
If the user had multiple templates, not all allows were shown in the visibility settings in myPBX.
This did not have an influence on the effective visibility.
SIP: Allow blind transfer to unknown destination
| Status | Closed |
| Id | 106126 |
Allow blind transfer to unknown destination.
Results in REFER with "Refer-To: <sip:domain.com>".
No userpart in Refer-To-URI.
PBX Mobility; Wrong Calling Party Number when initiating calls with myPBX together with Nodes/Escapes
| Status | Closed |
| Id | 106136 |
Unnecessary escapes where added
PBX Boolean: When monitoring the boolean, state update only with delay
| Status | Closed |
| Id | 106169 |
A Boolean object can be monitored to obtain the current state. When the state was manually updated by a call to the boolean object or by the administrator, this update was delayed up to 10s.
IP-DECT: PBX-registration update with changed authorisation name
| Status | Closed |
| Id | 106205 |
The registration to the PBX is not updated if the user's authorisation name is changed. This is fixed now.
103433: JKI-SWPhone-V10: "User Configuration" not accessible
| Status | Closed |
| Id | 106212 |
Due to a problem with changing local @ the configuration url, based on actual @es could not be accessed. Fix using localhost instead of @
103997: Not registered in tray icon
| Status | Closed |
| Id | 106213 |
When clicking on the tray item it showed "Not registered" when it actually were.
104935: Softphone does not start if swphone_commands.cfg > 50k
| Status | Closed |
| Id | 106219 |
if swphone_commands.cfg > 50 softwarephone crashed on start.
PBX: Call Filter were not checked for <number>@domain calls
| Status | Closed |
| Id | 106229 |
This was a security issue
IP222,IP232,IP241: Disabled debug traces
| Status | Closed |
| Id | 106243 |
Disabled debug traces which have been active since v10sr2.
myPBX: Pickup did not work correctly with multiple parked calls
| Status | Closed |
| Id | 106245 |
Always the first in the list was picked. After that the other calls could not be picked at all.
The myPBX parking logic is also changed. Calls are always parked on position 0 and picked from any parking position (-1).
Voicemail Objekt: Trap During Reconfiguration
| Status | Closed |
| Id | 106274 |
Wasn't reproducable. Added a counter-measure against a suspected scenario.
PBX: myPBX and Partner key presence/dialog subscriptions and IM did not work accross PBX in e.164 setup
| Status | Closed |
| Id | 106372 |
The routing of calls for these services by number did not work the same way as voice calls. Some mechanisms needed in e.164 setups were missing.
SIP: Do not generate an ERROR log on each and every negative DNS request
| Status | Closed |
| Id | 106377 |
Set a single ALARM in case of 'local error'.
QSIG: Answering a Facility=ctInitiate.invoke with a Facility=ctInitiate.error
| Status | Closed |
| Id | 106436 |
By replying with ctInitiate.error, where error will be notAvailable(3), the remote PINX is requested to switch from the procedure "Transfer-By-Rerouting" to the procedure "Transfer-By-Join".
PBX: Dialog Info for subscribtions accross PBX's/Nodes sometimes wrong
| Status | Closed |
| Id | 106447 |
Adjustments did not work correctly with Nodes including escapes
myPBX: Abort queued commands after a timeout
| Status | Closed |
| Id | 106489 |
In some cases the browser queues HTTP requests for a long time before they are sent to the PBX. For example this happens when there maximum number of connections to the server is reached. The delay might be very long, like some minutes.
In this case myPBX should abort all further queued command requests instead of sending them after a long delay.
PBX: Visibility definitions did not work as expected
| Status | Closed |
| Id | 106507 |
A Definition for a name should overrule a definition for a group and a definition for a group should overrule a definition for a domain.
PBX: The 'Twin Phone' option was set on users imported from a csv file
| Status | Closed |
| Id | 106529 |
The default for busy-out was set wrong when creating users on csv import. This resulted in the 'Twin Phone' checkmark to be set. The default was fixed so that the checkmark is not set anymore.
SIP: Give subscriptions time to terminate before unregistering
| Status | Closed |
| Id | 106581 |
Give outbuond subscriptions time to terminate before unregistering from SIP server.
IP6000 IP2000: Crypto driver stopped working after receiving bad SRTP packets
| Status | Closed |
| Id | 106681 |
Better protection against receiving non-SRTP packets.
SIP: Do not increase SDP version when answering session refresh
| Status | Closed |
| Id | 106703 |
Do not increase SDP version in when answering re-INVITE for session refresh.
If SDP offer in re-INVITE did not increase it's version,
SDP answer should keep it's SDP version also.
myPBX: Support for additional mobile browsers
| Status | Closed |
| Id | 106711 |
- Mobile Safari is also detected as a mobile browser.
* The touch version can now be turned on or off using an URL parameter (touch=true, touch=falase)
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: 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.
104086: V10 softphone and call recoridng
| Status | Closed |
| Id | 107074 |
When call recording was enabled the Audio stream only went to the recording device. Call participants could not hear each other
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)
Logging: Do not log "Excessive loss of Data" more than once for one call
| Status | Closed |
| Id | 107108 |
Do not log "Excessive loss of Data" more than once for one call
myPBX: Map all presence activites to one of the activity icons
| Status | Closed |
| Id | 107110 |
Presence activities that do not have an icon in myPBX were wrongly mapped to "available". Now the mapping is like that:
available: available, looking-for-work
away: away, in-transit, permanent-absence, shopping, sleeping, travel
busy: appointment, meeting, performance, playing, presentation, spectator, steering, tv, working, worship
lunch: breakfast, dinner, lunch, meal
vacation: holiday, vacation
on-the-phone: on-the-phone
other: unknown, ...
| Status | Closed |
| Id | 107199 |
Use "mouseout" instead of "mouseleave".
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.
SIP: Insufficient buffer space for request construction
| Status | Closed |
| Id | 107529 |
Some SIP proxies bloat SIP messages by adding countless number of Route headers to a dialog (e.g. Avaya).
Need to increase buffer space to hold all the very important Route headers in REFER, UPDATE and BYE.
PBX UI: Changing of object type
| Status | Closed |
| Id | 107685 |
This was implemented in v10, but was broken later on
Gateway: Potential Trap on collision of call clearing and transfer
| Status | Closed |
| Id | 107686 |
A trap could happen if a call was transfered by the remote side and cleared locally at the same time.
SIP: Changed registration refresh interval
| Status | Closed |
| Id | 107716 |
Changed registration refresh interval to TTL-32sec.
Where TTL is the server provided time-to-live.
And 32secs is the maximum life-time of a REGISTER transaction.
phone: importing a phonebook may result in memory leaks
| Status | Closed |
| Id | 107760 |
happens when phonebook entries containing non UTF8 characters are deleted
dect_comcerto: improved check for missing ethernet interrupts
| Status | Closed |
| Id | 107772 |
-
myPBX: Support for Safari private browsing
| Status | Closed |
| Id | 107788 |
Accessing DOM storage threw an exception if Safari was in private browsing mode.
PBX: Potential Trap during call connect
| Status | Closed |
| Id | 107792 |
null pointer access
107810: sofwarephone name
| Status | Closed |
| Id | 107846 |
softwarephone product naming fixed
Fixed receiving of larger HTTP responses inside directory esarch and linux module
| Status | Closed |
| Id | 107884 |
Directory Search HTTP connections and the Linux check might have failed if larger HTTP responses were received.
H.323: Alternate RAS port only worked until the first time a registration failed
| Status | Closed |
| Id | 107886 |
Then the port switched back to the default
IP232,IP222,IP241: Overlapping of local name and number on call display
| Status | Closed |
| Id | 107899 |
Local name and number may interfere on first line of call control.
But only if local name is too long.
Faxserver: Better diagnostics if webdav read/write fails
| Status | Closed |
| Id | 107918 |
This problem could only be found by looking at a wireshark trace
TLS: Fragmentation did not work properly
| Status | Closed |
| Id | 107956 |
Sent data has to be fragmented to records that are smaller than 16k.
IP232,IP222,IP241: Display information of pickup fkey truncated too much
| Status | Closed |
| Id | 107962 |
Display information of pickup fkey truncated too much
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: dont' log calls in call lists if source/destination is marked as HIDE in local directory
| Status | Closed |
| Id | 103574 |
V 10.10 and higher:
A source/destination is marked as HIDE in the 'flags' attribute of a directory entry.
the 'flags' attribute can be added/modified in a downloaded directory in CSV format and then uploaded again, for example by changing
"broadcast",,,"203"
to
"broadcast",,,"203",,,"FLAG_HIDE"
If the 'flags' field contains already some FLAG_... values the new value must be appended with a preceeding space character, i.e. "FLAG_IMP" is changed to "FLAG_IMP FLAG_HIDE".
The flag attribute can also be added/modified in the directory entries in a downloaded config file, for example by changing
mod cmd FLASHDIR0 add-item 102 (cn=broadcast)(e164=203)(guid;bin=662579A0E909D311AD850090332A0094)(usn=3)
to
mod cmd FLASHDIR0 add-item 102 (cn=broadcast)(e164=203)(flags=4)(guid;bin=662579A0E909D311AD850090332A0094)(usn=3)
The 'HIDE' flag has the value 4. If there is already a flags attribute the 'HIDE' value must be 'ored' to the existing value, i.e. flags=1 is changed to flags=5.
The currently defined flags are:
FLAG_IMP 1 -- entry was created by an 'import' type directory upload
FLAG_IP 2 -- interpret h323 alias as IP address when dialing
FLAG_HIDE 4 -- don't log call from/to this target
myPBX: Send own email address with connect message
| Status | Closed |
| Id | 103812 |
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).
Use '@' as name for visibility definition to specifiy default visibility for any foreign domain
| Status | Closed |
| Id | 105750 |
To avoid having many visibility definitions for each user
PBX: Support for inconsistent Nodes like swiss area codes in e.164 scenarios
| Status | Closed |
| Id | 105997 |
In switzerland area codes exist, with an escape of 0, but it is not possible to call within the area code without dialing 0+area code. For this reason users do not expect to see calling line ids without the area code, because this is something which cannot be dialed in the public network.
There is now a checkmark at the node configurfation which allows the generation of this kind of calling id.
Additional MSI parameters for myPBX launcher
| Status | Closed |
| Id | 106124 |
New MSI parameters:
SHOWINTASKBAR="[true|false]"
STARTMINIMIZED="[true|false]"
DOCKING="[none|left|right]"
HOTKEY="{number}"
HOTKEYMOD="{number}"
HOTKEYACTION="[copy|show]"
VIDEO="[true|false]"
VIDEOACTIVE="[true|false]"
VIDEOPROXY="{host:port}"
NOTIFICATIONS="[true|false]"
SOUNDS="[true|false]"
LANG="{two-letter-code}"
Changed MSI parameters:
AUTOAPPEAROFFLINE="[true|false|{minutes}]"
PBX Waiting: Allow to disable mobility for operators
| Status | Closed |
| Id | 106238 |
This way the behaviour of v9 can be achieved
"DELETE" Assertion traces caller
| Status | Closed |
| Id | 106293 |
For debugging purposes
PBX-SOAP: Provide 'Send Number' in UserInfo
| Status | Closed |
| Id | 106456 |
The configured 'Send Number' is provided in the UserInfo record
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"!
Phones: Play 'fast-busy' tone in case of network errors
| Status | Closed |
| Id | 106625 |
Play 'fast-busy' tone in case of network errors
to distinguish network errors from 'user busy' condition.
IP-DECT: Refresh RAS registration if behind NAT
| Status | Closed |
| Id | 106634 |
The DECT base station does not refresh the RAS registration after a reregistration. Incoming calls are not possible if the base station is behind a NAT. This is fixed now.
104537: Softwarephone: Type-of-Service (TOS) bit values for RTP and Signalling packets
| Status | Closed |
| Id | 107182 |
During Installation it is checked if a TOS value for softwarephone is set, if not a Default value of 46 is set
introduce 'ready' flag in xml-info
| Status | Closed |
| Id | 107744 |
this flag is neccessary to mark the end of the initialisation routine especially for ip28, which requires a long calibration phase after config reset.
PBX: Send presence in alert message only if explicitly enabled
| Status | Closed |
| Id | 107791 |
Not everybody wants this feature because of privacy reasons
myPBX dial: Support phone: URIs
| Status | Closed |
| Id | 107863 |
Support additional URI: phone
V10 Service Release 4 (100933)
Changes included in Version 10 Service Release 4 Definition
SIP: Tell the SIP endpoint to stop sending media while remote party has been put on HOLD by myPBX
| Status | Closed |
| Id | 106693 |
Tell the SIP endpoint to stop sending media while remote party has been put on HOLD by myPBX.
PBX: A diverted call should allow more dialing digits to be sent if original call is still in overlap dialing
| Status | Closed |
| Id | 107770 |
A diverted call was always sent as sending complete, but sometimes with an implizit diversion, e.g. by a Map object, more dialing digits should be possible to send after the diversion
PBX: Potential hanging failed dialog/presence subscription accross PBX's
| Status | Closed |
| Id | 107901 |
If a dialog/presence subscription from master to slave or slave to master failed because the other PBX was temporarily not available, it could happenm that this subscription was hanging in this failed state. It only worked again by terminating and restarting it (e.g. by restarting myPBX)
myPBX: Can't override calendar presence with available
| Status | Closed |
| Id | 107934 |
If there was a calendar: presence and no tel: presence the user could not overwrite the calendar: presence with tel:avaiable.
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.
myPBX: Could not confirm visibility request for users with spaces
| Status | Closed |
| Id | 108234 |
It was not possible to confirm visibility requests for users ther had an URL that contained spaces or other special characters.
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.
Phones: Handling of a failed re-route was wrong
| Status | Closed |
| Id | 108327 |
If INVITE was redirected to a destination not available a spooky REFER(sipfrag) was sent.
SIP: Subscription was not re-newed sometimes
| Status | Closed |
| Id | 108423 |
Bug when receiving NOTIFY while SUBSCRIBE transaction is pending.
Scenario:
1. Sending SUBSCRIBE to refresh subscription
2. Receiving NOTIFY with "Subscription-State: terminated;reason=timeout"
3. Receiving response for SUBSCRIBE
Response for SUBSCRIBE was not handled.
Subscription was not re-established, because previous SUBSCRIBE transaction was still pending from subscription's point of view.
PBX: myPBX did not work for standby case
| Status | Closed |
| Id | 108518 |
The license check failed when a user wanted to work with myPBX on a PBX for which he was not configured
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: Visibility configuration for a domain did not work anymore
| Status | Closed |
| Id | 108569 |
A visibility setting of the form @<domain-name> was ignored
myPBX: IE8 script error when using drop-down boxes
| Status | Closed |
| Id | 108583 |
Collateral damage from #107199: myPBX: Drop-down menus not closed automatically in Chrome.
event.relatedTarget does not work in IE8. Using event.toElement, instead.
myPBX: Hide chat button for offline users in directory search
| Status | Closed |
| Id | 108762 |
If a user is offline, no chat button should be displayed.
myPBX: Improved TLS tracing
| Status | Closed |
| Id | 108841 |
"TLS Plaintext" tracing now uses local port 8 instead of the actual port. This helps wireshark to distinguish between the actual packets from "All TCP/UDP" and the fake "TLS Plaintext" packets.
SIP: Memleaks at collision of incoming call and sig_event_listen_cancel
| Status | Closed |
| Id | 108858 |
Memleaks at collision of incoming call and sig_event_listen_cancel.
myPBX: Offer pickup only at internal favourites
| Status | Closed |
| Id | 108875 |
Do not offer the pickup button at favourites that are an external SIP URI or phone number.
PBX SOAP: Invalid SOAP coding if 'Send Number' configured at a user results in TAPI not working
| Status | Closed |
| Id | 108940 |
Some applications ignore the invalid coding, some for example TAPI don't, so if a Send Number is configured at a PBX object, TAPI does not work.
This was a collateral damage of
fix #106456: PBX-SOAP: Provide 'Send Number' in UserInfo
for SR3
PBX: When receiving a dialog/presence subscription from other domain the own id was sent without domain
| Status | Closed |
| Id | 108950 |
If the subscription was sent by myPBX because a favorite was to be added, a wrong uri was put into the favorite, but only if H.323 was used for the subscription end to end.
myPBX: Use object filter from LDAP configuration
| Status | Closed |
| Id | 109036 |
Before the fix the object filter was ignored by myPBX.
Fax: Private User-User Information caused protocol error on ISDN
| Status | Closed |
| Id | 109068 |
By accident user-user-info, which was used for the communication of the FAX interface and the FAX Server were forwarded on ISDN. They caused a protocol error which eventually disconnected the call.
myPBX: Ignore otherTelephoneNumber in external directory
| Status | Closed |
| Id | 109082 |
The special meaning of the attribute in the PBX directory should not be applied to the external directory.
myPBX: Accept non-international numbers from LDAP
| Status | Closed |
| Id | 109166 |
For name resolution for incoming phone calls, myPBX only accepted international numbers like +497031730090. Now dialable numbers like 007031730090 also work.
This is needed for the trick of having a phone book inside the PBX.
add_view_record - ambigous views
| Status | Closed |
| Id | 109171 |
Fixed a problem with the phone directories. If a new view had been added via cmd add-view the next_id variable was not set above the ID of this view. If the system allocated new views afterwards their ID clashed with the existing ones. On Android this happened very likely because we feed the start configuration via cmd add-view and cmd add-item and if a directory entry was added afterwards it caused the error message
add_view_record - ambigous views id 101/101 name cn=call-list-0/cn=phone-dir rec 294380/0
Now we adjust next_id to at least the id added and also set it back to 100 if all views are deleted.
SIP: Reject new re-INVITE with 491 if previous re-INVITE transaction is not complete yet
| Status | Closed |
| Id | 109235 |
Reject new re-INVITE with 491 if previous re-INVITE transaction is not complete yet.
Instead of rejecting with 488.
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.
PBX: Presence subscription to a Map object (e.g. by configuring a myPBX contact) caused a call being sent to Map destination
| Status | Closed |
| Id | 109307 |
It was tryed to retrieve the presence from the Map destination instead of the Map object itself.
This was a collateral damage of
fix: #107682: PBX Map: Overlap dial thru a Map Object on Slave with a call via the Master did not work
myPBX launcher: Config dialog crash when no language is selected
| Status | Closed |
| Id | 109334 |
The problem occured if the windows language is not available in myPBX. In this case no item is selected in the language drop-down. When saving, myPBX crashed with a NullPointer Exception.
Problem with hold/retrieve, not hearing original party
| Status | Closed |
| Id | 109450 |
Problem:
- call exists between swphone and another Party.
- swphone places call on hold
- swphone calls another Party
- Party does not answer-
- swphone terminates this call
now swphone still hears ringback tone and not the original Party
pcap/tracing shows only a tiny fragment of debug outputs
| Status | Closed |
| Id | 109451 |
The traces written to the trace buffer for tracing/pcap covered only a tiny Fragment of the available debugs.
Softwarephone: Mute ringer is not saved/does not work
| Status | Closed |
| Id | 109453 |
Mute Ringer in the tray Icon does not work. Now it works for the Duration of swphone running.
PBX Waiting: 'No mobility for Operators' did not work for calls with Name Id
| Status | Closed |
| Id | 109471 |
Wrong handling of facilities in the Waiting Queue caused this strange dependency
SIP: Interworking issue with OpenStage systems
| Status | Closed |
| Id | 106761 |
Interworking issue with OpenStage systems.
Pass proprietary signaling options to phoneapp.
Updated translations
| Status | Closed |
| Id | 108349 |
Updated translations
PBX: Make Node/PBX at Config Template configurable
| Status | Closed |
| Id | 108506 |
For management of administration rights
PBX: Make VoicemailUser license configurable at WQ and other non-user objects
| Status | Closed |
| Id | 108507 |
To allow the use of the new VoicemailUser licenses with these object types
IP-DECT: Remote control connect
| Status | Closed |
| Id | 108770 |
With a remote control connect the handset goes off-hook. myPBX uses remote controls and now it is possible to accept an incoming call for a DECT handset with myPBX. This only works with IP61 and IP63 after firmware update.
when a problem occurs before the programm comes up there is not method of tracing
| Status | Closed |
| Id | 109452 |
allow to additionally write traces into a file. this comes in handy when a problem occurs before the logging via webbrowser is accessible or when a rather large amount of tracing info is required
V10 Service Release 5 (100958)
Changes included in Version 10 Service Release 5 Definition
SIP: Memory leak when "Group Indications" are activated on PBX user object
| Status | Closed |
| Id | 107967 |
Memory leak when "Group Indications" are activated on PBX user object.
But only if a SIP device is registered.
Media channel diagnostics
| Status | Closed |
| Id | 108639 |
Added some traces to support debugging of media channel issues.
PBX: Registration page sometimes broken for outgoing registrations (SBC)
| Status | Closed |
| Id | 109140 |
Invalid characters were put in the XML information
SIP: Hold/retrieve of a second SRTP call causes white noise when separate AVP and SAVP is used
| Status | Closed |
| Id | 109263 |
Processing of alternative media descriptions (AVP and SAVP) was buggy.
Resulted in heavy noise after hold/retrieve.
Prevent trap on certain wireshark rpcap connections
| Status | Closed |
| Id | 109356 |
Certain wireshark message caused a box to trap.
SIP: Memory leak when hold/retrieve a call to waiting
| Status | Closed |
| Id | 109439 |
Memory leak when hold/retrieve a call to waiting.
Call and channel object is not deleted on PBX after call end.
LDAP Replication: Mutual Coexistance for LDAP/AD-Replicator Instance and PBX/Replication Instance
| Status | Closed |
| Id | 109461 |
So is for: Mutual Coexistance for LDAP/AD-Replicator Instance and DECT(Mirroring)/Replication Instance
Either of both replicator instances may be active at a time. The LDAP/AD-Replicator instance has internally a low priority.
If another high-prio replicator instance comes to live, the LDAP/AD-Replicator instance will be auto-disabled. If the other high-prio replicator instance disappears, the LDAP/AD-Replicator instance will be auto-started.
Voicemail: Resume from Suspend Not Working During <exec> Statement
| Status | Closed |
| Id | 109485 |
An <exec url="mailto:..."> statement was executed more than one time
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
myPBX: Do not hide phone actions for favourites without registration
| Status | Closed |
| Id | 109674 |
The favourites might still be reached via forking or call diversion.
PBX Waiting: A forking to waiting object did prohibit a CFNR execution
| Status | Closed |
| Id | 109683 |
A mechnanism implement to prohibit the execution of a CFNR at a Waiting Queue operator had this unwanted side effect.
myPBX: Number resolution not working without dialing location configured
| Status | Closed |
| Id | 109731 |
The internal PBX phonebook did not work if no dialing location was configured.
Change of only the password of the first registration didn't work
| Status | Closed |
| Id | 109797 |
If only the password of the first registration was changed the phone didn't initiate a reset and cleared out the password UI field.
myPBX: Dialing numbers with decoration did not work correctly
| Status | Closed |
| Id | 109802 |
Dialing number with decoration could lead to additional digits being dialed.
Example: +49(7031)73009-987 dialed 987987
wrong G711 conversion table
| Status | Closed |
| Id | 109820 |
Currently used only in IP800 conference.
SIP: In-dialog request are sent to wrong destination port
| Status | Closed |
| Id | 109895 |
In-dialog request (such as BYE) are sent to wrong destination port,
if remote Contact-URI contains domain-name as hostpart and
if remote peer runs on non-default port 5060.
Support start of call inside an Office 2013 contact
| Status | Closed |
| Id | 109934 |
There was a small fix needed to be able to start a call from an Office 2013 contact.
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.
IP-DECT/Analog Features: Pick-up with myPBX
| Status | Closed |
| Id | 110049 |
Pick-up a call with myPBX is not possible if features are enabled. 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.
SIP: Trap in SIP stack when incoming call is rejected
| Status | Closed |
| Id | 110753 |
Trap in SIP stack when incoming call is rejected.
SIP: One-way media after re-negotiation (collateral damage of #106693)
| Status | Closed |
| Id | 110830 |
One-way media after re-negotiation.
Collateral damage of #106693: SIP: Tell the SIP endpoint to stop sending media while remote party has been put on HOLD by myPBX
myPBX: Pickup accross PBX failed and could cause Trap
| Status | Closed |
| Id | 110862 |
Monitoring a user on a remote PBX worked, but pickup of a call failed
myPBX: Calls rejected with myPBX did not appear in call list
| Status | Closed |
| Id | 110922 |
These calls were handled as if they were accepted somewhere else
myPBX: Remote media not working on secondary URL
| Status | Closed |
| Id | 110925 |
For remote media always the primary URL was used.
Faxserver: Better diagnostics if outgoing call fails
| Status | Closed |
| Id | 110927 |
The cause code for the failed call is recorded
PBX SOAP: Initiate outgoing call for a Gateway object with Max Calls=1 was not possible
| Status | Closed |
| Id | 110928 |
The dummy call sent to the local registration first, was counted as call as well.
myPBX: Classify URIs containing spaces as internal favourites
| Status | Closed |
| Id | 110963 |
When adding a URI like "John Doe HQ" as a favourite, it was classified as an external favourite, even if it is an internal PBX user.
Note: It is recommended to use short names in the PBX that give valid SIP URIs.
PBX Waiting: A presence without activity should not disable Operator
| Status | Closed |
| Id | 110967 |
The feature 'Presence disables Operator' allows to disable an operator of an waiting queue if the operator has set a presence. This should not happen for a presence without activity but only a note. The Exchange Connector does this for upcoming dates.
Contact without further email address might be wrongly resolved
| Status | Closed |
| Id | 111027 |
Wrong contact resolving on new contact on certain conditions.
IP-DECT: Handset display update for parked calls
| Status | Closed |
| Id | 111032 |
The handset display update for parked calls after hung-up is fixed.
IP-DECT: Wrong error 'AC missing' with user delete
| Status | Closed |
| Id | 111038 |
If an user is deleted, the wrong error 'AC missing' can occur. This is fixed now.
PBX: If a calendar presence was overwritten by the user a subseqent calendar presence preview was not shown
| Status | Closed |
| Id | 111043 |
The calendar presence had to be deleted once for this mechanism to work again
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.
Translation updates
| Status | Closed |
| Id | 111269 |
Fixed some date formats.
<--
pbx_client_localisation.cpp
-->
DHCP: A server with "Reserved and same Vendor Clients only" checked did not provide leases to IP62 phones
| Status | Closed |
| Id | 111276 |
Video: RTP sequence not set properly for 3rd party conference
| Status | Closed |
| Id | 111297 |
The participant mixing video streams sends two different RTP streams but I was using just the same variable for both streams.
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.
PBX: Don't accept calls, when the PBX is about to stop
| Status | Closed |
| Id | 111471 |
This could cause a trap when receiving calls right when the PBX is switched off
Quickdial: Configured Display Attribute Duplicated On "Apply"
| Status | Closed |
| Id | 111492 |
Linux: Disable feature
| Status | Closed |
| Id | 111515 |
It can occur that the Linux cannot be disabled. This is fixed now.
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.
myPBX launcher: Remove additional spaces from configured URI
| Status | Closed |
| Id | 111568 |
If the URL had trailing spaces, loading the web application did not work. Now additional spaces are removed when the configuration is saved.
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.
Call Recording for incoming calls did not work properly
| Status | Closed |
| Id | 111621 |
When configuring softwarephone for call recording, incoming calls got recorded but there was no Audio stream to the caller
crash during startup after some period of working properly
| Status | Closed |
| Id | 111623 |
After a while of working properly softwarephone crashed at Startup, only reinstaling solved the Problem temporaryly
Change the filesizes to dynamic instead of static to avoid filebuffer overflows when swphone_commands.cfg increases
| Status | Closed |
| Id | 111624 |
When the filesize of swphone_commands.cfg increased beyond the fixed size (50k) due to storing of the call history softwarephone crashed during startup
Ringer settings in phone/preferences (melody,volume, duration) did not work
| Status | Closed |
| Id | 111631 |
These Settings were ignored
when the http port was is not configured the configuration url does not work
| Status | Closed |
| Id | 111632 |
After configuration changes to the http port the URL to the configuration did not work anymore, thus making configuration Access via the start menu entry not possible nay more
SIP: Trap on media negotiation
| Status | Closed |
| Id | 111651 |
When performing coder selection on an offer that contains separated audio descriptions for un-encrypted and encrypted audio.
phone: under soap control no audio data was sent when a call was retrieved after another call has been transferred
| Status | Closed |
| Id | 111662 |
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.
myPBX launcher: Expand environment variables in path and parameters for external application
| Status | Closed |
| Id | 109456 |
Environment variables like %ProgramFiles% are needed for deployment on different platforms.
Support client timezone for call list in myPBX
| Status | Closed |
| Id | 110032 |
myPBX shows times in the client timezone now.
phone: ip150: changed handset speaker parameters for hardware 102/602
| Status | Closed |
| Id | 110048 |
SIP: Provide physical PBX location to endpoint when redirecting REGISTER
| Status | Closed |
| Id | 110712 |
Provide physical PBX location to SIP endpoint when redirecting REGISTER.
Provide as URI parameter in Contact-URI like this:
\tSIP/2.0 301 Moved Permanently
\tVia: SIP/2.0/UDP 172.16.16.217:5060;branch=z9hG4bK-B318A5EB;rport
\tFrom: ;epid=0090331010bc;tag=731526080
\tTo: <sip:pluto@Orion>;tag=2507679873
\tCall-ID: 7f457e8ae909d31184310090331010bc@172.16.16.217
\tCSeq: 1000 REGISTER
\tContact: <sip:pluto@Orion;phys=slave1@Orion;maddr=172.16.16.210>
\tContent-Length: 0
\tServer: (innovaphone IP6000/11.00 dvl [10.XXXX/100203/107])
SIP: New config file option /nat-keepalive-interval
| Status | Closed |
| Id | 110747 |
New config file option /nat-keepalive-interval to adjust (or disable) NAT ping interval (keep-alive packets).
Default interval is 30 seconds now (was 10 seconds before).
E.g.
config change SIP /nat-keepalive-interval 10
Fax server: Switching to T.38 by the caller
| Status | Closed |
| Id | 110895 |
Some SIP provider don't switch to T.38. Now the FAX interface switches also to T.38 in the calling mode after timeout.
FXO: Support for calling line Id
| Status | Closed |
| Id | 111077 |
Calling line id is received from an FXO interface after the first ring, so an enblock route must be used to delay the signaling of the call until the calling line id is recived
Voicemail: "$_pbxfwdrel=conn" releases session after successful <pbx-fwd>
| Status | Closed |
| Id | 111143 |
This new URL query string variable allows to release the interpreter session right after a forward destination(<pbx-fwd>) accepted a call.
see http://wiki.innovaphone.com/index.php?title=Howto:Configure_the_innovaphone_Voicemail#URL_Query_String_Variables
Fax server: Configurable modem class
| Status | Closed |
| Id | 111407 |
Now it is possible to configure the supported modem class of the fax server interface. There are available:
- V.17, V.29 and V.27
- V.29 and V.27
- V.27
installer package was not signed and did not have a build version number
| Status | Closed |
| Id | 111622 |
The package is signed now and the build number is set as Version number
Bind to port 0 to avoid conflicts with other modules listening on port 1720
| Status | Closed |
| Id | 111625 |
In order to avoid conflicts with other Software listening on port 1720 now a bind is made to dyn port 0
simple method required to gather all neccessary debug information
| Status | Closed |
| Id | 111633 |
Collecting all the necessary info (config, registry entries, traces..) should be simple. A command is now provided to gather all this and zip it up to debug-info.zip which can then be attached to the mantis report
V10 Service Release 6 (100970)
Changes included in Version 10 Service Release 6 Definition
myPBX launcher: Remove hash part from URLs in trace files
| Status | Closed |
| Id | 103818 |
The hash part might contain sensitive information that is unwanted in trace files.
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.
Admin UI: Add cancel button to join and leave realm dialogs
| Status | Closed |
| Id | 110687 |
A cancel button was added to the join realm and leave realm dialogs.
Phones: Pickup list sometimes contains doublets
| Status | Closed |
| Id | 111725 |
Same call could be is listed more than once.
myPBX: EP requests returned "found" after timeouts
| Status | Closed |
| Id | 111772 |
Requests better should return "not found" if there was a timeout.
A configured device name with 16 or more non-ascii character, could break the user interface
| Status | Closed |
| Id | 112022 |
The resulting string contained 32 or more bytes, which caused a buffer overrun.
PBX: User admin rights were lost after XML Export/Import
| Status | Closed |
| Id | 112071 |
The "admin" attriute was not written back to the PBX object.
| 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
SIP: Trap in rare case with interrupted media negotiations
| Status | Closed |
| Id | 112210 |
Trap in rare case with interrupted media negotiations.
Interruped by other media negotiation.
HTTP-Client: Trap if debug tracing is on
| Status | Closed |
| Id | 112236 |
Trap if debug tracing is enabled.
Null pointer trap.
Gateway: Potential Trap on incoming calls on FXO interface
| Status | Closed |
| Id | 112277 |
Null pointer access
LDAP: Name resolution did not work for local numbers
| Status | Closed |
| Id | 112320 |
The customer did not entry the phone number im LDAP Server including area code for local numbers.
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.
hook-flash event not sent to peer when 'passive' mode is activated
| Status | Closed |
| Id | 112370 |
hook-flash events have to be transmitted to peer side when the FXS is configured to 'passive'.
myPBX launcher: Lag on minimize window
| Status | Closed |
| Id | 112442 |
The auto appear offline feature caused a lag when minimizing the window. Use raw input API instead of global mouse and keyboard hooks in order to prevent that.
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.
phone: no call list entries written for non-connected calls terminated by myPBX
| Status | Closed |
| Id | 112688 |
This happened for example when call to a busy or not-responding peer was terminated bx myPBX.
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.
myPBX: Login was rejected in standby case if UC license was used
| Status | Closed |
| Id | 112695 |
With myPBX license it worked
HTTP MOVE did not work with a 'Destination:' URI containing special characters
| Status | Closed |
| Id | 112715 |
In when storing a voice mail record the User Name becomes a part of the URI.
If this name contained special characters the recording could not be stored because the 'Destination:' request header field was not URL-encoded.
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.
PBX Waiting: If mobile only operators were present only each second call worked
| Status | Closed |
| Id | 112753 |
The call clearing to the mobile only operator did not work correctly, so when the next call arrived the call was not sent to the operator. Only when this call was cleared the clearing of the call to the operator was completed so the next call worked again.
Trap in webdav client when processing XML directory listing
| Status | Closed |
| Id | 112764 |
Trap in webdav client when processing XML directory listing.
crash at startup when a second registration was configured and the first deleted
| Status | Closed |
| Id | 112773 |
mute ringer in tray icon does not work
| Status | Closed |
| Id | 112776 |
when there are a few dump files in the roaming dir the compression script times out prematurely
| Status | Closed |
| Id | 112779 |
Fax server: Trap on invalid DCS message
| Status | Closed |
| Id | 112860 |
If an invalid T.30 DCS message is received, the device traps during the training validation of an incoming fax document. This is fixed now.
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: 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
Updated translations
| Status | Closed |
| Id | 112173 |
Translated texts have been improved.
phone: ip222,ip232: support for Jabra BIZ 2300, Sennheiser Presence UC
| Status | Closed |
| Id | 112335 |
myPBX msi installer option "OFFICEPROVIDER"
| Status | Closed |
| Id | 112407 |
Added msi installer option "OFFICEPROVIDER" to set the wished office presence provider.
Default is "myPBX".
the new icollect feature is now available in the path immediately after installtion
| Status | Closed |
| Id | 112777 |
without the user having to logout/Login first
increase trace file limit to capture large traces
| Status | Closed |
| Id | 112778 |
V10 Service Release 7 (100998)
Changes included in Version 10 Service Release 7 Definition
PBX: CFNR configured at PBX object was executed under unexpected circumstance
| Status | Closed |
| Id | 111587 |
The CFNR at a PBX object is used for rerouting a call when the IP connectivity to the location is not available. It should not be executed for calls which fail when routed back to the location for node external.
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
SIP: Client must auto re-open TLS connections if closed by server
| Status | Closed |
| Id | 112922 |
According to RFC-5626 "Client-Initiated Connections in SIP"
the client is responsible to open, keep-alive and re-open
transport connections all the time the client is registered.
Status:
Fixed in 11.00 and 10.00
LDAP Server Statistics: Connection Counter Could Get Wrong
| Status | Closed |
| Id | 113034 |
The counter for connections with write access could wrap below zero. A merge from v11
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.
myPBX: More compact message for updating allows
| Status | Closed |
| Id | 113101 |
The message size was reduced so the user is able to manage visibility settings with more entries before the message exceeds the maximum command line size.
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
WebDAV-Client: Trap when parsing directory listing
| Status | Closed |
| Id | 113145 |
Trap when parsing XML directory listing.
If directory listing is bigger than 4000 bytes and delivered chunked encoded.
PBX: When a CF was executed on a Gateway/Trunk object to another node, additional dialed digits were not handled correctly
| Status | Closed |
| Id | 113146 |
This maybe used to divert to a trunk on another node if the local trunk is not available
PBX: When calling by name together with nodes with escapes, wrong number displayed in ringback
| Status | Closed |
| Id | 113167 |
In this state there is no number available, this number was adjusted anyway as it could be dialed by the calling user, which resulted in all the escapes added to the called number (which was empty) as ptrefix
PBX: Subscription calls to for other locations, when the other PBX was not online generated CDRs
| Status | Closed |
| Id | 113211 |
Subscription calls should never generate CDRs. Because these subscription calls are retried, it could be a high number of CDRs
SIP: Bug on media negotiation (LYNC interoperability)
| Status | Closed |
| Id | 113244 |
LYNC interop.
Introduced with v10sr5.
If receiving LYNC's SDP offer:
v=0
o=- 29 1 IN IP4 192.168.10.3
s=session
c=IN IP4 192.168.10.3
b=CT:1000
t=0 0
m=audio 55978 RTP/AVP 97 101 13 0 8
c=IN IP4 192.168.10.3
a=rtcp:55979
a=label:Audio
a=sendrecv
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
where RED/DTMF/CN are offered as most-preferred,
an SDP answer is generated without any real audio codec:
v=0
o=- 59 1 IN IP4 192.168.10.2
s=-
t=0 0
m=audio 16902 RTP/AVP 101 13
c=IN IP4 192.168.10.2
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
myPBX launcher: No line wrap in multi-line desktop notifications
| Status | Closed |
| Id | 113262 |
If a desktop notification had multiple text lines and the first line had a line wrap the second text line was not shown. For example for incoming calls no phone number was shown in that case.
Now both text line are cut if the text is too long.
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.
myPBX: Prevent caching of requests to the reporting
| Status | Closed |
| Id | 113305 |
Browsers should not cache requests from myPBX to the reporting because of the response headers from the server. However on some PCs IE cached those requests, most probably because of the browser settings.
Now a unique dummy parameter is added to each request in order to prevent any caching.
SIP: Bug on media negotiation (Switch to T38)
| Status | Closed |
| Id | 113308 |
Bug on media negotiation if handling switch to T38
triggered by a complex SDP offer.
Eg:
\tv=0
\to=HuaweiSoftX3000 32082603 32082604 IN IP4 213.148.136.222
\ts=Sip Call
\tc=IN IP4 213.148.136.222
\tt=0 0
\tm=image 38028 udptl t38
\ta=T38FaxVersion:0
\ta=T38MaxBitRate:14400
\ta=T38FaxRateManagement:transferredTCF
\ta=T38FaxUdpEC:t38UDPRedundancy
\tm=audio 36624 RTP/AVP 8 103 0 127 101
\ta=rtpmap:8 PCMA/8000
\ta=rtpmap:103 PCMA/8000
\ta=gpmd:103 vbd=yes
\ta=rtpmap:0 PCMU/8000
\ta=rtpmap:127 PCMU/8000
\ta=gpmd:127 vbd=yes
\ta=rtpmap:101 telephone-event/8000
\ta=ptime:20
\ta=silenceSupp:off - - - -
\ta=ecan:fb on -
\ta=X-fax
\ta=fmtp:101 0-15
Must return an SDP answer with with same media-descriptions.
Not only the accepted media-description.
also the rejected media-description.
SIP: Bug on media negotiation when receiving complex offer with 'sendonly'
| Status | Closed |
| Id | 113353 |
Bug on media negotiation when receiving 'sendonly' offer with multiple media descriptions
E.g.
v=0
o=- 7868 7871 IN IP4 10.38.60.15
s=-
c=IN IP4 10.38.60.15
b=AS:512
t=0 0
m=audio 0 RTP/AVP 18
m=audio 2290 RTP/SAVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=X-mitel-dtmf-det-required:yes
a=sendonly
a=crypto:61 AES_CM_128_HMAC_SHA1_32 inline:OsT62HHcplTaldUJEPG7dyhMfTVdAtpwPWidfINu
Answer should not contain a port for unencrypted audio (first media description).
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
Video: no video displayed in second monitor if video capability was disabled prior to the call
| Status | Closed |
| Id | 113867 |
If video is enabled during the call and video window locates in second monitor, initialization fails and no video is displayed.
SIP: Memory leak when subscriptions are rejected
| Status | Closed |
| Id | 113895 |
Memory leak when subscriptions are rejected
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.
SIP: SUBSCRIBE without Expires header was not handled
| Status | Closed |
| Id | 114004 |
SUBSCRIBE request without Expires header was handled as UNSUBSCRIBE.
Must be handled as SUBSCRIBE with defualt expiration time.
SIP: Trap in federation scenario
| Status | Closed |
| Id | 114012 |
Trap in federation scenario when processing INVITE.
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.
SIP: Bug in media negotiation on incoming SIP calls to waiting queue
| Status | Closed |
| Id | 114146 |
Bug in media negotiation on incoming SIP calls to waiting queue.
Video: webcam was not working due to unsupported frame rate
| Status | Closed |
| Id | 114197 |
new frame rate format added, {30000, 1001}
SIP: Failed to handle huge SDP bodies
| Status | Closed |
| Id | 114213 |
Failed to handle huge SDP bodies like this:
INVITE sip:claudiotest@ipva.hctech.se SIP/2.0
Record-Route: <sip:FE13.hctech.local:5061;transport=tls;opaque=state:T;lr>;tag=5A9958F1F83182177B8B6F02A94424B5
Via: SIP/2.0/TLS 172.31.210.31:52060;branch=z9hG4bK59CFDA17.C91FD5D9A3865680;branched=FALSE
Max-Forwards: 69
ms-application-via: SIP;ms-urc-rs-from;ms-server=FE13.hctech.local;ms-pool=FE13.hctech.local;ms-application=ad894dc3-55e0-44bf-a07e-3c073aaa4a57
P-Asserted-Identity: "Claudio Innovaphone"
Via: SIP/2.0/TLS 172.31.210.31:52601;branch=z9hG4bK368afcc;ms-received-port=52601;ms-received-cid=570C00
FROM: "Claudio Innovaphone"<sip:claudio.innovaphone@hctech.se>;epid=31964416546;tag=c66ee5e7b
TO: <sip:claudiotest@ipva.hctech.se>
CSEQ: 15961 INVITE
CALL-ID: c0c8a81d-c6c0-438f-ac39-1c03864a9207
CONTACT: <sip:claudio.innovaphone@hctech.se;opaque=user:epid:pHOn6L2qfFGyxFexNhwK9QAA;gruu>;text;audio;video;image;applicationsharing
CONTENT-LENGTH: 4179
EXPIRES: 600
PRIORITY: Normal
SUPPORTED: Replaces
SUPPORTED: ms-dialog-route-set-update
SUPPORTED: timer
SUPPORTED: 100rel
SUPPORTED: gruu-10
USER-AGENT: RTCC/5.0.0.0 UCWA/5.0.0.0 iPadLync/5.3.1085.0000 (iPad iPhone OS 7.0.4)
CONTENT-TYPE: multipart/alternative; boundary=XpkGhOpIXneTUNv0mJ3NxGMH80e02zdO
ALLOW: ACK
Ms-Conversation-ID: Ac8g/N7em/1n1FILS6CyuvX2OSGxkA==
ms-endpoint-location-data: NetworkScope;ms-media-location-type=Intranet
Session-Expires: 1800
Min-SE: 90
Allow: CANCEL,BYE,INVITE,REFER,MESSAGE,INFO,SERVICE,OPTIONS,BENOTIFY,NOTIFY,PRACK,UPDATE
ms-routing-phase: from-uri-routing-done
ms-user-data: ms-publiccloud=TRUE;ms-federation=TRUE
--XpkGhOpIXneTUNv0mJ3NxGMH80e02zdO
Content-Type: application/sdp
Content-ID: <1d5b309b9a8293cf3441cac6fb7dde95@LyncMobileHostName>
Content-Disposition: session; handling=optional; ms-proxy-2007fallback
v=0
o=- 0 0 IN IP4 172.16.3.25
s=session
c=IN IP4 172.16.3.25
b=CT:45292
t=0 0
m=audio 46806 RTP/AVP 9 111 0 8 97 13 118 101
a=candidate:w6/WXWlcC4Nk+0HxvHLmHrhulTL5VevRUIzxs3NhwU4 1 g2pg3QjJYe/pXNcUTVsLaA UDP 0.830 172.16.3.25 46806
a=candidate:w6/WXWlcC4Nk+0HxvHLmHrhulTL5VevRUIzxs3NhwU4 2 g2pg3QjJYe/pXNcUTVsLaA UDP 0.830 172.16.3.25 46807
a=candidate:ysgbE4G6GC1mYtc2BRsjSMHYBDYnP3LtRTNxgPenth8 1 UU2hDzw21fYDXpDImjQw9A TCP 0.110 195.67.92.245 58394
a=candidate:ysgbE4G6GC1mYtc2BRsjSMHYBDYnP3LtRTNxgPenth8 2 UU2hDzw21fYDXpDImjQw9A TCP 0.110 195.67.92.245 58394
a=candidate:xuKM/oyDT0Pvq+HF0qnW08To4aOla9ULuZkofs9PTiM 1 q4T0j4eo0qOX8FcvWxX5qA UDP 0.410 195.67.92.245 59592
a=candidate:xuKM/oyDT0Pvq+HF0qnW08To4aOla9ULuZkofs9PTiM 2 q4T0j4eo0qOX8FcvWxX5qA UDP 0.410 195.67.92.245 50096
a=candidate:om95MNklHK73JIqsJlCl4l3a6TWHRgyXVsBPfOghEpM 1 Bc5H52Yj/3ICbrlD1AVtHA TCP 0.250 145.253.157.4 64869
a=candidate:om95MNklHK73JIqsJlCl4l3a6TWHRgyXVsBPfOghEpM 2 Bc5H52Yj/3ICbrlD1AVtHA TCP 0.250 145.253.157.4 64869
a=candidate:hm4TP0ii741xcbx5ASbzhbCT2Czn7ngMFxm45N2zzhI 1 ZTCLUUVz85FrnUDHCZarsg UDP 0.550 145.253.157.4 50847
a=candidate:hm4TP0ii741xcbx5ASbzhbCT2Czn7ngMFxm45N2zzhI 2 ZTCLUUVz85FrnUDHCZarsg UDP 0.550 145.253.157.4 50848
a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 inline:UXUAHDIlwLsuoHuLcz8KFrEjMfiwQXIl1FBIGbHR|2^31|1:1
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:H80GkPvoKW+FwDG55yMnwTCcWX5FKHYVmSv7DU+3|2^31|1:1
a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:Mlpx3DuqRs/bVOMt6m1R3G6ngSXjnjM2YG6PgltN|2^31
a=maxptime:200
a=rtpmap:9 G722/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 RED/8000
a=rtpmap:13 CN/8000
a=rtpmap:118 CN/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
--XpkGhOpIXneTUNv0mJ3NxGMH80e02zdO
Content-Type: application/sdp
Content-ID: <0cfe4d04f6e9fada7ec9f8c961b0decb@LyncMobileHostName>
Content-Disposition: session; handling=optional
v=0
o=- 0 1 IN IP4 172.16.3.25
s=session
c=IN IP4 172.16.3.25
b=CT:45292
t=0 0
m=audio 44976 RTP/AVP 9 111 0 8 97 13 118 101
a=x-ssrc-range:2846109442-2846109442
a=rtcp-fb:* x-message app send:dsh recv:dsh
a=rtcp-rsize
a=label:main-audio
a=x-source:main-audio
a=ice-ufrag:98gz
a=ice-pwd:qU8OZPLs8g921cPjFnZWWS7B
a=candidate:1 1 UDP 2130706431 172.16.3.25 44976 typ host
a=candidate:1 2 UDP 2130705918 172.16.3.25 44977 typ host
a=candidate:2 1 TCP-PASS 174456319 195.67.92.245 53026 typ relay raddr 145.253.157.4 rport 64868
a=candidate:2 2 TCP-PASS 174455806 195.67.92.245 53026 typ relay raddr 145.253.157.4 rport 64868
a=candidate:3 1 UDP 184548351 195.67.92.245 54834 typ relay raddr 145.253.157.4 rport 50845
a=candidate:3 2 UDP 184547838 195.67.92.245 54742 typ relay raddr 145.253.157.4 rport 50846
a=candidate:4 1 UDP 1694235135 145.253.157.4 50845 typ srflx raddr 172.16.3.25 rport 58714
a=candidate:4 2 UDP 1694234622 145.253.157.4 50846 typ srflx raddr 172.16.3.25 rport 58715
a=candidate:5 1 TCP-ACT 174847999 195.67.92.245 53026 typ relay raddr 145.253.157.4 rport 64868
a=candidate:5 2 TCP-ACT 174847486 195.67.92.245 53026 typ relay raddr 145.253.157.4 rport 64868
a=candidate:6 1 TCP-ACT 1684796927 145.253.157.4 64868 typ srflx raddr 172.16.3.25 rport 45567
a=candidate:6 2 TCP-ACT 1684796414 145.253.157.4 64868 typ srflx raddr 172.16.3.25 rport 45567
a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 inline:UXUAHDIlwLsuoHuLcz8KFrEjMfiwQXIl1FBIGbHR|2^31|1:1
a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:H80GkPvoKW+FwDG55yMnwTCcWX5FKHYVmSv7DU+3|2^31|1:1
a=crypto:3 AES_CM_128_HMAC_SHA1_80 inline:Mlpx3DuqRs/bVOMt6m1R3G6ngSXjnjM2YG6PgltN|2^31
a=maxptime:200
a=rtpmap:9 G722/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 RED/8000
a=rtpmap:13 CN/8000
a=rtpmap:118 CN/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
--XpkGhOpIXneTUNv0mJ3NxGMH80e02zdO--
| Status | Closed |
| Id | 114292 |
Fix for #111319: Go back to ac494004ce3.680.10.pf.01 on ip28 and related because the customer reported hanging calls and unability to idle reset which points to DSP code instability.
Files:
ac_494.cpp
RSTP: improved link recovery behaviour
| Status | Closed |
| Id | 114303 |
When a link comes up again after a failure a gratituos ARP request is brodcasted to make the new MAC/IP address assignment known in the network.
If this request gots lost the other hosts in the network continued to a now invalid MAC/IP assignment until some host specific timeout.
To overcome this problem now 10 gratituos ARP request are sent in 1/2 second intervals.
when downloading config the stream does not end due to not all data transmitted
| Status | Closed |
| Id | 114305 |
call recording with softphone, remote side not recorded
| Status | Closed |
| Id | 114306 |
early media support
| Status | Closed |
| Id | 114307 |
SIP: BYE is sent to wrong remote TLS port
| Status | Closed |
| Id | 114396 |
LYNC interoperability.
BYE request is sent to wrong remote TLS port 5061 instead of 5067.
| Status | Closed |
| Id | 112938 |
If OFFICEPRESENCE=false, no presence related installation changes are done and e.g. a Lync installation won't be broken. So Lync can be used as presence provider and myPBX can be used for calls etc.
SIP: Interworking issue with OpenStage systems
| Status | Closed |
| Id | 113175 |
Interworking issue with OpenStage systems.
Pass proprietary signaling options to phoneapp.
Voicemail: Read status of Boolean Object from Voicemail Script
| Status | Closed |
| Id | 113502 |
New feature of voicemail script for special applications
myPBX: Allow searching by display name in PBX directory
| Status | Closed |
| Id | 113739 |
In v10sr6 a new config option was introduced in the phone directory config. The administrator can now choose if the long name or display name should be used for LDAP search.
This config option is now also used by the LDAP search in myPBX. So serching by display name is not possible, if configured.
SIP: New config file option /product-id
| Status | Closed |
| Id | 113922 |
New config file option /product-id to set value of "User-Agent" or "Server" message header.
config change SIP /product-id innovaphone
SDP: Support for CN for 16kHz sample rate
| Status | Closed |
| Id | 114206 |
Support for CN for 16kHz sample rate.
When decoding SDP like:
v=0
o=- 0 0 IN IP4 172.16.3.25
s=session
c=IN IP4 172.16.3.25
b=CT:45292
t=0 0
m=audio 46806 RTP/AVP 9 111 0 8 97 13 118 101
a=maxptime:200
a=rtpmap:9 G722/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 RED/8000
a=rtpmap:13 CN/8000
a=rtpmap:118 CN/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
V10 Service Release 8 (101037)
Changes included in Version 10 Service Release 8 Definition
Allow certificates with empty subject DN
| Status | Closed |
| Id | 112635 |
If a subjectAlternativeName is given, certificates may have an empty subject DN. Currently this leads to an error when parsing the certificate.
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.
myPBX: Add external line prefix to non-international numbers from tel URIs
| Status | Closed |
| Id | 114399 |
When myPBX dials tel: URIs the external line prefix should be added, if needed. This also affects numbers that are dialed using the Office 2010 Integration.
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.
Holding mobility calls in myPBX not working
| Status | Closed |
| Id | 114541 |
Calls made with the mobility device could not be put on hold.
myPBX: Disable non-working call buttons for mobilty calls
| Status | Closed |
| Id | 114547 |
The following buttons do not work for calls via the mobility device.
* DTMF
* Park
* 3PTY conference
Therefore they are now disabled.
Video: possible trap using windows h264 decoder
| Status | Closed |
| Id | 114614 |
it is possible to habe a trap in the video library if the windows h264 decoder does not deliver an output sample.
myPBX: Config option to disable LDAP search in external directory
| Status | Closed |
| Id | 114695 |
Some users like to use the external directory only for number resolution, not for searching by name. This can now be done with the following config option.
!config add PBX0 MY /disable-search-external on
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.
Partnerkey Status did not update with Group membership updates
| Status | Closed |
| Id | 114808 |
When the Group Membership of a user was updated, active monitoring sessions (e.g. by a partner key) were not updated. A reset of the phone was necessary.
| Status | Closed |
| Id | 114821 |
officepresence and officepresencelogging were only read from HKLM.
Voicemail failed in Chief+Secretary Scenario
| Status | Closed |
| Id | 114863 |
Audio prompting didn't start
SMTP: "DATE"-header missing
| Status | Closed |
| Id | 114889 |
alike: "Date: 13 Feb 2014 07:50 +0000"
SIP: re-INVITE(t38 and audio) was rejected with "SIP/2.0 488 Not Acceptable Here"
| Status | Closed |
| Id | 114924 |
re-INVITE(t38 and audio) was rejected with "SIP/2.0 488 Not Acceptable Here".
Better to send "SIP/2.0 200 OK" with an SDP answer that accepts audio and rejects t38.
Linux: IP configuration with device internal DHCP server
| Status | Closed |
| Id | 114962 |
If the Linux IP configuration is configured to DHCP client mode and the same device is the DHCP server, the IP route to the Linux ethernet interface isn't automatically set. This is fixed now.
Video: possible trap after fallback to windows h264 decoder
| Status | Closed |
| Id | 114969 |
one variable was still set to hardware decoder and because of that the library run into an infinite loop in the event of packet lost.
SIP: Authentication problem with XCAPI clients
| Status | Closed |
| Id | 114979 |
XCAPI uses different Contact-URI in REGISTER und INVITE.
INVITE was rejcted after authentication attempt.
Now a workaround works based on attribute 'username' of authorization header.
SIP: Do not start DNS query for "0.0.0.0"
| Status | Closed |
| Id | 115014 |
SIP stack takes "0.0.0.0" for a domain name and starts DNS query.
No sense.
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
SIP: Do not start interface until local IP address is available
| Status | Closed |
| Id | 115020 |
Do not start SIP interface until local IP address is available.
Problem on IPVA.
Default local address was known very soon,
but asking for local address by providing remote address
results in "0.0.0.0".
PBX XML Export/Import: Passwords lost after export/import cycle
| Status | Closed |
| Id | 115101 |
The XML export included the encrypted passwords as hexdump, whereas the import expected passwords as clear text. Now the import expects encrypted passwords as well.
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 Trunk: Set Calling=Diverted feature affected the internal leg of calls forked to trunk as well
| Status | Closed |
| Id | 115138 |
If ca call was forked to a trunk with the Set Calling=Diverted feature enabled, the call was sent on this trunk with a calling number of the forling user, which is correct, but to the forking user itself the call was sent with the same calling number, which is not correct.
myPBX: Presence not displayed for email addresses that do not match H.323 ID
| Status | Closed |
| Id | 115323 |
This problem occured if an email address was configured at the user object that was different from the H.323 ID. When searching a user by that email address, the presence was not displayed at the search result.
IP222/232: Cyrillic input was not possible
| Status | Closed |
| Id | 115471 |
Cyrillic input is now possible if phone's language is set to "Russian".
softphone not able to join kerberos with staging scripts
| Status | Closed |
| Id | 115492 |
Kerberos configuration from the "vars create" line below didn't work because the var was read in the constructor of the command module. Therefore need to create the vars earlier during startup before the command module is started. Now we do it before any other module is started.
vars create CMD0/KCMD p %3cjoin+realm="innovaphone.com"+user="_KADMIN_"+password="W6wkF;ihH2B9"+default-realm="innovaphone.com"+disable-local="false"+force="true"%3cserver+realm="innovaphone.com"+address="172.16.0.10"+port="88"+secondary-address="172.16.0.9"/%3e%3e%3c/join%3e
SIP: Must write c-line into each media description
| Status | Closed |
| Id | 115494 |
Must write c-line into each media description.
Even in SDP answers in rejected media descriptions.
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.
PBX Waiting: CDRs for call from another Waiting Queue did not show correct calling party
| Status | Closed |
| Id | 115609 |
When a call was forwarded from a Waiting Queue to another using DTMF dialing on the second Waiting Queue the CDRs only showed the first Waiting Queue as calling party
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 |
Voicemail: Backup URL wasn't considered, if script wasn't found
| Status | Closed |
| Id | 115747 |
Was only considered in server-down situations
Trap when natting FTP control connection
| Status | Closed |
| Id | 115770 |
Potential trap when natting FTP control connection.
myPBX: Parked calls could not be unparked from favourites list
| Status | Closed |
| Id | 115773 |
If the user parked a call at a favourite, no unpark button was displayed at the favourite.
myPBX launcher: Desktop notifications displayed incorrectly with scaled windows font
| Status | Closed |
| Id | 115774 |
If the user uses 125% or 150% windows fonts the desktop notifications were not displayed, correctly.
Video: trap if GetAdaptersAddresses returned no addresses
| Status | Closed |
| Id | 115793 |
pointer was not checked before executing strcmp.
SIP: Bug when handling re-INVITE with complex offer (AVP and SAVP)
| Status | Closed |
| Id | 115821 |
Bug when handling re-INVITE with complex offer (AVP and SAVP).
An answer with two AVP sections were returned.
myPBX launcher: Increase maximum size of websocket messages
| Status | Closed |
| Id | 115849 |
Maximum size of websocket messages of 4096 bytes was not enough for the DeviceInfoResult message, if the user had 6 phones registered.
Trap during restart
| Status | Closed |
| Id | 116012 |
softphone not able to join kerberos with staging scripts
| Status | Closed |
| Id | 116016 |
SIP: White noise during re-negotiation from RTP to SRTP
| Status | Closed |
| Id | 116048 |
White noise during re-negotiation from RTP to SRTP.
It sometimes takes a while until media endpoint gets SDP answer for an SDP offer.
During that time of waiting the media endpoint may already receive SRTP packets
which still are handled as unencrypted RTP.
=== RTP session ===
Receiving INVITE w/o offer
Sending 200/OK with offer
Meta-reporter waiting for answer Meta-reporter
Receiving ACK with answer
=== SRTP session ===
Trap on outgoing SIP federation
| Status | Closed |
| Id | 116132 |
Trap on outgoing SIP federation call/subscription.
IP232,IP222,IP241: iresetn command did not work after changing options /transparent-header and /transparent-status
| Status | Closed |
| Id | 116162 |
Changing options /transparent-header and /transparent-status require reset to show any effect.
But "reset-required" indicator was not set.
That's why iresetn command did not trigger reboot.
DSP code update to revision 680.12
| Status | Closed |
| Id | 116163 |
change DSP code revision, avoids a trap when sending CLIP
Fax server: Domain only in lower case
| Status | Closed |
| Id | 116209 |
The fax server domain must be in lower case and is converted now. Otherwise incoming mails are rejected.
SIP: Don't mix G.722 2-channel with G.722 1-channel
| Status | Closed |
| Id | 116217 |
Don't mix G.722 2-channel with G.722 1-channel.
If both are offered.
LYNC interop.
SIP: Do not mix RTP/AVP and RTP/SAVP descriptions
| Status | Closed |
| Id | 116220 |
Do not mix RTP/AVP and RTP/SAVP descriptions.
myPBX: Phone numbers in history not clickable
| Status | Closed |
| Id | 116224 |
The phone numbers in the history are supposed to be copied to the search field when they are clicked. This 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 }
Support for private keys in PKCS#8 format
| Status | Closed |
| Id | 114282 |
Currently the upload of device certificates using a PEM file requires the RSA private key to be in PKCS#1 format. Now also the PKCS#8 format should be supported.
Documentation: http://wiki.innovaphone.com/index.php?title=Reference10:Certificate_management#Uploading_a_certificate_chain_together_with_the_private_key
myPBX: Updated translations
| Status | Closed |
| Id | 115369 |
Improved translations for launcher and web application.
Phone UI: Updated translations
| Status | Closed |
| Id | 115474 |
Updated translations (french, russian, german, etc.)
SIP: Interop to "Linphone/3.0.0 MX Video (eXosip2/3.1.0)"
| Status | Closed |
| Id | 115979 |
Interop to "Linphone/3.0.0 MX Video (eXosip2/3.1.0)".
Lookup of registration fails when INVITE is received from ep.
Because Contact-URI is different:
REGISTER: Contact: <sip:Tuerkamera@10.49.1.193:5060;line=70eea592e338871>
INVITE: Contact: <sip:Tuerkamera@10.49.1.193:5060>
Now Contact-URI's are compared excluding any URI parameters.
Phones: Write error message into trace, if background image cannot be loaded
| Status | Closed |
| Id | 115984 |
Write error message into trace, if background image cannot be loaded due to network issues for diagnostics.
Saying
phone_app: Can't load background image due to inaccessibility (http://x.x.x.x/...)
myPBX: Pass normalized numbers to external application
| Status | Closed |
| Id | 115995 |
Two new parameters for the external Application
$I - normalized number in international format +4970317300988
$N - normalized number in national format 070317300988
Add rtp trace to debug
| Status | Closed |
| Id | 116009 |
RTP tracing is added to debug
V10 Service Release 9 (101073)
Changes included in Version 10 Service Release 9 Definition
IP232: Display timing fixed(2)
| Status | Closed |
| Id | 108730 |
Display clock inverted, Sync inverted
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.
myPBX: Hotkey was not working in some applications
| Status | Closed |
| Id | 115805 |
Users reported that the Hotkey did not copy text in some applications (like IE and Chrome).
This happened because myPBX didnt give the application enough time to copy the text.
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.
H.323: DTMF received as SIP INFO not forwarded on H.323 if no Media Relay
| Status | Closed |
| Id | 116393 |
Should be forwarded as UserInput message in this case.
SIP: Do not list audio codes when accepting t38 offer
| Status | Closed |
| Id | 116436 |
Do not list audio codes when accepting t38 offer:
\tv=0
\to=- 1 2 IN IP4 85.232.5.106
\ts=-
\tt=0 0
\tm=image 5002 udptl t38
\tc=IN IP4 85.232.5.106
\ta=T38FaxVersion:0
\ta=T38MaxBitRate:14400
\ta=T38FaxFillBitRemoval:0
\ta=T38FaxTranscodingMMR:0
\ta=T38FaxTranscodingJBIG:0
\ta=T38FaxRateManagement:transferredTCF
\ta=T38FaxUdpEC:t38UDPRedundancy
\tm=audio 0 RTP/AVP 8 101 13
\tc=IN IP4 0.0.0.0
\ta=rtpmap:101 telephone-event/8000
IP232: Delay when new screen is opened
| Status | Closed |
| Id | 116465 |
Construction of a screen takes very long.
Introduced with SR8.
Fax server: Switching to T.38 first after voice coder initialization
| Status | Closed |
| Id | 116487 |
The FAX interface shouldn't switch to the T.38 coder for incoming calls until a voice coder is initialized. Now this is fixed and the DTMF code is also removed.
It fixes a CCM compatibility.
IP-DECT: Remote control connect for second call
| Status | Closed |
| Id | 116636 |
The remote control connect feature for the second and more calls doesn't work. The DECT base station sends an alert instead of a connect event. This is fixed now. It is used in combination with myPBX.
IP-DECT: Display update after call transfer with alert
| Status | Closed |
| Id | 116637 |
The handset display is updated after a successful call transfer with the alert event instead of only the later connect event.
Shutting down stale TCP connections
| Status | Closed |
| Id | 116690 |
Close and re-open TCP connection if local IP addr changes.
PBX DTMF: Could not release call from myPBX or SOAP
| Status | Closed |
| Id | 116699 |
SIG_REL / SIG_DISC was intercepted.
New Checkmark Services/LDAP/Replicator/TLS
| Status | Closed |
| Id | 116705 |
This new checkmark "TLS"
*allows to activate LDAPS(LDAP over TLS) without the need to enter the well-known port as "<ip>:636"
*allows to activate LDAPS in hosted scenarios with TCP port forwarding
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: PRACK and other in-dialog requests was sent to wrong destination port
| Status | Closed |
| Id | 116859 |
PRACK and other in-dialog requests was sent to wrong destination port.
PBX Waiting: A DTMF mapping in a Waiting Queue with a name as destination addressing a Voicemail object could create wrong calls
| Status | Closed |
| Id | 116892 |
An overlap dialing timeout was used to connect to the Voicemail object, if within this timeout more digits were dialed, the call was sent to the number identified by these dialed digits. Now the Voicemail object is connected right away.
SIP: Fix for media problem on WLAN phones
| Status | Closed |
| Id | 116927 |
Fix for media problem on WLAN phones.
Try to avoid unnecessary CHANNEL_INIT to work-around DSP issue.
SIP: Fix for media problem on DECT gateways
| Status | Closed |
| Id | 116984 |
Fix for media problem on DECT gateways durring hold/retrieve actions.
H.323: Potential trap when receiving a connected number longer then 29 digits
| Status | Closed |
| Id | 117024 |
In real life this should not happen except as part of an attack.
SIP: Release call immediately if interworking of ISDN/DISCONNECT is not supported
| Status | Closed |
| Id | 117038 |
Release call immediately if INFO(application/isup) for DISCONNECT is rejected by remote side.
SIP: Do not offer mix of audio and t38
| Status | Closed |
| Id | 117046 |
Do not offer mix of audio and t38 simultaneously.
Better
\tv=0
\to=- 1 1 IN IP4 172.16.131.106
\ts=-
\tt=0 0
\tm=audio 16414 RTP/AVP 18 0 8 4 96 9 101 13
\tc=IN IP4 172.16.131.106
\ta=rtpmap:96 G726-16/8000
\ta=rtpmap:96 G726-24/8000
\ta=rtpmap:96 G726-32/8000
\ta=rtpmap:96 G726-40/8000
\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=image 0 udptl t38
\tc=IN IP4 0.0.0.0
Than
\tv=0
\to=- 1 1 IN IP4 172.16.131.106
\ts=-
\tt=0 0
\tm=audio 16414 RTP/AVP 18 0 8 4 96 9 101 13
\tc=IN IP4 172.16.131.106
\ta=rtpmap:96 G726-16/8000
\ta=rtpmap:96 G726-24/8000
\ta=rtpmap:96 G726-32/8000
\ta=rtpmap:96 G726-40/8000
\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=image 16416 udptl t38
\tc=IN IP4 172.16.131.106
\ta=T38FaxVersion:0
\ta=T38MaxBitRate:14400
\ta=T38FaxFillBitRemoval:0
\ta=T38FaxTranscodingMMR:0
\ta=T38FaxTranscodingJBIG:0
\ta=T38FaxRateManagement:transferredTCF
\ta=T38FaxUdpEC:t38UDPRedundancy
Webdav: Potential bug in directory listing
| Status | Closed |
| Id | 117088 |
Using bad pointer.
SIP: Do not send "Privacy: id" in re-INVITE on incoming call
| Status | Closed |
| Id | 117119 |
Do not send "Privacy: id" in re-INVITE on incoming with anyonymous calling party.
Remote Media: Redirect to different PBX did not work with non-standard ports
| Status | Closed |
| Id | 117129 |
If the target PBX of a redirect did not use a standard HTTP or HTTPS port, video did not work.
SIP: Config option "Filter incoming calls" did not always work as expected
| Status | Closed |
| Id | 117134 |
Neither did the uri-to-phonenumber conversion.
Did not work if Request-URI contains a port number.
E.g.
INVITE sip:622@def.com:5060 SIP/2.0
Protocol handlers didn't work if myPBX was not shown in taskbar
| Status | Closed |
| Id | 117158 |
The protocol handlers like tel:, im: ... didn't work if myPBX was not shown in taskbar, just in system tray.
IP232,IP222,IP241: Trap when loading background image from HTTP server
| Status | Closed |
| Id | 117226 |
Trap when loading background image from HTTP server.
But only with some HTTP servers using specific transfer mode.
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
PBX: Trap if number of concurrent connections exceeds 4095
| Status | Closed |
| Id | 117257 |
Endless loop trying to find new free transfer id
Do not set unknown phone numbers HOME/MOBIL with myPBX in Office
| Status | Closed |
| Id | 117309 |
Only set phone number work, as this number is known in myPBX.
Leave the other numbers unknown.
IP28: Qtrace loop current display shows alway zero.
| Status | Closed |
| Id | 117332 |
Hosting; Softwarephone could not obtain license in innovaphone Hosting setup
| Status | Closed |
| Id | 117345 |
The only the first license challenage was forwarded by the session object
PBX E.164: When dialing to a different node to public network from Slave, wrong number display in myPBX during Ringback
| Status | Closed |
| Id | 117352 |
For calls sent to the master the display number was adjusted in Ringback already, but the information about the correct number is not available in this state.
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.
SIP: Fixed handling of overlap dialing
| Status | Closed |
| Id | 117400 |
Fixed handling of overlap dialing.
If overlapping INVITE's are received very quickly,
dialing digits may get lost.
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.
SIP: Assign bearer capability "3.1 kHz audio" instead of "Speech"
| Status | Closed |
| Id | 117420 |
Assign bearer capability "3.1 kHz audio" instead of "Speech"
when interworking incoming INVITE to H.323/ISDN/etc.
To workaround fax problems.
IP222/IP232: Reduce memory footprint of display rendering
| Status | Closed |
| Id | 117501 |
Reduce memory footprint of display rendering.
Try to allocate from re-usable memory as long as possible.
ISDN: If interface mapping was used to change type of number, only the first digits used the new type
| Status | Closed |
| Id | 117571 |
So different digits were sent with different type of numbers. This was treated as protocol error by some switches and the call was released
PBX Mobility: Dialtone was generated even if a call was sent out already because of callthru
| Status | Closed |
| Id | 117828 |
callthru can be initiated by additional digits sent with the original call to the mobility object, or by "data callthru". In both cases no dialtone should be generated to the calling user.
SIP: Fix for Server/User-Agent header on DECT devices
| Status | Closed |
| Id | 117851 |
Fix for Server/User-Agent header on DECT devices.
Keep white space:
Before: Server: Mitel-5604-SIP-Phone4.0.10000000000000036470548635
After: Server: Mitel-5604-SIP-Phone 4.0.10 000000000000 036470548635
SIP: Trap in federation scenario
| Status | Closed |
| Id | 118046 |
Trap in federation scenario in context of a DNS query.
softwarephone crashed when the ldap server was not reachable
| Status | Closed |
| Id | 118054 |
Softwarephone ignore setup message if this message have big size.
| Status | Closed |
| Id | 118055 |
Set TOS value for calls instead of in the registry
| Status | Closed |
| Id | 118058 |
Registry Setting of TOS value does not work properly, set for every call instead
PBX: Potential restart on blind-transfer, call-clearing collision
| Status | Closed |
| Id | 118097 |
There is only a window of a fraction of a ms for this, so this restart should have been very unlikely
FXS with Feature Codes:Hanging calls if incomplete feature codes were dialed
| Status | Closed |
| Id | 118192 |
A collateral damage of fix: #88471: DTMF Features: Allow lokal functions if registration to PBX is not available. The hanging call did not use up any resources, so it was just a memory leak.
SIP: Fixed handling of t38-reject
| Status | Closed |
| Id | 118258 |
If reINVITE with t38 was rejected, the call was not re-configured back to audio.
Did not work on media-relay interfaces.
SIP: Do not reject REGISTER without Contact header
| Status | Closed |
| Id | 118290 |
Better send 200/OK instead of 400/Bad Request
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.
IP-DECT: Call disconnects by user data changes
| Status | Closed |
| Id | 118352 |
User data updates with LDAP and with an empty authorisation name cause call disconnects. This is fixed now.
PBX Trunk: Make "Send Number" configurable
| Status | Closed |
| Id | 114875 |
So that for incoming calls on a Trunk object not the Trunk number is added as prefix, but a configurable number.
myPBX: New installer
| Status | Closed |
| Id | 116438 |
Upgrade version 10 to Visual Studio 2013 and WiX installer.
IP222/232: Changed letter mapping of telephone keypad
| Status | Closed |
| Id | 116492 |
Letter mapping for all latin languages contains basic latin alphabet plus diacritic variants.
Letter mapping for all cyrillic languages contains basic cyrillic alphabet plus basic latin alphabet.
See here for details:
http://wiki.innovaphone.com/index.php?title=Howto:Typing_text_on_telephone_keypad
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.
SIP: New config file option /fixed-media-addr
| Status | Closed |
| Id | 117475 |
New config file option /fixed-media-addr
config change SIP /fixed-media-addr 10.10.10.10
PBX: Fax license for Gateway and Map objects configurable
| Status | Closed |
| Id | 117638 |
Needed for special setups with a fax server without personal fax, or Fax numbers associated to users, which are not just the user number with the Fax gateway number as prefix.
Fax server: Call proceeding event after enbloc timeout
| Status | Closed |
| Id | 117937 |
The FAX interface sends the call proceeding event now if the sending complete is set or after timeout. It is used by the fax server application to receive a fax to the fax gateway number without any other dialed digits.
Show corrupted file and directory names after check disk
| Status | Closed |
| Id | 118243 |
After a check disk with corrupter files or directories, the names are now shown.
SIP: Blacklist for IP addresses to fight brute-force attacks
| Status | Closed |
| Id | 118367 |
Blacklist for IP addresses to fight brute-force attacks.
Not answering requests from IP address on that list.
This list is maintained automatically (hosts get on that list for a while if an invalid registration originated from there, even if they are part of the configurable registration white lists).
V10 Service Release 10 (101084)
Changes included in Version 10 Service Release 10 Definition
phone: keep remote party name after connect when dialled and connected number differ in first digits only
| Status | Closed |
| Id | 118537 |
PBX Exec: Status of CFU setting was not displayed for secondary secretary (only primary)
| Status | Closed |
| Id | 118538 |
The Exec Partner Key display the CFU status of the secretary. But this did not work for secondary secretaries, only primary
SIP: Memory leak on DECT systems
| Status | Closed |
| Id | 118539 |
Memory leak on DECT systems.
myPBX: Customized logo was not displayed in Safari browser
| Status | Closed |
| Id | 118543 |
Changing the logo did not work in Safari.
SIP: INVITE wrongly accepted
| Status | Closed |
| Id | 118561 |
Mis-routed INVITE was wrongly accepted.
Check Request-URI of received INVITE against local Contact-URI.
Do not accept INVITE if no match.
Updated translations
| Status | Closed |
| Id | 118571 |
Use shorter polish translation for "indirect dialing".
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.
myPBX launcher: Some MSI parameters did not work in new installer
| Status | Closed |
| Id | 118874 |
The following parameters did not work properly:
* AUTOSTART
* SHOWINTASKBAR
phone: a call dialled via myPBX with the phone already off-hook was sometimes connected in handsfree instead of handset mode
| Status | Closed |
| Id | 118939 |
It did depend on the PBX response time if call was connected in handset or handsfree mode.
phone: ip110/150/200a/230/240: false "Excessive loss of Data" reports when playing Music on Hold (MOH)
| Status | Closed |
| Id | 119055 |
incomplete naming of hid table for jabra pro 94(60/70)
| Status | Closed |
| Id | 119058 |
SIP: Error while processing REGISTER with Contact header missing IP address
| Status | Closed |
| Id | 119223 |
Error while processing REGISTER with Contact header missing IP address.
No 200/OK was sent.
SIP: Offered wrong local IP address as RTP address
| Status | Closed |
| Id | 119269 |
Offered wrong local IP address as RTP address.
Offered ETH1 address to remote endpoint,
but RTP with ETH0 address as source.
RTP was ignored by remote party.
And RTP from remote party was ignored also.
Resulted in no-media condition.
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
Create myPBX IM Provider registry key for current user on first myPBX start
| Status | Closed |
| Id | 119277 |
If the msi has been installed for all users on an administrator account, the current user key must be created by myPBX on first start for other users.
SIP: Wrong trace message
| Status | Closed |
| Id | 119331 |
Wrong trace message:
37:3366:014:7 - ERROR: SIP message too large: 88 bytes
37:3366:015:0 - SIP: Failed to decode message from 109.235.234.65:3478
Mis-routed STUN message handled as SIP message.
Video: free display queue if display driver must be resetted
| Status | Closed |
| Id | 119406 |
if display driver needs to be reinitialize, no interest on keeping old samples.
SIP: Trap in federation scenario
| Status | Closed |
| Id | 119494 |
Trap in federation scenario in context of a DNS query.
SIP: Bug when decoding complex SDP containing multiple media descriptions
| Status | Closed |
| Id | 119505 |
Bug when decoding complex SDP containing multiple media descriptions.
E.g.
v=0
o=OpenStage-Line_0 1225584267 2078663080 IN IP4 195.97.14.71
s=SIP Call
c=IN IP4 195.97.14.71
t=0 0
a=partnerfsm:2032616430_3_22053_0_1
m=audio 0 RTP/AVP 8 9 0 18 101
c=IN IP4 192.168.0.127
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexa=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
m=audio 59130 RTP/SAVP 8 9 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:RWyybo9wspQ5HMV2wYhcVZKBz6/3XQMOsaEJIlWa
Rejected media description contains c-line, but accepted media description does not.
c-line on session level is to be used for accepted media description.
But was not.
SIP: Bug in SDP version field handling
| Status | Closed |
| Id | 119595 |
Keeping track of the received SDP version did not work.
The version field of SDP is used to detect session modifications.
Bug may result in negotiation error.
SIP: Bug in media re-negotiation
| Status | Closed |
| Id | 119635 |
Bug in media re-negotiation on media-relay interfaces.
Jabra Link 280 an Softwarephone, wrong ring device slected, stops working
| Status | Closed |
| Id | 119644 |
Audio device not recognized due to name change
| Status | Closed |
| Id | 119645 |
For some reason Windows assigns a new name for an already known usb Audio device. For examle: "microphone (Jabra 2400 Mono)" can change to "microphone (2- Jabra 2400 mono). Then it is not recognized by softwarephone anymore and the user Needs to rerun the configuration. To alert the user to such a Situation a balloon tip Pops up every time the Audio device is beeing accessed
"Reset Required" indicated for IP config changes not requiring a reset (for example "ETHx/IP/Proxy ARP")
| Status | Closed |
| Id | 119734 |
This happened only on Linux capable devices as long as Linux was not configured.
PBX Mobility: Presence was sent with alert event if alert presence was not enabled in general config
| Status | Closed |
| Id | 120088 |
The check for this configuration was missing in the mobility case.
PBX-SOAP/CDRs: Show in CDRs that a call was initaited by SOAP
| Status | Closed |
| Id | 118572 |
There is a new tag "makecall" available in the CDRs. See CDR documentation.
V10 Service Release 11 (101106)
Changes included in Version 10 Service Release 11 Definition
Video: 352x288 RGB24 video format not working with win8.1
| Status | Closed |
| Id | 119054 |
RGB24 is the standard video format delivered by Logitech Webcams but this format in combination with a resolution of 352x288 is not working anymore in Windows 8.1.
As a workaroung in win8.1 I pick 320x240 as default resolution for this video format.
ip38 polarity detection false detect
| Status | Closed |
| Id | 119518 |
ip38 polarity detection false detect causes a line drop when 'drop line after polarity reversal' is enabled.
PBX: Potential restart on prickup from WQ
| Status | Closed |
| Id | 119958 |
Trap was only observed in version 11, but it could be a problem in 10.00 as well
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-SOAP: Trap when calling Admin with empty string
| Status | Closed |
| Id | 120015 |
Check for empty string added
SIP: Interop to Nortel CS1000 SIP GW
| Status | Closed |
| Id | 120054 |
Calls not handled correctly due to "phone-context" notation.
E.g.
To: <sip:78851;phone-context=cdp.udp@TELEFONI.RM.DK;user=phone>
From: <sip:anonymous@anonymous.invalid;user=phone>;tag=37a40b8-a2a530a-13c4-55013-3f2a69-7a7796d5-3f2a69
SIP: Interop to LYNC
| Status | Closed |
| Id | 120065 |
Ignore SDP offers tagged with "ms-proxy-2007fallback" due to incompatible ICE encoding.
Only in case of "multipart/alternative" offering.
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.
SIP: Carrier calls are rejected with "404 Not Found"
| Status | Closed |
| Id | 120166 |
Carrier calls are rejected with "404 Not Found" if the Request-URI
of the received INVITE does not match the registered Contact-URI.
Change was introduced with SR10.
Fax server: T.30 failure with bad-sig-end indication
| Status | Closed |
| Id | 120167 |
There is a T.30 failure with incoming fax calls if a bad-sig-end indication is received after the data. The call is disconnected. This is fixed now.
Remote Number display in ringback state with master/slave and nodes with escapes sometime with too many escapes
| Status | Closed |
| Id | 120201 |
Number adjustment did nor work under special conditions
LDAP Extensible Match Filter Encoding Failed
| Status | Closed |
| Id | 120234 |
The ASN.1 encoding of an LDAP filter term containing an extensible match failed for some variations.
TLS: Resuming session could fail depending on timing
| Status | Closed |
| Id | 120371 |
The session could be deleted from the cache after the client decided to do a resumed handshake but before the master key was read from the cache. If this happened a wrong master key was used and the trace showed "TLS DECODE ERROR!".
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 |
SIP: Wrong Contact-URI in response for INFO request
| Status | Closed |
| Id | 120563 |
Wrong Contact-URI in response for INFO request.
Returned the Contact-URI of the UAC in the response.
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
Voicemail: <pbx-fwd> failed in ACD scenario
| Status | Closed |
| Id | 120740 |
The session ended prematurely
SIP: New config file option /answer-all-options
| Status | Closed |
| Id | 120753 |
Interop fix for "User-Agent: commend SIP Series 3.1".
After registration this client send OPTIONS without Contact header.
Due to missing Contact header the OPTIONS request cannot be associated with the existing registration.
Unrelated OPTIONS requests are not answered due to security reasons.
If this new config file option is set, even unrelated OPTIONS requests are answered.
PBX: CSV import did not recognize UTF8
| Status | Closed |
| Id | 120763 |
When uploading a CSV file the UTF8 BOM was not always detected, depending on HTTP chunked encoding. As a result non-ascii charaters were broken in the imported user objects.
phone: ip222,ip232: Jabra UC VOICE 550/750 Version A - Microphone occasionally mute
| Status | Closed |
| Id | 120815 |
SIP: URI sometimes not escaped
| Status | Closed |
| Id | 120829 |
Sometimes userpart of SIP-URI misses escaping of reserved characters.
Softwarephone: Missing default update URL for provisioning
| Status | Closed |
| Id | 120857 |
The provisioning URL was not configured in the update script.
Presence Management on Phone worked different to myPBX
| Status | Closed |
| Id | 120946 |
It was not possible to clear own presence information
ip38 stops rerouting process if interface busy or unconnected
| Status | Closed |
| Id | 120972 |
ip38 call attempts to busy FXO are answered with cause 'busy' and to unconnected FXO with cause 'destination out of order'. Those causes forced the rerouting procedure to stop rerouting. The cause values are now changed into values 'no channel available' and 'network out of order', which do not stop rerouting.
PBX Waiting CDRs: Incomplete CDRs on calls to WQ with CFNR
| Status | Closed |
| Id | 120978 |
If a call to a waiting queue was forwarded because of a CFNR after the announcement was completed, this call resulted in a CDR without a <rel-from/> event.
H.323: Small memory leak when changing a registration
| Status | Closed |
| Id | 121021 |
If an unsuccessful registration was canceled by changing the registrarion parameters a queued registration could end up as leak.
PBX: URL as configuration option of a user object was removed by accident
| Status | Closed |
| Id | 121050 |
This was a collateral damage of "114875: PBX Trunk: Make "Send Number" configurable"
ISDN: Early Media did not wor for calls to an NT mode interface if turned on with PROGRESS
| Status | Closed |
| Id | 121059 |
For an outgoing call to an NT mode interface the other side can indicate in-band tones with a Progress Indicator in a PROGRESS message. This did not work. A Progress Indicator in SERUP-ACK, CALL-PROC or ALERT was no problem
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.
Softwarephone ringing through Jabra 930 base station
| Status | Closed |
| Id | 121109 |
Softwarephone ringing through Jabra 930 base Station enabled
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.
myPBX launcher: Desktop notifications displayed incorrectly with scaled windows font
| Status | Closed |
| Id | 121210 |
When Windows 7 users changed the font size to 125% or 150% the desktop notifications were not displayed correctly. A previous fix from v10sr8 did not work properly.
ip38 no audible dialtone
| Status | Closed |
| Id | 121224 |
Audible connection for outgoing ip38->PSTN calls was established right after dialing the destination number. Same time a 'connect' was assumed to the IP side. The connection is now already established with the detection of a central office tone, or, if no co detection is checked, 800ms after hook-off. The 'connect' to IP is sent 10sec after dialing or earlier, when a polarity reversal denotes a connection to the PSTN peer.
ip38 '+' character from FSK-Clip data in CGPN info
| Status | Closed |
| Id | 121227 |
former firmware versions didn't handle a '+' character in CGPN info resulting from FSK-Clip. The character is now removed.
SIP: Wrong order of digits in KPML subscription
| Status | Closed |
| Id | 121234 |
Wrong order of digits in KPML subscription.
SIP: Avoid unnecessary allocations
| Status | Closed |
| Id | 121286 |
Avoid unnecessary allocations of empty packets.
ip28 transmit-gain and receive-gain need higher attenuation values
| Status | Closed |
| Id | 121408 |
for special applications transmit-gain values -6dB and -12dB are added. Also receive-gain value -13dB is added.
PBX CDRs: Duplicate misleading entries in call list if a user has multiple endpoints registered
| Status | Closed |
| Id | 118312 |
If a user has multiple endpoints and one of these endpoints is not responding (e.g. a wireless endpoint out of range) a failed call was displayed even if the call was accepted by another endpoint in addition to the connected call.
SNMP: Trap Destination With Port Configuration
| Status | Closed |
| Id | 120701 |
A trap destination may now be configured as <ip address>":"<port>
myPBX: Use minimal browser UI on IOS 7.1
| Status | Closed |
| Id | 121226 |
Add the minimal-ui property to the viewport meta tag. This makes Safari display the web page in full screen mode.
V10 Service Release 12 (101154)
Changes included in Version 10 Service Release 12 Definition
Restart myPBX on Outlook COM exception
| Status | Closed |
| Id | 120744 |
The COM server is now restarted on a COM exception.
Therefor the myPBX UI is also restarted.
PBX: For replication to a standby PBX, configuration of dyn PBX Id did not work
| Status | Closed |
| Id | 121332 |
The user interface for configuring the dyn PBX Id did not work.
change to DSP code 680.25 due to stability problems
| Status | Closed |
| Id | 121410 |
this DSP code is proofed more stable
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
PBX CDRs: For calls forwarded by a CFNR from a WQ after announcement stops, CDRs did not show the CF
| Status | Closed |
| Id | 121501 |
The call was shown in the CDRs as if it was disconnected and the forward to the new destination was not visible.
Mobility: Inband ringback tone not always played
| Status | Closed |
| Id | 121540 |
Inband ringback tone not always played in mobility scenarios.
PBX Licenses: A license obtained on a slave from the master was not deleted when the license on the master was deleted
| Status | Closed |
| Id | 121598 |
Only the number of the still on the master existing licenses was updated
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.
Put calling party number in From-URI (not name)
| Status | Closed |
| Id | 121728 |
Put calling party number in From-URI (not name).
myPBX: Better handling of incomplete dialing location
| Status | Closed |
| Id | 121779 |
If the dialing location was incomplete the number normalization for the hotkey returned some unexpected results.
H.323 potential trap if call was received right after unregistration from same client
| Status | Closed |
| Id | 121796 |
This could happen either by a misbehaved client or some sort of race condition
PBX SOAP: Potential trap when UserRetrieve or similar function was attempted on the wrong call
| Status | Closed |
| Id | 121857 |
A null pointer access could result when calling UserRetrieve for a call handle which was just allocated by UserCall.
Video: 3rd party Conference did not work on Windows 8.1
| Status | Closed |
| Id | 121895 |
forgot to set size of a buffer. Strange that this issue was not a problem in Windows 7.
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.
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.
PBX Mobility: No voice after transfer of a call to a mobile phone to another mobile phone
| Status | Closed |
| Id | 122122 |
The signaling of the transfer did work, but MOH was continued on one mobile phone and silence was on the other.
IP38 'connect' signaling delay to calling phone too long
| Status | Closed |
| Id | 122182 |
ip38 used a 10sec delay to send SIG_CONN to the calling IP user. If the calling phone mutes its transmit data, the called peer cannot hear the caller for this 10s ec period. The delay is now reduced to 4sec. A DTMF digit sent by the caller will restart this delay to the whole 4sec again.
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
PBX: Unexpected Restart if group information was changed while an unacknowledged subscription was pending
| Status | Closed |
| Id | 122214 |
An obvious null pointer check was missing
phone recording - supress calling tones and call status display for calls to recording device
| Status | Closed |
| Id | 122221 |
IP232,IP222,IP241: Failed to show image data of some camera
| Status | Closed |
| Id | 122320 |
Problems decoding JPEG data.
Quickdial: Display Name Wasn't Processed
| Status | Closed |
| Id | 122322 |
A quickdial object's configuration field "Display Name" was evaluated incorrectly.
PBX: Limit reroutes to a fixed maximum number
| Status | Closed |
| Id | 122443 |
If a reroute loop is configured in endpoints, high load will be generated on the PBX, because these reroutes are executed quickly. A SIP phone with a call diversion uses reroute, because the call diversion cannot be configured in the PBX.
PBX: VoicemailUser license check did not work if the user and the Voicemail object were on different PBX
| Status | Closed |
| Id | 122495 |
There was no voicemail license found
IP232,IP222,IP241: Cannot delete call list entries of type callback
| Status | Closed |
| Id | 122507 |
Cannot delete call list entries of type callback.
Cannot even open context screen of such entries.
Gateway: Interface CGPN maps were executed twice if overlap call was received with no called party number in SETUP
| Status | Closed |
| Id | 122517 |
Interface maps are executed with the first dialing information received. This can be a SETUP message dor blockdial or an INFO message for overlap dial. In case the first dialing information came with INFO the cgpn map was executed twice, which could result in a wrong CGPN
PBX Waiting: "Max Call/Operator(%)" did not work correctly with "Presence disables Operator" and "Set Operator Presence"
| Status | Closed |
| Id | 122636 |
When calculating the number of operators, operators with presence set were not counted if "Presence disables Operator" was set. But a call to such an operator even if this call triggered the setting of presence because of "Set Operator Presence" was counted, so additional calls were rejecetd even if the configured percentage was not reached.
myPBX URL handlers didn't work with Windows 8/8.1
| Status | Closed |
| Id | 122645 |
As Microsoft changed the way how default applications are determined, URL protocol handlers have to be registered differently.
myPBX is now compatible to the Windows 8 behaviour.
PBX: Transfer to incomplete destination did not complete, if group-indications were configured at the transfered party
| Status | Closed |
| Id | 122664 |
The transfered party received music on hold until it hung up
Video: disconnect and clean up video call takes too long
| Status | Closed |
| Id | 122666 |
If the other participant starts a new video call just after he may receive two video streams because the old participant keeps sending video.
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.
ip38 calibration of Tx and Rx gains
| Status | Closed |
| Id | 122919 |
Txgain correction factor -1.2dB, Rxgain correction factor +0.3dB
Change to DSP code 700.00.pf.01 due to stability problems
| Status | Closed |
| Id | 122920 |
SIP: Mobility did not work with SIP
| Status | Closed |
| Id | 122942 |
Mobility did not work with SIP since RTP-DTMF was ot suppressed.
Video: RTCP socket not bound.
| Status | Closed |
| Id | 122966 |
I just tried to bound to one port (RTP+1) for rtcp channels instead of trying with some others.
PBX: Called number display on calls page or in myPBX wrong during ringback when calling a destination in another node by name
| Status | Closed |
| Id | 122970 |
In case a destination in another node was called, not the complete number including prefixes was displayed but just the number of the destination within its node
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.
| Status | Closed |
| Id | 122993 |
This was an incomplete merge of fix "110862: myPBX: Pickup accross PBX failed and could cause Trap" from v11 to v10
Voicemail: <exec url="mailto:..." out-error="$exec-err"/> passes mailto result into script
| Status | Closed |
| Id | 121518 |
A numeric value '1' indicates that an error occurred while executing a "mailto:"-request.
Set pwd input control to numeric mode to speed up hotdesking (for PIN input)
| Status | Closed |
| Id | 121922 |
Pre-set pwd input control to numeric mode to speed up hotdesking (for PIN input).
Can manually be changed back to alpha by long-press.
V10 Service Release 13 (101160)
Changes included in Version 10 Service Release 13 Definition
myPBX uses the primary email address of the PBX users for Outlook integration now
| Status | Closed |
| Id | 121287 |
Before this fix the email address was the combination of h323 and domain, but this mustn't be necessarily the primary email address, which is determined inside the PBX.
phone: ip241: 'Ok' key inserts newline characters in number/name input fields
| Status | Closed |
| Id | 123369 |
This way numbers may be misinterpreted as names.
PBX Waiting: Only/Only Not filter on diversion did not work for last diverting endpoint
| Status | Closed |
| Id | 123382 |
Problem when chaining WQs by diversions with filter confitions
HTTPCLIENT: trap when an application cancels a request inmidst DNS-name resolution
| Status | Closed |
| Id | 123401 |
ip38 : unable to detect norway-style central office tone
| Status | Closed |
| Id | 123787 |
norway-style central office tone is defined as 600ms pulse and 20ms pause. Detection did not work if detected pause was below 60ms, which was the allowed deviation in the detection algorithm.
Video: myPBX crashes when it is closed while webcam window is open
| Status | Closed |
| Id | 124195 |
tried to free videoReader class twice.
Gateway: Clearmode calls from or to an ISDN interface did not support SRTP
| Status | Closed |
| Id | 124226 |
The SRTP config was simply ignored in this case.
Gateway: Enblock timeout did not work if no number was dialed
| Status | Closed |
| Id | 124227 |
This is intended behaviour in case of an FXS interface without any destination number in mappings or routes, so that a user can pick up the receiver and take his time to think about the number to dial.
It does not make sense for an incoming call on an FXO interface with a destination number configured in routes.
Video: possible trap if video output could not be initialized.
| Status | Closed |
| Id | 124376 |
also in combination with a monitor changed.
myPBX: Do not remove 0 from Italian phone numbers
| Status | Closed |
| Id | 124411 |
In Italy the 0 after the country code is not removed in international numbers.
0XXXXXXXXXX -> +490XXXXXXXXXX
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.
ip38 : assume empty CLIP if no CLIP was detected on incoming FXO call
| Status | Closed |
| Id | 124446 |
if no CLIP was detected within 3500ms, an empty SIG_INFO message is sent to relay
ip38: detection of disconnect/busy tone improved
| Status | Closed |
| Id | 124766 |
now twice as fast, typically 2seconds
phone: optionally the calling tone can be replaced by a repeated notification tone when an new call arrives
| Status | Closed |
| Id | 122717 |
This can be configured via
config add PHONE APP /knock-into-calling 1
and disabled again via
config add PHONE APP /knock-into-calling 0
The calling tone is restored when the new call is released before the outbound call is connected or dropped.
Logging/CDR: permit both ip-address and domain name to address log servers connected via HTTP/HTTPS
| Status | Closed |
| Id | 123378 |
This applies to the log server Types HTTP, HTTPS, REMOTE-AP, REMOTE-AP-S
ip38 : Mark calls as internal/external calls by means of AC ring pulse length
| Status | Closed |
| Id | 124529 |
ring pulse length is variable
V10 Service Release 14 (101176)
Changes included in Version 10 Service Release 14 Definition
HTTPCLIENT: automatically retry simple PUT requests when the server requests re-authentication
| Status | Closed |
| Id | 124867 |
HTTPCLIENT supports three types of PUT requests:
- burst: the length of the request body and the complete body are provided in the initial put(),
i.e. httpclient::put(context, url, defaultfilename, header, data, data->length())
- stream: the length of the request body is provided in the initial put(),
the body is provided in parts by subsequent send_data() calls (for example a larger file).
- chunked: the length of the request body is not provided in the initial put(),
the body is provided in parts by subsequent send_data() calls.
Only a 'burst' type request can be retried when the server requests re-authentication.
For other request types the 401 status is passed to the user who can restart the request then.
IP111/222/232/241: lcd_bump did not show active popups
| Status | Closed |
| Id | 124902 |
/lcd_dump.bmp did not show active popups
SIP: CLEARMODE was not interworked into "Unrestricted Digital Information" if offered with "telephone-event"
| Status | Closed |
| Id | 124929 |
CLEARMODE was not interworked into "Unrestricted Digital Information"
if offered with "telephone-event".
v=0
o=- 1 1 IN IP4 172.16.131.102
s=-
t=0 0
m=audio 16386 RTP/SAVP 97 101
c=IN IP4 172.16.131.102
a=rtpmap:97 CLEARMODE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ZmMRYPO+msOBGiQwDnWtp5vRNygpLMCztohKSP/M
HTTPCLIENT: flow control by user did not work when "Connection: close" was set in the response header
| Status | Closed |
| Id | 124945 |
data was received at maximum speed and buffered by httpclient when the application could not consume the data at this speed. This could cause an out of memory trap when receiving large files for example in firmware upload.
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.
Voicemail: File System Provider Not Reset For Backup Skript-URL
| Status | Closed |
| Id | 124991 |
The primary and backup URL may now point to a local and/or remote script.
myPBX: Allow sending DTMF before connect
| Status | Closed |
| Id | 124997 |
The user should also be able to send DTMFs in the call states "complete" and "alerting". This is needed when the user wants to use DTMF in announcements before the connect.
SIP: Do not reject reINVITE with 482 Loop Detected
| Status | Closed |
| Id | 125068 |
Do not reject reINVITE with 482 Loop Detected.
Better reject with 491 Request Pending.
If previous INVITE transaction is not complete yet.
SIP: Call remains on 'sendonly' after transfer from WQ object to SCRIPT object
| Status | Closed |
| Id | 125084 |
Call remains on 'sendonly' after transfer from WQ object to SCRIPT object (vm).
Config Wizard did not work on IP302/IP305 if DHCP address was configured
| Status | Closed |
| Id | 125117 |
The last page of the wizard was hanging
myPBX: Wrong favourite type stored when adding the first favourite
| Status | Closed |
| Id | 125128 |
If the user does not have a favourite list, a list is automatically created when adding a favourite. In this case the favourite was stored with a wrong type (always internal).
myPBX: Hide DTMF keypad if the phone does not support it
| Status | Closed |
| Id | 125132 |
The DTMF keypad was also shown for phones that can't create remotely controlled DTMF.
SIP: Only one remote address was saved and used if host name was configured as proxy
| Status | Closed |
| Id | 125185 |
Only one remote address was saved and used if host name was configured for proxy.
No problem if network name was configured for proxy.
IP4: TCP sessions sometimes lost after changes under IP4/ETHn/IP although DHCP was activer for the interface
| Status | Closed |
| Id | 125186 |
ip38: improve ring-verification for non-standard ring voltage behaviour
| Status | Closed |
| Id | 125438 |
as seen on Alcatel-PABXes
SIP: Only two remote IP addresses were resolved from one proxy name
| Status | Closed |
| Id | 125663 |
Only two remote IP addresses were resolved from one proxy name.
If more than two IP addresses were configured for fallback reasons, SIP could stop working.
SIP: Bad encoding of Display Name
| Status | Closed |
| Id | 125690 |
Bad encoding of display name in To/From header if dn contains double quotes.
Fax server: Fast sending improved
| Status | Closed |
| Id | 125718 |
Fast sending of fax documents can result in confirmed but not sent documents. This is fixed now. Also time stamps in the log files are added.
SIP: Lock registration during call activity
| Status | Closed |
| Id | 126198 |
Lock registration during call activity.
Do not change back from secondary/fallback server to primary server during a call.
Interoperability issue on CUCM.
Memory leak in DTMF facility features
| Status | Closed |
| Id | 126234 |
Direct dial timeout could cause a memory leak.
SIP: Trap when processing SDP from iSoftPhone 3.6037
| Status | Closed |
| Id | 126262 |
Trap when processing SDP from iSoftPhone 3.6037
myPBX: Resync after receiving unexpected message did not work
| Status | Closed |
| Id | 126318 |
Instead there was a skript error (line 8161).
http configuration port determined during installtion disappears after first configuration
| Status | Closed |
| Id | 126336 |
Fax server: Error description for WebDAV read error changed
| Status | Closed |
| Id | 126345 |
The file read error description is changed so if a WebDAV read error occurs during a call, the fax application retries the call.
myPBX: Use configured IP address and gatekeeper ID for internal LDAP directory
| Status | Closed |
| Id | 121840 |
If values are given in the phone config, myPBX should also use them.
Testlicenses are displayed with count of 0
| Status | Closed |
| Id | 124844 |
The count in test licenes is now always 0 and the licenses are granted anyway. This way no consideration about the count to be used in test licenses is needed anymore.
Also v11 test licenses with expiration date are recognized as normal test licenses in v10.
V10 Service Release 15 (101196)
Changes included in Version 10 Service Release 15 Definition
SIP: Did not used mapped ports in case of "Restricted Cone" and "Port Restricted Cone"
| Status | Closed |
| Id | 126695 |
Did not used mapped ports in case of "Restricted Cone" and "Port Restricted Cone".
Only in case of NAT variation "Full Cone" a SIP interface used mapped ports.
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.
SIP: Memory leak when REGISTER messages are blocked by IP black list
| Status | Closed |
| Id | 126736 |
Memory leak when REGISTER messages are blocked by IP black list.
Objects of type "sip_client" are not freed.
phone: going on hook just after creating a consultation call by pressing the R-key dropped the held call
| Status | Closed |
| Id | 126863 |
Now when going onhook before any digit has been entered the held call will be kept. A blinking display line or LED indictes the existence of the held call. The call an be reconnected by going offhook again.
SIP: Fix for trap caused by fix for service release 14
| Status | Closed |
| Id | 126976 |
Fix for trap caused by fix #125068 "SIP: Do not reject reINVITE with 482 Loop Detected".
Video: h264 stream not decoded properly
| Status | Closed |
| Id | 127011 |
h264 start code can contain 2 or 3 null bytes. Only Polycom had problems with that.
PBX: Presence/Dialog Subscriptions could not be routed thru a Gateway object
| Status | Closed |
| Id | 127044 |
Presence or Dialog info from the Gateway object itself was returned
phone: inverse name resolution did not work for calling party number in international format as provided by SIP
| Status | Closed |
| Id | 127045 |
SIP may provide the calling party number in international format, for example as '+4930....'. The inverse name resolution did not work although correct entries were present in directory.
myPBX MSI parameters didn't always work
| Status | Closed |
| Id | 127060 |
Some boolean parameters didn't work with "false" and sometimes user settings have been overridden by MSI parameters.
RTP: Audio RTP packet with Marker bit set were not received
| Status | Closed |
| Id | 127081 |
The marker bit is set on the first packet after a silence persiod in case of silence compression, so this does not do a lot of harm if this packet is lost, except if the other side is setting this bit for every packet, which happens sometimes.
DTMF: Speed-up transmission of RTP-DTMF (RFC-2833)
| Status | Closed |
| Id | 127083 |
Speed-up transmission of RTP-DTMF (RFC-2833).
Transmission of an RTP-DTMF took 200ms time from first START event to last END event.
It now takes 120ms.
This allows faster DTMF sequences.
Remote Media: Support for UTF8 characters in hardware ID and realm
| Status | Closed |
| Id | 127087 |
When the hardware id of the selected device contained non-ASCII characters, the remote video codec could not register to the phone.
ISDN PRI: Call rejection in Alerting not possible if Interop option "No Disc" set
| Status | Closed |
| Id | 127108 |
With "No Disc", no call rejection was possible, but the idea of this option was not to use DISC, but RELEASE instead.
myPBX: Button for DTMF dialpad could be clicked if it was greyed out
| Status | Closed |
| Id | 127123 |
The following fix from v10sr14 was not complete:
Fix: #125132: myPBX: Hide DTMF keypad if the phone does not support it
The button for the dialpad was greyed out, but a click on it opened the dialpad anyway.
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: inverse name resolution did not work when the directory search returned entries not matching the filter expression
| Status | Closed |
| Id | 127138 |
When searching the ESTOS metadirectory for a plain number (without any wildcards) the result set contains all objects with a tail match in one of the number attributes specified in the search request. The number of objects to be returned in search response was set to one by the phone and thus it happened that the really matching objet was not received. To overcome this problem more objects are requested now and the returned objects are checked for a match in the phone again.
PBX Waiting: An operator joining should receive calls which are waiting at the time of the join
| Status | Closed |
| Id | 127191 |
After the join the call was not sent to the operator.
PBX Mobility: CFNR/CFB was not executed on call thru Mobility
| Status | Closed |
| Id | 127235 |
A call from a mobile phone calling in thru the mobility was not diverted by a CFNR or CFB.
PBX: Call completion did not work across nodes and pbxs in case of configs without escapes
| Status | Closed |
| Id | 127259 |
A call completion from a slave to a user on the master in a different node was rejected.
PBX: Group editor page empty after apply, for objects with empty PBX
| Status | Closed |
| Id | 127296 |
Save did work
SIP: Fix for trap
| Status | Closed |
| Id | 127435 |
Fix for trap
SIP: Fix for fax fallback
| Status | Closed |
| Id | 127465 |
Fix for handling of reject of re-INVITE for t38.
Only concerns fallback to encrypted audio.
SIP: re-INVITE sent with two media descriptions for t38
| Status | Closed |
| Id | 127609 |
re-INVITE sent with two media descriptions for t38.
PBX: Wrong display of called number on calls to local objects on slaves
| Status | Closed |
| Id | 127639 |
The node prefix of the slave was displayed.
PBX Waiting: DTMF forwarding call was sent with wrong transfering number
| Status | Closed |
| Id | 127684 |
This could lead to wrog routing in configurations with multiple PBXs and node numbers
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.
PBX Waiting: A CF configured at an operator, should not forward the call to the WQ
| Status | Closed |
| Id | 127696 |
Operator CFs are not executed for call to WQ.
RTP: Timeout for "No media" event increased from 4s to 10s
| Status | Closed |
| Id | 127815 |
4s was too short for some traffic cases and unnecessary events were generated
phone: old fashioned feature (updated) - automatically set up a call when a certain digit is entered when phone is idle
| Status | Closed |
| Id | 127865 |
This behaviour can be enabled via
config add PHONE APP /auto-handsfree <digits>
config add PHONE APP /auto-handsfree X<digits>
where <digits> is the sequence of all digits which shall trigger a call, for example
config add PHONE APP /auto-handsfree 0
config add PHONE APP /auto-handsfree X0
To disable this behaviour use
config rem PHONE APP /auto-handsfree
With the first form the call is always setup in handsfree mode.
With the second form the call is setup in the mode assumed to be most suitable.
When a call via handset was terminated by the disconnect key and a digit is entered
thereafter with the handset kept lifted the new call is set up in handset mode too.
If the handset is on hook the the call is set up in headset mode when the headset
is enabled, otherwise in handsfree mode.
PBX-SOAP: UserCall did not terminate correctly, when initial call out to endpoint or the transfer failed
| Status | Closed |
| Id | 127928 |
There was no message indicating this signaled back to the application. If the call was initiated for a entity on the Gateway (e.g. the FAX interface) it could lead to hanging calls on the Gateway.
phone: ip110,200,230,240: display for a connected inbund call redirected via a waiting queue was different to former versions
| Status | Closed |
| Id | 127972 |
Usually a connected call is displayed in two lines:
line 1: called party
line 2: connected party
For a connected call redirected via a waiting queue the diverting party (i.e. waiting queue number/name) was displayed in line 1 instead of the original called party as in former versions.
PBX: myPBX messages should not show in the admin log
| Status | Closed |
| Id | 127975 |
Could also be a performance issue with many myPBX clients
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.
PBX: Check for availability of some licenses did not work for Test licenses
| Status | Closed |
| Id | 128034 |
In case the user interface displayed the option only if the license was available this did not work for test licenses. This was the case for Session Border Objects.
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.
HTTPCLIENT: A request could fail if sent immediately after a request responded with "Connection: close" in the response header
| Status | Closed |
| Id | 128271 |
This happens only if both requests use the same URL
SIP: Ignore 'inactive' offer while on 'sendonly'
| Status | Closed |
| Id | 128304 |
Ignore SDP offer with 'inactive' attribute while on 'sendonly' (HOLD-NOTIFY).
Interworking issue with Cirpack/v4.58 (gw_sip).
PBX Map: Connected number was not displayed even if "Hide connected Number" was not set
| Status | Closed |
| Id | 128319 |
.
Add maximum length of internal numbers to dialing location
| Status | Closed |
| Id | 126377 |
If in myPBX a number is dialed from an external application (tel link, office integration, hotkey), myPBX needs to guess if the number is an external or an internal number. For external numbers that are not in international number format it needs to add the external line prefix.
This fix adds the maximum length of internal numbers to the dialing location. That information can then be used by myPBX to identify external numbers that are missing the external line prefix for dialing.
IP-DECT: DTMF tones with myPBX
| Status | Closed |
| Id | 127226 |
Now it is possible to send DTMF tones with the DECT handset by myPBX.
softwarephone: Jabra PRO 930 OC support included
| Status | Closed |
| Id | 127402 |
hid codes for the Jabra PRO 930 OC included
PBX CDRs: New info-from/info-to tags, to get dialed number
| Status | Closed |
| Id | 128053 |
The numbers in conn-from/conn-to could be adjusted by a received "connected number"
V10 Service Release 16 (101209)
Changes included in Version 10 Service Release 16 Definition
phone: transparent recording was not started for automatically connected inbound calls
| Status | Closed |
| Id | 128366 |
Inbound calls are automatically connected when the checkmark
"Phone/User-n/Preferences/Treat any Call as Announcement"
is set. For calls connected this way the recording modes "transparent" and "optional" did not work, i.e. recording was not started at connect time.
phone: ip222,232,ip241: no media on inbound calls automatically connected to headset after ringing
| Status | Closed |
| Id | 128408 |
This happened with the following checkmarks set:
"Phone/User-n/Preferences/Treat any Call as Announcement"
"Phone/Preferences/Play Configured Ring Melody before Automatically Connecting an Announcement Call"
"Phone/Preferences/Route Automatically Connected Inbound Calls to Headset (if enabled)"
After playing the ring melody for the configured time the connection was established but the headset was mute.
PBX Waiting: A called picked from a WQ operator was shown as missed
| Status | Closed |
| Id | 128670 |
I should not be indicated as a missed call, because it was answered by someone else.
PBX: A call-completion was executed even if the destination was still busy
| Status | Closed |
| Id | 128679 |
This happened when one call of a conference or in a consultation scenario was disconnected
PBX: Call completion was executed sometimes even if destination user was still busy
| Status | Closed |
| Id | 128811 |
Counting of calls was incorrect
Gateway: No voice for calls from one local interface to another, with direct route, no voip and use of TONE
| Status | Closed |
| Id | 128891 |
There was only a loud noise
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.
PBW Waiting: When calling an operator, the Display name not Long name should be used as diverting name
| Status | Closed |
| Id | 129012 |
On the called phone the operator gets displayed that the call was forwarded by a Waiting Queue. Here the display name of the Waiting Queue should be used when configured not the Long Name.
Web-Admin-UI: Client addresses are not displayed on Services/HTTP/Server
| Status | Closed |
| Id | 129394 |
Client addresses are not displayed in list of sessions on Services/HTTP/Server.
SIP: Problems on address resolution by DNS
| Status | Closed |
| Id | 129445 |
Problems on address resolution by DNS.
If SRV query delivers two hostnames but no IP addresses, two A queries are issued.
First A query is for primary hostname.
Second A query is for secondary hostname.
In case the A record for secondary hostname arrives before the A record for primary hostname,
the secondary host's IP address gets lost.
Only primary IP address is used.
Avoid X509_ALARM_SYSTEM_TIME_NOT_SET on boxes that do not do certificate validation
| Status | Closed |
| Id | 129479 |
Only set the alarm if an actual certificate validation failed because of the missing system time.
ip38: Polarity reversal causes illegal call at end of previous call
| Status | Closed |
| Id | 129616 |
When 'Establish call on polarity reversal' is checked and PSTN drives line in reversed polarity in conversation/connected state, an illegal call may be initiated at the end of the preceding call due to the polarity reversal at the end of the call.
Trap after changing the phone admin config without immediate restart
| Status | Closed |
| Id | 129718 |
The trap can be reproduced through these steps:
- Start with an account where only user name and password is specified
- Go to "Settings/Accounts/xxx/Login settings". Enter a primary address.
- OK, "Restart now?" -> No
- Go to "Login settings"
- OK, "Restart now?" -> Yes
IP232: Display timing fixed(3), sync to V11
| Status | Closed |
| Id | 129781 |
Display clock inverted, Sync inverted
Service/DNS/Query input field width now 20
| Status | Closed |
| Id | 129805 |
Was 24, which led to problems with Win/Chrome
PBX: Number display after transfer for calls accross nodes together with "Not to be called" nodes did not work
| Status | Closed |
| Id | 129865 |
It could happen that a number starting with a "not to be called" node was used.
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).
PBX Waiting: When a operator was deleted which there was an active call, the operator was called again
| Status | Closed |
| Id | 129915 |
There is a chance that this could have resulted in a trap
SIP: Fix for media negotiation
| Status | Closed |
| Id | 130007 |
Sometimes wrong codec was shown as selected in case of media-relay.
Trap in PBX during boot after downgrade from v11
| Status | Closed |
| Id | 130015 |
Trap in PBX during boot after downgrade from v11.
SIP: Bad SDP offer with invalid crypto key in rare cases
| Status | Closed |
| Id | 130045 |
Bad SDP offer with invalid crypto key in rare cases.
But only when re-routing an incoming call without offer.
PBX: Potential trap on subscriptions to user without name
| Status | Closed |
| Id | 130114 |
Null pointer access
SoftwarePhone: icollect cannot be run from command shell
| Status | Closed |
| Id | 130189 |
path to vbs script missing
SoftwarePhone:icollect hangs when the mypbx folder is empty
| Status | Closed |
| Id | 130190 |
SoftwarePhone: Pressing the ok button in audio and usb configuration tab resets to default "Please Select" values
| Status | Closed |
| Id | 130191 |
check for changed values missing
SIP: Federation interface should reject call with Q931_CAUSE_AddressIncomplete_InvalidNumberFormat
| Status | Closed |
| Id | 130280 |
Federation interface should reject call with Q931_CAUSE_AddressIncomplete_InvalidNumberFormat (not Q931_CAUSE_RequestedCircuit_ChannelNotAvailable)
if destination is not a URI (e.g. no domain part).
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.
ISDN: Reject calls for subscriptions with cause "Facility rejected"
| Status | Closed |
| Id | 130324 |
This allows the phone to decide that retry is useless
PBX Waiting: A trunk no-answer was executed even if the WQ was connected
| Status | Closed |
| Id | 130359 |
This happened if a operator phone was ringing
IP-DECT: Set HLC to type telephony
| Status | Closed |
| Id | 127228 |
The coding of High Layer Compatibility (HLC) information element of IP-DECT calls is set to type telephony now.
ip28: add values 0dB and -4dB as 'Receive Gain' choices
| Status | Closed |
| Id | 129531 |
SIP: New interop tweak "To Header when Sending INVITE"
| Status | Closed |
| Id | 129780 |
New interop tweak "To Header when Sending INVITE".
Used to specifiy how to populate the To header URI when sending INVITE.
Selection between "Called Party" or "Original Called Party".
For SIP carrier interoperability in case of forwarded/redirected calls.
V10 Service Release 17 (101227)
Changes included in Version 10 Service Release 17 Definition
SIP: Interworking issue with OpenStage systems
| Status | Closed |
| Id | 130551 |
Handling of X-Siemens-Call-Type was wrong if list of values was provided.
SIP: Bug in media negotiation during interop with Microsoft MediationServer
| Status | Closed |
| Id | 130724 |
Bug in media negotiation during interop with Microsoft MediationServer.
If received offer was ordered unordinary:
\tv=0
\to=- 148 1 IN IP4 172.31.210.31
\ts=session
\tc=IN IP4 172.31.210.31
\tb=CT:1000
\tt=0 0
\tm=audio 49606 RTP/AVP 97 101 13 0 8
\tc=IN IP4 172.31.210.31
\ta=rtcp:49607
\ta=label:Audio
\ta=sendrecv
\ta=rtpmap:97 RED/8000
\ta=rtpmap:101 telephone-event/8000
\ta=fmtp:101 0-16
\ta=rtpmap:13 CN/8000
\ta=rtpmap:0 PCMU/8000
\ta=rtpmap:8 PCMA/8000
\ta=ptime:20
PBX Routing: Call forward on a slave to a number of a different node did not work
| Status | Closed |
| Id | 130754 |
In case of a configuration without escapes
PBX Waiting: Mobile operator could not do transfer (with **)
| Status | Closed |
| Id | 130764 |
The DTMF digits were not received on the mobility object
PBX Waiting: Potential trap on group membership change
| Status | Closed |
| Id | 130778 |
Only on very specific timing
PBX: Trap on wrong config in user object
| Status | Closed |
| Id | 130801 |
Should not happen except when downgrading from a higher version.
PBX: Original called/diverted number got lost when a call was forwarded on a gateway object by CFNR
| Status | Closed |
| Id | 130835 |
A CFNR are a gateway object is executed when a call was terminated with a retry cause. This itself is not treated as diversion, so not diverting leg2 info is included in the call for that, but diverting leg2 infos from previous CFs should be preserved
SoftwarePhone: intermittend crash when notebook switches between sleep and active mode
| Status | Closed |
| Id | 130842 |
SoftwarePhone: debug.xml entries do not reflect the functionality
| Status | Closed |
| Id | 130927 |
PBX CDRs: No Records when a call was forwarded by a WQ because of busy
| Status | Closed |
| Id | 131334 |
There should be records the same way as it is for a CFU
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
SIP: Interop with snom phones regarding call-completion
| Status | Closed |
| Id | 131359 |
Ignore subscriptions for RFC-6910.
RFC-6910 is not supported yet.
Subscriptions for "dialog;purpose=call-completion" must be rejected.
DNS: Trap
| Status | Closed |
| Id | 131445 |
Two results could have been sent for a single request. Such a sequence occurred in situations were the encoding of the DNS request failed internally.
SIP: Trap when no ACK is received on an incoming call
| Status | Closed |
| Id | 131455 |
Trap when no ACK is received on an incoming call.
Only occured on Android platform.
PBX Broadcast, config parameters fixed
| Status | Closed |
| Id | 131492 |
This is a fix to a fix done in the last service release
SIP: Bug when decoding Reason header
| Status | Closed |
| Id | 131632 |
Bug when decoding Reason header, but only if multiple causes are provided.
E.g. Reason: Q.850;cause=17;text="user busy",SIP;cause=486;text="Busy"
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
PBX CDRs: Sometimes calls rejected as busy were missing in Reporting
| Status | Closed |
| Id | 131691 |
The more flag was set, even if no other broadcasted call was pending for this incoming call.
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.
SDP: At most one instance of "a=fmtp" is allowed for each format
| Status | Closed |
| Id | 131908 |
At most one instance of "a=fmtp" is allowed for each format.
This is illegal:
m=audio 60728 RTP/AVP 0 9 8 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexa=yes
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
"annexa" is now dropped from SDP, since there's no need for a negotiation:
m=audio 60728 RTP/AVP 0 9 8 18 101
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20
a=silenceSupp:off - - - -
a=sendrecv
phone: no in-band (default) call waiting tone is played while a call is recorded
| Status | Closed |
| Id | 131979 |
CF read alarm was not unset after successfull read
| Status | Closed |
| Id | 132001 |
If a CF read alarm has been raised, the alarm wasn't unset if another request has been successfull.
IP-DECT: Phone book requests with special characters
| Status | Closed |
| Id | 132037 |
Phone book requests with special characters are fixed now.
CONF Interface: Trap with signal release race condition
| Status | Closed |
| Id | 132093 |
A signal release race condition can cause a trap by the CONF interface. This is fixed now.
IP-DECT: Normalized loaded IPEIs to match for subscription
| Status | Closed |
| Id | 132096 |
If the handset's IPEI is inserted with a checksum, it doesn't match during the subscription process and the system AC is used instead of the user specific AC or the subscription isn't allowed and is rejected. This causes a for the user unobviously failed subscription and 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.
SIP: No need to start DNS timer, if no DNS names are to be resolved
| Status | Closed |
| Id | 132201 |
No need to start DNS timer, if no DNS names are to be resolved.
30sec timer was started witout a need.
<--sip.cpp-->
Wrong local media address selected
| Status | Closed |
| Id | 132235 |
Wrong local media address selected when sending INVITE.
Select local media address based on destination of INVITE.
ip38 : unable to detect china-style busy tone
| Status | Closed |
| Id | 132256 |
-
SIP: Must escape userpart of URI in message headers History-Info and Diversion
| Status | Closed |
| Id | 132303 |
Must escape userpart of URI in message headers History-Info and Diversion.
Bad:
Diversion: "Dummy" <sip:#999@ip800.innovaphone.compat;user=phone>;reason=unconditional
Good:
Diversion: "Dummy" <sip:%23999@ip800.innovaphone.compat;user=phone>;reason=unconditional
H.323: Potential buffer overrun, when receiving large SDP
| Status | Closed |
| Id | 132331 |
happened with Lync
Softwarephone: Jabra 9450 button functionality fixed
| Status | Closed |
| Id | 132381 |
Softwarephone: xml malformed when User Confguration tab is selected
| Status | Closed |
| Id | 132383 |
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.
PBX: Match v11 hardware ids to legacy hardware ids
| Status | Closed |
| Id | 130852 |
In version 11 the mac address is used as hardware id. In version 10 an earlier a combiation of the product short name and the last 3 bytes of the mac address.
A mapping of the mac address to the legacy hardware id was added to version 10 to make sure phones still register if the phoness are upgraded to version 11 before the PBX
IP-DECT: Handset UTF-8 support for phone book requests
| Status | Closed |
| Id | 132038 |
Support for UTF-8 phone book requests added if the handset provides it.
phone: ip222/232 - added support for Jabra BIZ 2300 USB Duo headset
| Status | Closed |
| Id | 132131 |
Softwarephone: new Selection "No Ringing device" added
| Status | Closed |
| Id | 132379 |
Allows to disable ringing for users who wish to have incoming calls signalled by cti only
V10 Service Release 18 (101231)
Changes included in Version 10 Service Release 18 Definition
TLS: Allow SSL record for initial ClientHello
| Status | Closed |
| Id | 132410 |
The server side should accept the initial ClientHello, even if it is sent in a SSL 3.0 record. Some TLS implementations, like OpenLDAP send ClientHello messages like that for backward compatibility.
The fix does not affect the negotiation of the used TLS version.
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.
SIP: Handling of "message-summary" changed
| Status | Closed |
| Id | 132462 |
If this special combination of lines was received the message lamp was deactivated:
Messages-Waiting: yes
voice-message: 0/0
Now this will turn on the lamp.
SIP: Too many DNS requests when STUN server is configured wrong
| Status | Closed |
| Id | 132478 |
Too many DNS requests when STUN server is configured wrong.
PBX: Potential buffer overrun, when many checkmarks set on Config/General
| Status | Closed |
| Id | 132485 |
Could happen at other places as well, but this is the page with most checkmarks
PBX: Trap because of buffer overrun on registration with very long productId
| Status | Closed |
| Id | 132627 |
A productId of approx 100 Characters caused the problem
PBX: On call forward to a name, if additional digits were dialed, the call was forwarded to the additional digits
| Status | Closed |
| Id | 132633 |
Instead to the configured name.
PBX: Potential Trap if user was updated with dialiog monitor to an unknown destination
| Status | Closed |
| Id | 132638 |
Initialiasation of data was missing.
IP4: don't try to send ICMP(IC_DESTUR) for packets arriving before the IP stack is configured
| Status | Closed |
| Id | 132658 |
When a box reboots remote peers may continue to send packets directed to the former IP-address of the box (for example TCP keepalives). As long as the IP stack is not configured such packets must be silently discarded. Trying to send an ICMP(IC_DESTUR) via normal routing fails because there is no route back and this generates misleading "No route to destination" error events. Sending an ICMP(IC_DESTUR) directly to source seems not really helpful because usually DHCP will assign the former address to the box again.
PBX: Memory leak with direct pickup
| Status | Closed |
| Id | 133170 |
Direct pickup is the pickup, which connects directly to the caller, without displaying the call as incoming on the picking phone
SIP: Read called party number (CDPN) from Request-URI
| Status | Closed |
| Id | 133807 |
When INVITE is received:
CDPN is taken from To-URI if the Request-URI matches the own local Contact-URI.
CDPN is taken from Request-URI otherwise.
PBX CDRs: New attribute root for the number of the other party relative to the numbering root
| Status | Closed |
| Id | 132680 |
If available
Register myPBX protocol handler capabilities on Windows 7
| Status | Closed |
| Id | 132797 |
This allows users to select myPBX as a protocol handler in their system control center.
V10 Service Release 19 (101248)
Changes included in Version 10 Service Release 19 Definition
PBX: Dyn Group In/Out function key was not updated on group membership changes with DTMF Feature Codes
| Status | Closed |
| Id | 133803 |
Notification to phone was missing
PBX Trunk: Retrying calls on all available registrations did not work as documented
| Status | Closed |
| Id | 133844 |
It could happen that one registration was not tried.
LDAP Expert: Search Size/Page Size reduced to 25. Form method="POST"
| Status | Closed |
| Id | 133876 |
Search size was 50. Form method was GET.
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.
H.323: Hanging calls in PBX if a call to Node Dialtone was canceled with certain timing
| Status | Closed |
| Id | 134083 |
Happen usually only in high load situations
Fax server: Synchronisation lost error
| Status | Closed |
| Id | 134246 |
A successfully received document is dropped with a synchronisation lost error if ECM is used and the remote terminal supports mode changing. This is fixed now.
SIP: Handling of 491 was wrong
| Status | Closed |
| Id | 134289 |
Re-transmission of re-INVITE did not come to an end.
SIP: Wrong payload type for 'telephone-event' in SDP answer
| Status | Closed |
| Id | 134384 |
Wrong payload type for 'telephone-event' in SDP answer.
Did not match the payload type in SDP offer.
Status:
Bug was on Media-Relay interfaces when DTMF was automatically added
while forwarding answer which originally did not had DTMF on it.
H.323: Early media not working for reverse media negotiation calls to a media-relay/exclusive coder interface
| Status | Closed |
| Id | 134434 |
Media was only after connect
Fax server: Received multiple page documents with EOM command
| Status | Closed |
| Id | 134441 |
If further pages are received after the EOM command instead of the normal MCF command, the pages are dropped. This is fixed now.
Video: add support for webcams with frame rate 60/1
| Status | Closed |
| Id | 134457 |
Trust webcam does not offer any allowed combinations (resolution, frame rate, aspect ratio), now 60/1 frame rates will be accepted.
Fax server: Weaker training condition
| Status | Closed |
| Id | 134461 |
The training condition is mitigated to fix connection errors with some remote devices.
(clone of #133807) SIP: Read called party number (CDPN) from Request-URI
| Status | Closed |
| Id | 134463 |
When INVITE is received:
CDPN is taken from To-URI if the Request-URI matches the own local Contact-URI.
CDPN is taken from Request-URI otherwise.
There was a bug in v10sr18:
Comparing local Contact-URI aganist received Request-URI needs to be done case-independent.
PBX: Memory leak, when deleting Executive, Bc Conference or Conference object
| Status | Closed |
| Id | 134637 |
The object which is used to store the object specific state way not deleted
ISDN: Bug on inbound CCNR (call completion on no-response) on PRI interfaces
| Status | Closed |
| Id | 134801 |
Bug on inbound CCNR (call completion on no-response).
A returnResult for CCBS was sent.
SIP: Wrong hostpart in URI's in header lines 'Diversion' and 'History-Info'
| Status | Closed |
| Id | 135001 |
Wrong hostpart in URI's in header lines 'Diversion' and 'History-Info'.
Remote domain was used instead of local domain when sending INVITE with 'Diversion' and 'History-Info'.
phone: directory access could not be locked via fine grained function locking (PHONE_LOCK_DIRECTORY)
| Status | Closed |
| Id | 135013 |
When "Phone/Protect/Functions to lock via PIN" is set to 0x80200044
any directory access should be blocked when the phone is locked. But when entering a letter or a digit a directory search was started possibly exposing internal information to an unauthorized user.
+0x80000000 - restrict phone access to emergency use
+0x00200000 - disable directory access
+0x00000040 - disable directory search on inbound calls
+0x00000004 - disable directory search on outbound calls
=0x80200044
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.
sometimes Alarm and Event Forward Server Type SYSLOG could not be configured
| Status | Closed |
| Id | 135054 |
The type SYSLOG is available on all gateways but sometimes this option was not offered in the WEB interface
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
Fix for dial tone for France (440Hz)
| Status | Closed |
| Id | 135114 |
Fix for dial tone for France (440Hz).
Was 425Hz before.
Phones: Updated text translations
| Status | Closed |
| Id | 135192 |
Updated text translations.
Import from text database.
Gateways FXS: Call Completion as DTMF facility did not work
| Status | Closed |
| Id | 135203 |
A call completion could be registered, but executions failed
SIP: Call gets stuck during rerouting of an inbound call with reverse negotiation
| Status | Closed |
| Id | 135215 |
Call gets stuck during rerouting of an inbound call with reverse negotiation.
If inbound call did not provide SDP offer.
SIP/TLS: Use existing inbound transport connection to send requests
| Status | Closed |
| Id | 135236 |
Use existing inbound transport connection to send requests.
Instead of opening outbound transport connection.
Already worked for SIP over TCP, but not in case of SIP over TLS.
SIP: Lost remote signaling port if a Contact-URI was received without remote signaling port
| Status | Closed |
| Id | 135257 |
If a Contact-URI was received without remote signaling port was received
the server side signaling (e.g. PBX) lost the remote signaling port.
E.g.
Contact: <sip:dummy1@IP800-PBX;opaque=urn:uuid:563E3713-BC5A-5EB0-93E4-a462f18c3cc4;gruu>
Trap: Flash Directory: LDAP Substring Search Caused MAX_BUSY_TICKS
| Status | Closed |
| Id | 135368 |
Consequtive asterisks weren't skipped.
PBX DTMF: Picking user did not get the correct peer displayed
| Status | Closed |
| Id | 135508 |
Instead the number dialed for the ickup (e.g. *0#) was still displayed
SIP: Trap if interop tweak "To Header when Sending INVITE" is set
| Status | Closed |
| Id | 135585 |
Collateral damage of
#129780: SIP: New interop tweak "To Header when Sending INVITE"
introduced in v10sr16.
SIP: Re-negotiation (t38 -> audio) was rejected
| Status | Closed |
| Id | 135610 |
Re-negotiation (t38 -> audio) was rejected with "488 Not Acceptable Here".
PBX Waiting: A call picked up from a WQ, did not disapear from the call list
| Status | Closed |
| Id | 135635 |
It was still displayed in the list of calls of applications monitoring the WQ (e.g. Operator) and could be picked again.
Trap caused by error handling of IPv6 UDP fragmentation
| Status | Closed |
| Id | 138210 |
Trap caused by error handling of IPv6 UDP fragmentation.
phone: ip222/232 - when a consultation call was opened while recording was active the dial/ringback tone was missing
| Status | Closed |
| Id | 138327 |
myPBX: For favourites use numbers as entered by the user
| Status | Closed |
| Id | 111054 |
When adding favourites currently the number from the presence monitoring is used. This is a problem for objects like the voicemail, where additional digits are used for addressing individual mailboxes. Those additional digits are cut in presence monitoring.
Gateway: Media Relay for Video
| Status | Closed |
| Id | 134435 |
For video calls to external endpoints (Federation)
ISDN: Receive old style Redirecting Number and treat as diverting leg2
| Status | Closed |
| Id | 134785 |
Some old ISDN networks use this information element instead of facilities
SIP: New config file option /always-send-100-trying
| Status | Closed |
| Id | 135193 |
New config file option /always-send-100-trying.
For compatibility to SIP carrier in finnland.
V10 Service Release 20 (101264)
Changes included in Version 10 Service Release 20 Definition
PBX Diversion filters did not always take diverting endpoints into account
| Status | Closed |
| Id | 138465 |
For example the was a problem with a call thru a Map object and a CFNR with Only <this Map Object>
Avoid hanging calls after failed fax calls
| Status | Closed |
| Id | 138546 |
If fax calls failed in some special ways, it could happen that calls were hanging in the gateway with the FAX interface.
H.323: No Media, if G.729B was the only coder in an offer
| Status | Closed |
| Id | 138834 |
The coder was removed if silence compression was not enabled and no coder was left. This does not make sense, even if we do not do silence compression, we should signal, that we are able to receive G.729B data.
SIP: Losing registration of a SIP interface if another gateway interface is reconfigured
| Status | Closed |
| Id | 138888 |
Losing registration of a SIP interface if another gateway interface is reconfigured.
SIP stack takes IP address for domain name, starts DNS query, gets NX_DOMAIN response and reports REG-DOWN.
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.
TLS: ClientHello v2 decode error
| Status | Closed |
| Id | 138961 |
There was an error when decoding the length field of ClientHello messages in SSL 2.0 format, that is sent by some clients to provide downward compatibility with historic SSL 2.0 servers. This led to a DECODE_ERROR alert when the ClientHelloV2 was bigger than 127 bytes.
Registration using MAC address did not work for some IP240 and IP110 phones
| Status | Closed |
| Id | 139391 |
The mapping of MAC address to the legacy HW-ID was incorrect for some address ranges. Therefore some IP240 and IP110 phones could not register using MAC address.
SIP: Media re-negotiation fails in some cases
| Status | Closed |
| Id | 139506 |
Media re-negotiation fails in some cases.
SIP: Problem with reuse of inbound TLS connections
| Status | Closed |
| Id | 139546 |
Problem with reuse of inbound TLS connections.
Occurs when remote SIP client restarts without de-registering.
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.
IP222/232: Display was vertically out of center
| Status | Closed |
| Id | 140201 |
Display was vertically out of center.
Select myPBX as presence provider on first configuration
| Status | Closed |
| Id | 140264 |
If myPBX has been installed by a logged administrator account and not inside the user account, the default presence provider hasn't been myPBX.
This has been changed now, if no default presence provider has been given by an MSI parameter.
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.
Setup Wizard: Configure Gateway page broken
| Status | Closed |
| Id | 140583 |
Input field for trunk number strangely positioned. IE and Chrome only.
Update of phone strings (different languages)
| Status | Closed |
| Id | 138958 |
Update of phone strings (different languages)
PBX Trunk: Improvemment of fake-connect
| Status | Closed |
| Id | 140081 |
The fake-connect feature is useful to avoid timeouts for calls coming in from a trunk, which may take a long time until there is an alert, because they are diverted back to the PSTN.
Fake-Connect works if there is in-band info. If there is no in-band info, now an alert is sent on this call in this case.
V10 Service Release 21 (101280)
Changes included in Version 10 Service Release 21 Definition
SIP: Wrong CDPN in incoming calls
| Status | Closed |
| Id | 139726 |
Fix for a change introduced in sr18.
Gateway: For mapping of diverting number only the matching map of the calling party number was used
| Status | Closed |
| Id | 140729 |
This logic was not useful. Now all calling party number maps are evaluated for the diverting number also.
Gateway: An emergency call should not disconnect another emergency call
| Status | Closed |
| Id | 140899 |
Should search for non-emergency calls to disconnect and should fail if only emergency calls
PBX Session Border Object: Sending DTMF with myPBX did not work for phones registered thru the SBC
| Status | Closed |
| Id | 140914 |
The flag to indicate DTMF support was not forwarded to the PBX
NAT: Local addr for outgoing packets was set wrong in case packet matched an inbound mapping an was not sent to ETH0
| Status | Closed |
| Id | 141221 |
The local addr was determined based on the remote addr in the mapping, which was 0.0.0.0
SIP-IF config got lost if number mapping was changed
| Status | Closed |
| Id | 141222 |
SIP-IF config was damaged if number mapping was changed.
Video: library crashes if empty NAL arrives
| Status | Closed |
| Id | 141345 |
There was no checking for empty NALs.
SIP/SDP: Ignore PCMA/PCMU with vbd=yes
| Status | Closed |
| Id | 141385 |
E.g.
v=0
o=HuaweiSoftX3 000 20657322 20657324 IN IP4 213.148.136.178
s=Sip Call
c=IN IP4 213.148.136.178
t=0 0
m=audio 55436 RTP/AVP 8 102 0 127 101
a=rtpmap:8 PCMA/8000
a=rtpmap:102 PCMA/8000
a=gpmd:102 vbd=yes
a=rtpmap:0 PCMU/8000
a=rtpmap:127 PCMU/8000
a=gpmd:127 vbd=yes
a=rtpmap:101 telephone-event/8000
a=X-modem
a=fmtp:101 0-15
IP222 IP232 IP241: First RTP packet sometimes has the wrong timestamp
| Status | Closed |
| Id | 141390 |
One packet of the previous rtp stream with an "old" timestamp" was sent.
'Idle Reset' via WEB UI or 'ireset' command must not reset before all log messages and CDRs are sent
| Status | Closed |
| Id | 141482 |
SIP: SIP-IF fails to register if initial DNS query fails to resolve domain name
| Status | Closed |
| Id | 141651 |
SIP-IF fails to register if initial DNS query fails to resolve domain name.
No retry of DNS.
PBX: Favorites were sent to the phone, even if "Store phone confg", was not configured
| Status | Closed |
| Id | 141748 |
If "Store phone config" is not set, the favorites are configured locally on the phone.
IP-DECT: Node support
| Status | Closed |
| Id | 141777 |
IP-DECT base stations support duplicate numbers with different nodes now.
Box could trap during boot when there are plenty of licenses installed
| Status | Closed |
| Id | 141892 |
Box could trap during boot when there are plenty of licenses installed.
IP222 IP232 IP241: First RTP packet sometimes has the wrong timestamp -2-
| Status | Closed |
| Id | 142042 |
One packet of the previous rtp stream with an "old" timestamp" was sent.
SIP: STUN not always used to map RTP ports
| Status | Closed |
| Id | 142047 |
STUN not used to map RTP ports, if NAT type discovery failed.
H.323: Potential trap on unusual call clearing
| Status | Closed |
| Id | 142267 |
An assertion because of duplicate RELEASE message from stack could happen. Circumstances under which this could happen unclear, but very rare.
PBX: OEM Registration licenses did not work anymore
| Status | Closed |
| Id | 142371 |
New handling of license versions broke the OEM licenses
Voicemail: Freeing resources for <store-get-msgcount> asap
| Status | Closed |
| Id | 142523 |
Reducing memory footprint in large-scale installations
PBX SOAP: Possible mixup with two simultaneous UserCall opertions
| Status | Closed |
| Id | 142854 |
It could happen, that the same outgoing call was actually indicated on both UserCall operations
IP222 IP232 IP241: First RTP packet sometimes has the wrong timestamp -3-
| Status | Closed |
| Id | 142857 |
One packet of the previous rtp stream with an "old" timestamp" was sent.
PBX Waiting: No missed call at operator, if CFNR at waiting queue
| Status | Closed |
| Id | 142894 |
The call was cleared as if it was accepted somewhere else
myPBX: "Start minimized" didn't work
| Status | Closed |
| Id | 142925 |
When the option "Show in taskbar" was disabled, the option "Start minimized" did not work.
<--
ctiwin.cs
-->
SIP: Missing angle brackets around To-URI
| Status | Closed |
| Id | 142972 |
Missing angle brackets around To-URI in INVITE and PUBLISH.
SIP: Validate SIP Request-Line before go on parsing
| Status | Closed |
| Id | 143002 |
Validate SIP Request-Line before go on parsing.
Ignore badly formated requests.
Phones: Cut off leading whitespace from dial string
| Status | Closed |
| Id | 143006 |
IP241/240/230/200/110
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.
PBX CDRs: Tranfered calls thru a Broadcast object were not shown correctly in CDRs from the Broadcast object
| Status | Closed |
| Id | 143307 |
The rel-to/from events were missing so reporting classified these as incomplete
myPBX: New innovaphone logo
| Status | Closed |
| Id | 145273 |
The claim was changed from "PURE IP-COMMUNICATIONS" to "PURE IP COMMUNICATIONS".
V10 Service Release 22 (101294)
Changes included in Version 10 Service Release 22 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.
Refresh myPBX call list if last call couldn't be found inside call list update
| Status | Closed |
| Id | 142923 |
If a CDR isn't processed within the 2 second timeout of myPBX, the returning call list won't contain the last call.
SIP: Trap when call is terminated while inbound INVITE transaction is in progress
| Status | Closed |
| Id | 143631 |
Trap when call is terminated while inbound INVITE transaction is in progress.
PBX Session Border Object: All endpoints were deregistered when one Session Border object was modified
| Status | Closed |
| Id | 143668 |
This also cleared current calls
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
SIP: UPDATE request was sent with wrong Session-Expires header
| Status | Closed |
| Id | 143765 |
UPDATE request was sent with wrong Session-Expires header.
Value of refresher-param was wrong ("uac" instead of "uas").
IP-DECT: Support for new IP1202
| Status | Closed |
| Id | 143904 |
Support for new IP1202 is added.
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.)
PBX: Potential rare trap on disconnect with mobility
| Status | Closed |
| Id | 144262 |
A loop of sending SIG_REL messages could happen
PBX: Provider license did not work anymore
| Status | Closed |
| Id | 144270 |
A provider license of the form <lic-type>@<num> allows the use of %<num> licenses from the start
RTP: Threshold for wrong payload type event increased from 10 to 50
| Status | Closed |
| Id | 144272 |
This allows for a minimum of 1s of wrong traffic
Fax Server: Handle file write errors, if file writing fails on WebDAV with "401 Unauthorized"
| Status | Closed |
| Id | 144376 |
Handle file write errors, if file writing fails on WebDAV with "401 Unauthorized".
FAX Interface: Faster file close to recognize a "401 Unauthorized" write error
| Status | Closed |
| Id | 144594 |
The Webdav file is closed immediately if the T.30 allows it, so a write error with "401 Unauthorized" can be forwarded and the connection closed before the fax is confirmed.
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"
Prevent duplicate calllist entries on use of myPBXDial.exe
| Status | Closed |
| Id | 144768 |
Sometimes multiple calllist entries have been created on use of myPBXDial.exe on a terminal server.
myPBX: Case insensitivity for attribute names in LDAP search results
| Status | Closed |
| Id | 144962 |
If the directory returned attribute names that were in a different case than configured in myPBX, the search result was discarded.
SIP: Wrong IP address in Contact-URI
| Status | Closed |
| Id | 144975 |
Wrong IP address in Contact-URI.
Used mapped ip address in Contact-URI although no STUN server configure at the interface.
SIP: PBX sends re-INVITE after REFER was handled
| Status | Closed |
| Id | 145005 |
PBX sends re-INVITE after REFER was handled.
Client expects to receive nothing but NOTIFY(sip-frag) or BYE from PBX.
Gateway: Mapping of diverted number did not work, if a calling number was mapped as well
| Status | Closed |
| Id | 145035 |
Some digits could be duplicated.
SIP: Adjust offered framesize in media offers on media-relay interfaces
| Status | Closed |
| Id | 145079 |
Adjust offered framesize in media offers on media-relay interfaces.
Until now the original offered framesize was passed through.
Status:
Fixed in 10.00, 11.00, 11r2, 12r1
Updated innovaphone banner in setup dialog
| Status | Closed |
| Id | 145106 |
The banner has slightly changed.
V10 Service Release 23 (101298)
Changes included in Version 10 Service Release 23 Definition
DHCP-Client:: a changed "IP Routing" option propagated via the server "Renew" button had no effect at the client
| Status | Closed |
| Id | 145284 |
PBX SOAP: 6s timeout to pickup receiver if call is initiated for analog phone
| Status | Closed |
| Id | 145387 |
When a call is initiated for an anlog phone with SOAP, or any other phone, which cannot be made to accept a call atomatically, first a call rings at this phone. After accepting this call, the outgoing call is initaited. There was a timeout of 6s to accept this call. It is now increased to 60s
LDAP: Trap in Flash Directory UI
| Status | Closed |
| Id | 145405 |
A deleted memory region was re-accessed.
PBX: Connected number was not adjusted after SOAP pickup
| Status | Closed |
| Id | 145427 |
The call looked as if connected to the original called number
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.
SIP: Coder preference not always applied
| Status | Closed |
| Id | 145867 |
Coder preference not always applied.
Licenses containing digits (e.g. G729channel) did not work
| Status | Closed |
| Id | 146486 |
Problem parsing the license string
Fix for trap if invalid coder config is received by DHCP
| Status | Closed |
| Id | 146490 |
Trap if invalid coder config is received by DHCP.
PBX CDRs: CDRs from a Broadcast object was incomplete if the caller did a transfer
| Status | Closed |
| Id | 146609 |
The rel-to/from and conn-from events were missing. No calculation of call duration could be done.
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
SIP: SIP interface should reject call with Q931_CAUSE_RequestedCircuit_ChannelNotAvailable
| Status | Closed |
| Id | 147234 |
SIP interface should reject call with Q931_CAUSE_RequestedCircuit_ChannelNotAvailable (not Q931_CAUSE_AddressIncomplete_InvalidNumberFormat)
if remote proxy is currently not available ("down").
V10 Service Release 24 (101310)
Changes included in Version 10 Service Release 24 Definition
Voicemail: Name Display missing within myPBX
| Status | Closed |
| Id | 140773 |
MyPBX displayed number info only when calling a voicemail object.
Fax server: Raw data trace option added
| Status | Closed |
| Id | 146596 |
There is a configuration option (/dtrace) for raw data tracing available now.
PBX: Adjust any call from an User/Executive endpoint to a speech bearer capability
| Status | Closed |
| Id | 146602 |
For compatibility with some ISDN phones
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).
myPBX: Display H.323 ID in history if there is no display name
| Status | Closed |
| Id | 147471 |
If the reporting gave no display name for the remote party of a call, "Unknown" was displayed. In that case the H.323 ID is now displayed, if present.
SIP: REGISTER rejected with "301 Moved Permanently"
| Status | Closed |
| Id | 147871 |
REGISTER gets rejected with "301 Moved Permanently"
if TCP or TLS is used as transport protocol for SIP,
but Contact-URI in REGISTER misses corresponding "transport" parameter.
myPBX dial trace didn't work correctly
| Status | Closed |
| Id | 148235 |
The trace file hasn't been written if tracing has been enabled by its MSI property.
CONF: Connected to a wrong conference room
| Status | Closed |
| Id | 148318 |
With block dialing without any number the conference is assigned to a wrong existing room. This is fixed now.
myPBX for Android could not obtain license from a v10 PBX
| Status | Closed |
| Id | 148538 |
License mechanism merged to v10
ISDN: Send Proigress Indicator "Originator is not ISDN" with audio calls
| Status | Closed |
| Id | 148574 |
A SIP call can only be mapped to audio on ISDN, because we do not know, if it will be fax. Some ISDN phones do not accept an audio call without the Progress Indicator "Originiator is not ISDN" because they assume it must be fax or modem
Fixed myPBXDial crashes on terminal server
| Status | Closed |
| Id | 148584 |
Some myPBX processes have been found on terminal servers, which were not accessible, causing myPBXDial to crash.
PBX: Add additional dialed digits to the call forward destination only in case of call forward to number
| Status | Closed |
| Id | 148607 |
The additional dialed digits were added to an empty number and the resulting destination was wrong
SIP: SDP offer with "vbd=yes" was rejected with 488
| Status | Closed |
| Id | 148803 |
Better to ignore "vbd=yes" attribute and accept as regular PCMA offer:
\tv=0
\to=AudiocodesGW 1243985021 1243984779 IN IP4 195.34.155.139
\ts=Phone-Call
\tc=IN IP4 195.34.155.139
\tt=0 0
\tm=audio 56814 RTP/AVP 8 101
\tc=IN IP4 195.34.155.139
\ta=rtpmap:8 PCMA/8000
\ta=gpmd:8 vbd=yes;ecan=off
\ta=rtpmap:101 telephone-event/8000
\ta=fmtp:101 0-15
\ta=ptime:20
\ta=sendrecv
myPBX URI should be case-independent
| Status | Closed |
| Id | 149495 |
The following things did not work correctly, if the URL was not in tht right case:
* Video
* Application Sharing
* WebRTC Softwarephone
Now the case of the URI doesn't matter any more.
PBX Broadcast: Potential trap if call of Broadcast did a transfer
| Status | Closed |
| Id | 149651 |
Collateral damage of Fix 146609: PBX CDRs: CDRs from a Broadcast object was incomplete if the caller did a transfer
Local time derived from UTC timestamps sometimes wrong.
| Status | Closed |
| Id | 149731 |
The UTC timestamp was adjusted by the time offset of the current time period, i.e. by the Daylight Saving Time offset or the non Daylight Saving Time offset. Thus the local time displayed for a timestamp taken in summertime was displayed wrong in wintertime and vice versa.
PBX Gateway Object: Outgoing Calls no Name/URL
| Status | Closed |
| Id | 149966 |
To supress internal information to be sent to other systems
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
PBX Waiting: Original called number got lost on diverted calls to a Waiting Queue
| Status | Closed |
| Id | 150188 |
This happens if the call was diverted more then once before the call is sent to the WQ. In this case the original called number should be displayed on the phone rather then the last diverting.
PBX: Append additional dialed digits to call forward destination for GW type destinations only
| Status | Closed |
| Id | 150321 |
This function could be abused by users
timestamps used in event logging could be wrong when setting of system time was delayed
| Status | Closed |
| Id | 150326 |
-
timestamps set by logger could be wrong when the system time was set delayed after boot
| Status | Closed |
| Id | 150327 |
-
unsent log data was not freed when the log server shadow was disabled
| Status | Closed |
| Id | 150334 |
Admin UI: Make SHA256 the default signing algorithm for certificates
| Status | Closed |
| Id | 149545 |
Change default value in drop-down menus for creating certificates
* Signature: SHA256
Use SHA256 for automatically created certificates
| Status | Closed |
| Id | 150332 |
Certificates that are created without any user interaction were created using SHA1. Now SHA256 is used.
V10 Service Release 25 (101324)
Changes included in Version 10 Service Release 25 Definition
Voicemail: Default of '$_divconn' is now 'false'
| Status | Closed |
| Id | 140832 |
was 'true'.
Voicemail: Sending Connected Number Within H.323 CONNECT Message
| Status | Closed |
| Id | 143478 |
The connected number as configured and with a numbering-plan=private
PBX Session Border Object: Deleting one Session Border object clears registrations/calls on all Session Border objects
| Status | Closed |
| Id | 145375 |
Happens only if the Session Border Object, which is deleted has active registrations
PBX Trunk: "No Presence/Dialog Subscribe" did not work for local subscriptions from myPBX
| Status | Closed |
| Id | 146875 |
These subscriptions were still sent out
PBX Mobility: Unexpected restart on very unlikely call clearing collision
| Status | Closed |
| Id | 150530 |
Missing null pointer check
phone: cc-exec-possible indications for a pending call completion lost in some cases when sent to a busy phone
| Status | Closed |
| Id | 150645 |
- always when call-waiting was disabled on the phone
- when the phone was put on hook to terminate the active call
<!- app_ctl.cpp app_cc.cpp -->
SIP: Wrong expires parameter in 200/OK for REGISTER
| Status | Closed |
| Id | 151067 |
Wrong expires parameter in Contact header in 200/OK for REGISTER, but only if in case of multiple bindings.
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.
Voicemail: Duplicate Leak Checks
| Status | Closed |
| Id | 151343 |
Occurred within regression tests
Refresh the NAT mapping also for packets from outside to inside
| Status | Closed |
| Id | 151387 |
NAT mappings were only refreshed for packets from inside to outside. This could cause loss of the media stream if silence compression was enabled or if ICE selected different routes for the forth and back traffic. Therefore refresh the mapping also for packets from outside to inside.
SIP: Try to handle offer/offer-collision
| Status | Closed |
| Id | 151819 |
Try to handle offer/offer-collision.
1. Send re-INVITE with t38 -> rejected with 491
2. Receive re-INVITE with t38 -> rejected with 488
Better handle as offer/offer-collision and send 200/OK instead of 488.
PBX Waiting: Set Operator presence did not work correctly
| Status | Closed |
| Id | 151930 |
Presence was sometimes reset before the configured timeout
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 Broadcast Conference: Call to WQ not closed
| Status | Closed |
| Id | 152074 |
If the PBX Broadcast Conference calls a PBX Waiting Queue, the call isn't recognized as closed at the end of an announcement. This causes that the Waiting Queue isn't called again. It is fixed now.
PBX Exec: Call was sent to secretary even if a CFU was set
| Status | Closed |
| Id | 152082 |
In case the CFU destination was busy, because of Busy on ... Calls. The caller should get busy instead.
PBX: Registrations on multiple users sometimes lost, when user objects were changed
| Status | Closed |
| Id | 152091 |
A registration for multiple ussers is used for example to register multiple FXS interfaces to different users. The changes could be things like presence of CF updates.
PBX Waiting: A call parked at an operator was regarded as active call
| Status | Closed |
| Id | 152095 |
The operator was then regarded as busy
PBX Waiting: A call parked at an operator was regarded as active call
| Status | Closed |
| Id | 152095 |
The operator was then regarded as busy
Session Border Registrations were lost, if a "License only" registration at the master was re-established
| Status | Closed |
| Id | 152107 |
Happend for example when the license master was restarted
Session Border Registrations were lost, if a "License only" registration at the master was re-established
| Status | Closed |
| Id | 152107 |
Happend for example when the license master was restarted
phone: ip222/232/241: accept packets from PC-link immediately after physical link-up
| Status | Closed |
| Id | 152146 |
If the PC link is enabled per configuration the PC-port of the switch is now kept in forwarding state independent of the physical link state.
If the PC link is disabled per configuration the PC-port of the switch is set to disabled state.
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.
PBX SOAP: LocationUrl broken, if standby slave takes over
| Status | Closed |
| Id | 152554 |
The URL contained the expession (NULL).
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.
V10 Service Release 26 (101328)
Changes included in Version 10 Service Release 26 Definition
register for notification on changed variables only once
| Status | Closed |
| Id | 152842 |
Logging of PBX SOAP Admin requests resulted in broken log messages
| Status | Closed |
| Id | 153261 |
The text contained NUL characters und no XML data as it should
PBX Number Map; Call was forwarded with diverting leg2 info
| Status | Closed |
| Id | 153279 |
The call thru a Number Objekt appeared at the called endpoint as a call diverted by the Number Map. This caused problems, when e.g. a Voicemail was called. The Number Map should be transparent for the called endpoint.
Reduced the sidetone gain on IP222
| Status | Closed |
| Id | 153490 |
The sidetone was perceived as too strong on IP222. Reduced it by 6 dB through different balancing of analog and digital mic gain.
Gateway: A route with the matching number terminated with '!' should cut off any following digits
| Status | Closed |
| Id | 153837 |
This worked fine for enblock calls, but not for overlap dialing.
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.
SIP: Trouble handling SDP offer with "vbd=yes"
| Status | Closed |
| Id | 153977 |
Trouble handling SDP offer with "vbd=yes".
E.g.
\tm=audio 43028 RTP/AVP 8 18 100 118 110 96
\ta=rtpmap:8 PCMA/8000
\ta=fmtp:8 vad=no
\ta=rtpmap:18 G729/8000
\ta=fmtp:18 annexb=no
\ta=rtpmap:100 telephone-event/8000
\ta=fmtp:100 0-15
\ta=rtpmap:118 PCMA/8000
\ta=gpmd:118 vbd=yes
\ta=rtpmap:110 PCMU/8000
\ta=gpmd:110 vbd=yes
\ta=rtpmap:96 CLEARMODE/8000
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.
myPBX: Allow non-breaking-spaces in phone numbers
| Status | Closed |
| Id | 154206 |
Phone numbers from Outlook can contain non-breaking-spaces. For example this happens with contacts that are synchronized from an iPhone.
Show calls with CFNR to another user as missed call in the myPBX call list
| Status | Closed |
| Id | 154442 |
These CFNR calls are now shown as missed call in the myPBX call list.
The myPBX and phone call list now behaves the same.
Such calls are also missed if the user, to which the CFNR pointed, connects the call.
SIP: Trap on calls with very long phone number
| Status | Closed |
| Id | 154470 |
Trap on calls with very long phone number.
Status:
Fixed in 10.00, 11.00, 11r2, 12r1
PBX CDRs: No info-from, info-to events after conn
| Status | Closed |
| Id | 154844 |
These carry no information at all, and could increase the volume of the CDRs significantly. They could be generated in case of AOC information received from some ISDN/SIP providers
IP-DECT: MWI update with handset change (login feature)
| Status | Closed |
| Id | 156971 |
If the handset is changed with the login feature, the MWI isn't updated correctly. This is fixed now.
IP2x P30x IPxx10: Tone is sometimes not switched off
| Status | Closed |
| Id | 153129 |
-
V10 Service Release 27 (101336)
Changes included in Version 10 Service Release 27 Definition
myPBX Hotkey: Use Windows Automation and STRG-C instead of WM_COPY
| Status | Closed |
| Id | 145040 |
myPBX Hotkey: Use ClipboardFormatListener instead of ClipboardViewer
| Status | Closed |
| Id | 145316 |
Use a different windows API that is more robust.
PBX Waiting: Remote number wrong after round robin recall, if transfer had happend on incoming call
| Status | Closed |
| Id | 154709 |
For example if a consultation call is made to the WQ and the then the call is transfered, the remote number on the operator phone changes from the phone used for the consultation call to the original caller. After round robin, the phone used for the consultation is displayed again as remote number
Trap while reading kerberos config after upgrade from v9
| Status | Closed |
| Id | 155804 |
Boxes with version 10 or higher could trap while starting after upgrade from version 9, if kerberos was configured.
PBX: Voicemal: Wrong connected number sent, in case VM was 'local' object
| Status | Closed |
| Id | 155881 |
The caller got a display of the VMs node number, which is not desired for 'local' objects.
SIP: Trap when parsing presence XML with many presence/tuple elements
| Status | Closed |
| Id | 156102 |
Trap when parsing presence XML with more than 5 presence/tuple elements.
ip28: incorrect measurement of pulse dial pulse length
| Status | Closed |
| Id | 156110 |
ip28 pulse dial measured the pulse length as 10ms too long. In some cases this crossed the threshold of 80ms and detected a hook-flash instead of a digit.
PBX: Wrong number display during ringback on diversion to a local object
| Status | Closed |
| Id | 156116 |
The number was displayed containing node prefixes
PBX/Quick Dial: Consider General Checkmark "Hide Connected Endpoint"
| Status | Closed |
| Id | 156598 |
Alpha display information was erronously generated, regardless of the setting of the checkmark named "Hide Connected Endpoint.
myPBX: Redirect to another PBX using HTTPS
| Status | Closed |
| Id | 154659 |
myPBX always redirected to an HTTP URI. Now the redirect keeps the current protocol.
V10 Service Release 28 (101344)
Changes included in Version 10 Service Release 28 Definition
SIP: Bug when handling REGISTER from same addr/port for different users with same Contact-URI
| Status | Closed |
| Id | 158382 |
Bug when handling REGISTER from same addr/port for different users with same Contact-URI.
Seconds REGISTER just got 200/OK without any processing.
IP-DECT: Release code for unconnected calls to radio
| Status | Closed |
| Id | 158502 |
If a gatekeeper call isn't alerted or connected by a handset and is released again, the release code isn't forwarded to the radio. This is fixed now.
phone: ip222,ip232,ip112: support additional product IDs for Plantronics Savi 740 and Jabra Pro 9460
| Status | Closed |
| Id | 158871 |
IP-DECT: Release string added for IP1202
| Status | Closed |
| Id | 159534 |
The release string is missed on the IP1202. This is fixed now.
Admin UI: Truncated Kerberos host name after config changes in CMD0
| Status | Closed |
| Id | 160137 |
When changing the configuration of CMD0, in some cases the host name of the box was erroneously truncated to the length of the realm name.
SIP: Interop with Jitsi client
| Status | Closed |
| Id | 158355 |
Adding "Jitsi-Conference-Room: xxx" to INVITE.
V10 Service Release 29 (101351)
Changes included in Version 10 Service Release 29 Definition
TLS: Verifying of RSA signatures didn't always work
| Status | Closed |
| Id | 160354 |
If the signature of a certificate started with a null byte the verification could fail in some special cases.
PBX Trunk: Option to discard Diverting info received with incoming calls
| Status | Closed |
| Id | 160866 |
Diverting Info from a provider is sometimes not desired
IPXX10: Flash Directory Space Increased To 16MB
| Status | Closed |
| Id | 160906 |
Was 8MB
PBX: Partnerkeys with Group Indications, did not show outgoing number in case of block dialing
| Status | Closed |
| Id | 161013 |
With overlap dialing it was ok.
myPBX Android: For H.323/TLS one way audio with peers that do not support ICE
| Status | Closed |
| Id | 162284 |
For H.323/TLS no default local IP address was reported for the media and thus resulted in one way audio if the peer didn't support ICE.
ip38: possible trap if received FSK CallerID information corrupt
| Status | Closed |
| Id | 162349 |
if the lenghth field of a FSK CallerID has values above 128, an internal counter may overflow and cause an endless loop.
myPBX for Android: Hook button of cable headsets not taking effect
| Status | Closed |
| Id | 162685 |
The hook switch button on cable headsets was not taking effect on myPBX Android. This button should allow to accept incoming calls and hang up active connections.
phone: ip222,ip232: USB headset mute when a call was released by remote peer and a new call was signalled imediately thereafter
| Status | Closed |
| Id | 162963 |
FAX: Judged training failure in some cases where TCF was well acceptable
| Status | Closed |
| Id | 163056 |
During FAX reception noise patterns with alternating good and bad bytes at the beginning or end of the TCF were judged as training failures even though the pattern was good for a sufficient interval.
Fax server: Mode bit check removed
| Status | Closed |
| Id | 163240 |
The mode bit check is removed because of non-compliant remote devices.
Fax server: Maximum frame timeout increased
| Status | Closed |
| Id | 163244 |
The maximum frame timeout is increased for compatibility issues.
V10 Service Release 30 (101353)
Changes included in Version 10 Service Release 30 Definition
DHCP: on a change from disabled to client mode without reboot the received lease parameters were not propagated to IP stack
| Status | Closed |
| Id | 163580 |
Trap when option "Outgoing Calls No Name" is set on PBX object
| Status | Closed |
| Id | 163968 |
Trap when option "Outgoing Calls No Name" is set on PBX object.
Available on objects of type "Gateway".
phone: a cc-exec-possible sent to a busy phone got lost when the active call was relased by myPBX
| Status | Closed |
| Id | 164084 |
-
phone: ip222,ip232,ip112: Plantronics VOYAGER FOCUS UC BT Headset support
| Status | Closed |
| Id | 164726 |
V10 Service Release 31 (101358)
Changes included in Version 10 Service Release 31 Definition
SIP: Ports are not mapped when STUN server address has been configured
| Status | Closed |
| Id | 166436 |
Worked only when DNS name (instead of IP address) was configured.
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
CDRs: Forward information missing on CDRs generate for a call which was diverted to the user and then diverted to the next
| Status | Closed |
| Id | 166946 |
In this case the CDR at the user did not show that the call was already diverted to this user.
myPBX: Chat messages sent while the destination has not responded, got lost if call to different PBX
| Status | Closed |
| Id | 167061 |
Problem in the PBX to PBX signaling
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.
PBX: Max Call Duration setting did not work for call, with all legs incoming
| Status | Closed |
| Id | 167195 |
The assumption that there is always an outgoing call leg, was wrong.
PBX Waiting: Sometimes not all members of primary group were called, when blocked because of presence
| Status | Closed |
| Id | 167206 |
Calls need to be retried to operators which have been blocked because of presence once.
PBX: Reporting license counting wrong
| Status | Closed |
| Id | 167404 |
If a user confiuguration was changed while calls were active, it could happen that an additional reporting license was acquired, which was never released.
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.
IP-DECT: Channel trace added
| Status | Closed |
| Id | 168014 |
Channel trace added.
IP-DECT: DTMF through RTP fixed
| Status | Closed |
| Id | 168157 |
-
SIP: Fix for memory leak
| Status | Closed |
| Id | 168255 |
If SIP stack is flooded with messages not all memory was freed.
Fax server: Wrong CRP message with ECM on error data frames
| Status | Closed |
| Id | 168484 |
A wrong command repeat message (CRP) is sent if an error data frame is received after the end of the data frames (RCP) in error correction mode. The synchronisation between the devices is disturbed or lost. This is fixed now.
V10 Service Release 32 (101365)
Changes included in Version 10 Service Release 32 Definition
IPVA: _kernel::timestamp_delta(), Sample Frequency Altered
| Status | Closed |
| Id | 160286 |
Was 125us. Now cpu freq.
Media: Don't forward invalid G.711 channel with rate=16000
| Status | Closed |
| Id | 166773 |
Such a channel is sent by some SIP equipment
phone: ip222,ip232: no calling tone when a call completion indication was accepted via USB headset
| Status | Closed |
| Id | 169505 |
phone: ip222,ip232,ip112: USB headset support - Jabra SPEAK 810, Jabra BIZ 2400 II
| Status | Closed |
| Id | 169555 |
PBX: Compatibility with phones with v11r2sr9 or higher (boolean fkeys)
| Status | To-decide |
| Id | 170006 |
Boolean object of PBX v10 could not be controlled with boolean fkey on a phone with v11r2sr9 (or higher).
myPBX: fixed MSI parameter override with manual installations
| Status | Closed |
| Id | 170532 |
If MSI parameters have been given on a previous installation and the user installs a newer version of myPBX manually (without parameters), the old MSI parameter settings have been ignored.
PBX Waiting: For DTMF forwarded calls the destination was displayed even with Hide Connected Endpoint set
| Status | Closed |
| Id | 170829 |
A Transfer Complete was sent to the caller
myPBX Android: On Samsung smartphones parking a Wideband call crashed the app
| Status | Closed |
| Id | 170923 |
Since december 2015 the app crashed on Samsung devices if a connected call with wideband codec was parked. Somehow the audio API crashes with the floating point exception SIGFPE some 10 ms after turning off 16000 Hz record and playback. It helps to switch to 8000 Hz and then turn off. This is what happens now due to the tones that follow the park operation.
Office integration with Office 2013 didn't always work properly
| Status | Closed |
| Id | 171317 |
After closing and restarting an Office application, the presence didn't work anymore.
This Microsoft issue has been worked around by creating a new instance for each Office application itself instead of using a shared one.
Fax server: DCS not accepted
| Status | Closed |
| Id | 171453 |
The fax interface doesn't accept a DCS message if the extend field bits don't match with the message length. This is fixed now.
SIP: Trap in case of very long Via header
| Status | Closed |
| Id | 171533 |
Trap in case a request is received with a very long Via header.
Traps when response is constructed.
Fixed presence view issues in Office applications if the own contact is a favorite in myPBX
| Status | Closed |
| Id | 171579 |
Otherwise there have been issues for a busy presence when calling.
V10 Service Release 33
Changes included in Version 10 Service Release 33 Definition
phone: ip222,ip232,ip112: USB headset support - workaround for Plantronics dongle D100A (Savi W440)
| Status | To-decide |
| Id | 171671 |
The D100A dongle has the same Vendor/Product signature as the D100 dongle
but does not work with the HID-reports built in for the D100 dongle.
To overcome this problem an internal dummy signature for the D100A is added.
To make a D100A working it's D100 signature must be mapped to this dummy signature.
ip222, ip232: vars create KEYS0/HID-MAP p 047f:ab01=047f:d64a
ip112: config add KEYS0 /hid-map 047f:ab01=047f:d64a
The firmware uses the signature to assign the built in HID-reports to the device.
To discriminate between devices with identical signature the HID-report-descriptor
has to be parsed at plugin time which would require a lot of effort.
SIP: Calls are dropped after retrieve
| Status | To-decide |
| Id | 172001 |
Calls are dropped after retrieve if endpoints using ICE.
End-to-end re-negotiation missing after retrieve.