This is the Firmware 11r2 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!
11r2 Service Release 1 (113182)
Changes included in Version 11r2 Service Release 1
Definition
IP232/222/111: Partner fkey did not display icon as it did on old phones
Partner fkey did not display icon as it did on old phones.
On old phones a partner fkey displays:
- bell-icon while partner is ringing (pickup is possible)
- handset-icon while partner is connected or calling (pickup is not possible)
SIP: PBX sends re-INVITE after REFER was handled
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
Some digits could be duplicated.
myPBX Hotkey: Use Windows Automation and STRG-C instead of WM_COPY
IP232/222/111: Warning symbol was displayed if phone was connected to switch via PC port
Warning symbol ("LINK-DOWN") was displayed if phone was connected to switch via PC port.
Although the phone was registered and working.
SIP: Adjust offered framesize in media offers on media-relay interfaces
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
PBX: No CLIR for internal calls, did not work for Pickup
The dialog-info/group-indication did not show the number
myPBX Hotkey: Re-register if main window handle changes
Under some circumstances the window handle can change during runtime. If that happens the launcher has to re-register for some window messages.
myPBX: Remove fish from notification window
The fish in myPBX was replaced in 11r2 with the new myPBX logo. But in the notification windows there was still the fish. Not it is removed.
Fill missing bits with zeros in encode_base64
Additionally, an array boundary has been violated.
myPBX: New innovaphone logo
The claim was changed from "PURE IP-COMMUNICATIONS" to "PURE IP COMMUNICATIONS".
DHCP-Client:: a changed "IP Routing" option propagated via the server "Renew" button had no effect at the client
Video/Collaboration: ICE compatibility between v11 and v11r2
hmac_sha1 was calculated different in v11.
myPBX Hotkey: Use ClipboardFormatListener instead of ClipboardViewer
Use a different windows API that is more robust.
phone: DTMF tones received from remote were not played locally
Trap when restart is initiated
Trap in SIP stack when restart is initiated.
PBX SOAP: 6s timeout to pickup receiver if call is initiated for analog phone
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
A deleted memory region was re-accessed.
PBX: Connected number was not adjusted after SOAP pickup
The call looked as if connected to the original called number
IP232/222/111: Silent Monitoring did not work
Silent Monitoring could not be started.
IP-DECT: Wrong name with reverse phone book search
If there is a similar number in the LDAP directory, the number can be resolved in a wrong name. This is fixed now.
SIP: Problem with failover on failed call attempt
Problem with failover on failed call attempt.
PBX Wakeup Call: If Waiting was used for announcement, a restart happened if the call was rejected by the user
This trap could also happen with other object types
PBX Executive: WebRTC could not be configured for executive object
Should be the same as normal user
IP-DECT: Phone book search filter
The configured phone book search filter isn't considered in the search string. This is fixed now.
145308: IP232/222/111: Block dialing calls should be marked as 'sending complete'
Calls stared with enbloc dialing should be marked as 'sending complete'.
PBX: Forward received UUI on forwarded call, after CFB or CFNR
Needed for some special applications only
IP232/222/111: Bug in Executive/Secretary scenario
Bug in Executive/Secretary scenario.
Executive's phone show wrong presence information, when a secretary joins or leaves the group of secretaries.
Trap in conjunction with call completion
The new test cases in test/11.00/phone_android/phone-app-ip2x2 revealed this bug. To reproduce carry out these steps:
- Start an outgoing call.
- When ringing press the call completion button
- Press "Send Message"
- Send the message, click hangup.
- Go to the phone screen, change to the diversion settings and back.
myPBX: Some window icons were only available in low resolution
Use the program icon file that also contains the high resolutions.
SIP: Coder preference not always applied
Coder preference not always applied.
SIP: STUN not used if IP address was configured
STUN not used if IP address was configured.
STUN used if domain name was configured.
SIP: Close unused UDP sockets
Close UDP sockets used for NAT type discovery after NAT type discovery is done.
ASN1 tracing fixed
Encoded ints were displayed wrong
phone ip111,ip112: prevent duplicate stack dump after assert
SIP: Wrong error message in trace
Wrong error message in trace.
E.g.
sip_client::unbind_call(SIP-CLIENT.0) invalid call handle
Phones: Call initiated via call list was sent to wrong gatekeeper
Call initiated via call list was sent to wrong gatekeeper.
Save gatekeeper information in call list entries.
Phones: Fkeys stop displaying partners presence and call activity
Subscription are terminated and not re-established in rare cases.
E.g. Temporary call routing over ISDN line (during outage of IP link).
STUN: Binding response contained no IP address
STUN: Binding response contained no IP address.
But only if binding request came from an addr:port
that also has been configured as destination for an inbound forwarding.
Simple Traversal of UDP Through NAT
[Request In: 6816]
[Time: -359.000375000 seconds]
Message Type: Binding Response (0x0101)
Message Length: 0x0018
Message Transaction ID: 63383537316633376633356135353031
Attributes
Attribute: MAPPED-ADDRESS
Attribute Type: MAPPED-ADDRESS (0x0001)
Attribute Length: 20
Protocol Family: IPv6 (0x0002)
Port: 5060
IP: :: (::)
IP232/222/111: No name suggestion when adding new favourites with some directory configurations
No name suggestion when adding new favourites, but only with some directory configurations.
IP232/222/111: Allow REDIAL key to be used to initiate a headset call
Allow REDIAL key to be used to initiate a headset call.
But only on phone devices without dedicated HEADSET key on it.
On phone devices with dedicated HEADSET key, the REDIAL key opens the list of outbound calls.
Video: do not use rtp marker but the timestamp to detect end of access unit
I was using the rtp marker but this is not reliable if packetization-mode equal 0 is used.
Media: Webmedia channel in ECHO mode did not echo DTMF
Webmedia channel in ECHO mode did not echo DTMF
Licenses containing digits (e.g. G729channel) did not work
Problem parsing the license string
Fix for trap if invalid coder config is received by DHCP
Trap if invalid coder config is received by DHCP.
PBX SOAP: Potential unexpected restart when using the Devices function
In case the call to Devices used an invalid session, maybe because the session was just lost.
PBX-SOAP: If a call was initiated for a mobile endpoint, the call was indicated duplicate
The call was indicated with two different call handles
Call Lists on CF: Duplicate entries if call was sent to multiple registrations or mobility - fix for this fix
The last ix was not complete. A call which was accepted on one registration still showed up multiple times
SIP: Wrong coder in SDP answer after switch from "inactive" to "sendrecv"
Wrong coder in SDP answer after switch from "inactive" to "sendrecv".
Fax server: Raw data trace option added
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
For compatibility with some ISDN phones
PBX CDRs: CDRs from a Broadcast object was incomplete if the caller did a transfer
The rel-to/from and conn-from events were missing. No calculation of call duration could be done.
IP232/222/111: Partner fkeys did not follow language change
If phone's language is changed without restart, partner fkeys kept on displaying partner's presence in previous language.
PBX: Group dialog info subscriptions did not work for groups without members on the master
The result was, that dialog info from other slaves was missing and many failed calls from the slave to the master could be seen, which may also be sent to an extern interface on the master.
IP232/222/111: Sorting of favorites different from myPBX
Sorting of favorites different from myPBX, but only for names containing LATIN LETTER ETH or LATIN LETTER THORN.
PBX Pickup: With callidentifier to identify the call should work independent of the position
It should be possible to pick a call parked to a specific position by using the callidentifier allone, without park position. The park position is redundant in this case. This is how a park key does.
DTLS: Fix for negotiation of protocol version
The ClientHello should not only be accepted for DTLS 1.0. It should be accepted for all higher versions as well, but DTLS 1.0 should be negotiated.
PBX Trunk: "No Presence/Dialog Subscribe" did not work for local subscriptions from myPBX
These subscriptions were still sent out
PBX Wakeup: Was not executed if object had no registration but mobility
Check to avoid unnecessary executions did not cover this case.
SIP: SDP body was ignored if no Content-Length header line was present
SDP body was ignored if no Content-Length header line was present.
Content-Length header line is not mandatory for SIP/UDP.
No DNS server address with mobile data connectivity
If the smartphone had mobile data connectivity instead of Wifi, myPBX Android didn't know the DNS server addresses and couldn't resolve e.g. the STUN server if it was specified by host name.
SIP: Unsymetrical codec choice at call pickup
May apper on handling of INVITE with Replaces.
Phone: Trap if hotdesking registration fails
Trap if hotdesking registration fails.
Status:
Fixed in 11.00, 11r2, 12r1 (phone2)
IP232/222/111: Incoming call is dropped after 5 minutes when accepted from call-waiting state
Incoming call is dropped after 5 minutes when accepted from call-waiting state.
IP6000: Prevent blinking error LED on old IP6000 with HW-Build 201
Conference DSP driver was started on old hardware that doesnt support the conference DSP
IP232/222/111: Trap when CCNR/CCBS is activated on consultation call
Trap when CCNR/CCBS is activated on consultation call.
Status:
Fixed in 11.00, 11r2, 12r1
phone: numbers sent to the phone by myPBX to are dialled enbloc now
IP232/222/111: Fkey 'message' does not send prepared text message
Fkey 'message' does not work for prepared text message and prepared destination.
SIP: SIP interface should reject call with Q931_CAUSE_RequestedCircuit_ChannelNotAvailable
SIP interface should reject call with Q931_CAUSE_RequestedCircuit_ChannelNotAvailable (not Q931_CAUSE_AddressIncomplete_InvalidNumberFormat)
if remote proxy is currently not available ("down").
IP232/222/111: App "Favorites" can be disabled now
App "Favorites" can be disabled now.
Symbol does not appear on phone.
PHONE_HIDE_FAVORITES (0x00000010)
SIP: Domain part missing in Contact-URI of 302 response
Domain part missing in Contact-URI of 302 response.
PBX Waiting: Hide Connected Endpoint did not work if call was connected without announcement
This feature is for example used to hide the number of the waiting queue to external callers.
PBX CDRs: Use Uptime in events and not call relative time
Call relative time is difficult to calculate correctly if different calls contribute to a CDR because of Transfer, Pickup, ...
PBX: Registration by number failed if an object marked 'local' shadowed the destination object
When searching the destination object of an incoming registration by number, the local flag was evaluated. This was wrong.
SIP: Config option "No ICE" did not work in transit mode
Config option "No ICE" did not work in transit mode.
Config option "No ICE" only worked with media-relay and local-media.
SIP: Memory leak when rejecting request messages with "482 Loop Detected "
Memory leak when rejecting request messages with "482 Loop Detected".
SIP: Changed handling of History-Info header and stop sending Diversion header
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
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.
myPBX: Possible crash with the "Autostart softwarephone" feature
When the "Autostart softwarephone" feature was enabled the launcher could crash on exit or on restart.
phone: Audible Signalization of Announcement Calls did not work as expected with default settings
By default announcement calls have been signalled by a short inband tone on all types of phones. This tone was only hearable on ip222/232 but not on ip111(a),150,200a,230,240(a) and ip111.
Supressing this tone by checking "Phone/User-x/Announcement Calls/Audible Signal Off" did not work on ip222/232 but on the other phones.
Setting the checkmark "Phone/Preferences/Play Configured Ring Melody before Automatically Connecting an Announcement Call" fixed this Problem.
Announcement calls were then signalled by a configurable ring tone before connect but connected silently when ".../Audible Signal Off" was checked.
Now the phone always behaves as if "Phone/Preferences/Play..." has been checked, the checkmark itself is removed from WEB config page.
PBX Waiting: Outgoing call to trunk resulted in no audio
Worked to normal users
SIP: Missing response to re-INVITE(inactive)
No channels_app available to send 200/OK(inactive).
Oscillations at the beginning of speakerphone mode for IP111<->IP111
Two IP111 in speakerphone mode tend to oscillate at the beginning of the call. Tried to fix this by attenuating high frequencies a bit in the speaker equalizer.
PBX: Memory leak when serving SIP endpoints
PBX: Memory leak when serving SIP endpoints
SIP: Coder preference not always applied
Configured coder preference not always applied.
Trap due to double free
Trap due to double free of a packet.
IP111/222/232: Changing image on Camera app may fail with "Allocation limit exceeded"
Changing image on Camera app may fail with "Allocation limit exceeded".
Fixed memory management.
Reducing memory footprint of display rendering.
SIP: REGISTER rejected with "301 Moved Permanently"
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.
IP232/222/111: Presence control did not follow language change
If phone's language is changed without restart, presence control kept on displaying presence in previous language.
Annoying gaps in the peer signal during double talk in handset mode
Disabled the NLP in handset mode to avoid any gaps. The LEC should normally converge such tightly that there is no perceivable residual echo.
Lowered the NLP threshold to avoid as much of the gaps that it produces as possible for the handset and headset monitoring mode.
IP222/232/111: Trap when reboot is initiated
Trap when reboot is initiated.
Language setting not applied to extension module
Extension module was redered in German language.
Now language setting from phone device is applied to extension module.
11r2 Service Release 2 (113190)
Changes included in Version 11r2 Service Release 2
Definition
SIP: Must follow re-negotiation even while holding the call
Must process re-INVITE with new SDP offer even during 'inactive'.
Video: h264 stream wrongly decoded if poc type equal to 2
if poc (picture order count) was equal to type 2 the video stream was wrongly decoded.
Status:
frame gap at 255?
No restart needed on dialtone type change
A change of the dialtone type already applies without restart. To reproduce the odd behaviour
- Change the dialtone type of the primary reg
- Click "OK"
- When asked for restart click "No"
- Results in message "Change activated" and indeed it's changed
myPBX dial trace didn't work correctly
The trace file hasn't been written if tracing has been enabled by its MSI property.
CONF: Connected to a wrong conference room
With block dialing without any number the conference is assigned to a wrong existing room. This is fixed now.
IP-DECT: Release reasons for OEM PBX
Release reasons for an OEM PBX are changed.
PBX CDRs: clir flag sometimes mission
Was only in the first event of the call present
ISDN: Send Proigress Indicator "Originator is not ISDN" with audio calls
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
Call-Lists: Calls to users with multiple registrations, which were forwarded, were shown multiple times
Indicate in CDRs that there are more CDRs for the same call
Fixed myPBXDial crashes on terminal server
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
The additional dialed digits were added to an empty number and the resulting destination was wrong
Video: do not show video window if no webcam and no video received
If both sides have no webcam video windows are still shown although no one is sending video.
With certain debug settings the app could crash
E.g. the modified test phone_android/phone-presence-ip2x2 crashed the app due to its command
!config change PHONE CONF-UI /trace on
when it afterwards configured fkeys.
SIP: SDP offer with "vbd=yes" was rejected with 488
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
IP232/222/111: Own presence not updated on phone display after a while
Own presence not updated on phone display.
Self-subscription is terminated.
DNS resolution for automatic configuration of softwarephone
The softwarephone in version 11 doesn't support configuration of the gatekeeper using hostnames. Instead an IP address must be given. Therefore the launcher needs to do hostname resolution using DNS, if myPBX is configured using a hostname.
SIP: Wrong local RTP address in SDP in some special scenarios
Wrong local RTP address in SDP in some special scenarios.
Better use local IP address that is used for signaling (e.g. in Contact-URI).
myPBX URI should be case-independent
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.
Media Recording: If manual recording was configured a small file was generated even for not recorded calls
The file did not contain any RTP
SDP: Unable to process SDP messages bigger than 4096 bytes
Unable to process SDP messages bigger than 4096 bytes.
8192 bytes is the new limit.
IP232/222/111: Do not leave screen when touching presence info of a favourite
Do not leave screen when touching presence info of a favourite.
PBX: Potential Trap related to mobility and no response timeouts
Hard to find the real cause
PBX Broadcast: Potential trap if call of Broadcast did a transfer
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.
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.
SIP: Fix for media negotiation in early-media scenario
Fix for media negotiation in early-media scenario.
PBX Gateway Object: Outgoing Calls no Name/URL
To supress internal information to be sent to other systems
PBX: Dyn PBX could not turned off an on again
Response from DUMMYVOIP (WebRTC) was missing.
OEM Registration licenses did not work anymore
Collateral damage of fix: #146486: Licenses containing digits (e.g. G729channel) did not work
PBX Waiting: Trap when changing the config of a WQ with active calls to mobile operators
Duplicate delete
Dial pad not shown after call park
After a call has been parked the dial pad should be shown again in the phone screen because we may start a new call then by just typing a number. The same on incoming message. Until now the dial pad didn't show up even if the according button was pressed.
Application trap on start if logged in as a secondary user
If logged in to the smartphone as a secondary user the system throws an exception if we try to clear our own package preferred activities settings for the case of dialer claim "manual".
java.lang.SecurityException: Neither user 1010120 nor current process has android.permission.SET_PREFERRED_APPLICATIONS.
...
\tat android.app.ApplicationPackageManager.clearPackagePreferredActivities(ApplicationPackageManager.java:1458)
\tat com.innovaphone.phoneandroid.PhoneAndroidService.forms_set_forms_property(PhoneAndroidService.java:760)
PBX: Don't do RTP Proxy for WebRTC calls
RTP Proxy (or media relay) is not suppoprted by the WebRTC signaling
PBX Waiting: Original called number got lost on diverted calls to a Waiting Queue
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.
SIP: Add "Allow" and "Accept" and "Supported" headers to OPTIONS response
Add "Allow" and "Accept" and "Supported" headers to OPTIONS response.
Admin UI: Make SHA256 the default signing algorithm for certificates
Change default value in drop-down menus for creating certificates
* Signature: SHA256
IP241: New config file parameters /solid-header and /solid-status
New config file parameters /solid-header and /solid-status
IP232/222/111: More options for Fine grained function hiding
More options for Fine grained function hiding
Allow to hide APP_CONF, APP_LSIT and APP_DIR from display.
Allow to hide FKEYS from APP_HOME.
11r2 Service Release 3 (113236)
Changes included in Version 11r2 Service Release 3
Definition
PBX Session Border Object: Deleting one Session Border object clears registrations/calls on all Session Border objects
Happens only if the Session Border Object, which is deleted has active registrations
H.323: No Media in case of calls from trunks with media-relay/exclusive coder to a PBX with rtp-proxy to broadcast destinations
Enabling Media-Relay on the trunk and do rtp-proxy in the config is not a good idea for performance reasons allow, but should still work.
H.323: No Media after Pickup of a call to a trunk incomplete destination with RTP Proxy enabled
Media negotiation did not complete
PBX: Append additional dialed digits to call forward destination for GW type destinations only
This function could be abused by users
timestamps used in event logging could be wrong when setting of system time was delayed
-
timestamps set by logger could be wrong when the system time was set delayed after boot
-
PBX CDRs: Forwarded calls where missing in the call lists
Collateral damage from
148583: Call-Lists: Calls to users with multiple registrations, which were forwarded, were shown multiple times
unsent log data was not freed when the log server shadow was disabled
PBX Mobility: Unexpected restart on very unlikely call clearing collision
Missing null pointer check
myPBX Android sometimes incorrectly preferred ppp0 over wlan0
Changed the strategy when to prefer ppp0. Now we take wlan0 if the wlan0 local address matches the remote address better than the ppp0 local address, i.e. if the number of matching msb's is bigger for it.
SIP: No fast re-INVITE after reject for re-INVITE for t38
If switch to t38 has been rejected, there's no need to send
another re-INVITE for audio (except in case of ICE).
SIP: Fix for memory leak
Fix for memory leak when handling REGISTER with "gruu" and "+sip.instance".
IP222/232/111: Suppress "Audible Signal" of Pickup fkey while DND is ON
Suppress "Audible Signal" of Pickup fkey while DND is ON.
phone: cc-exec-possible indications for a pending call completion lost in some cases when sent to a busy phone
- 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 -->
H.323/TLS: Authentication with device certificate for analog interfaces of IP22, IP24, ... family
The certificate name is checked against the beginning of the registration name, so a certificate name of 009033xxxxxx is good for a registration of 009033xxxxxx-TEL1 as well.
Phones: Immediate cleanup resources when rejecting 'exec-possible' (call completion)
Immediate cleanup resources when rejecting 'exec-possible' (call completion).
IP232/222/111: Ghost call was displayed during transparent recording
Ghost call was displayed during transparent recording.
PBX Waiting: Input field for "Operator Presence Clear after ..." too small
It was not visible if too many digits were entered
SIP: Re-negotiation to fax did not work in some cases
Re-negotiation to fax did not work in some cases.
TCP/UDP: Logging did not show correct IP addresses
For IPv4 alway 0.0.0.0 was displayed
PBX Conference: Trap
There is a trap in the PBX conference call. This is fixed now.
SIP: Must reject any re-INVITE for t38 if "Enable T.38" is not set
Must reject any re-INVITE for t38 if "Enable T.38" is not set.
Return 488 Not Acceptable Here.
IP222/232/111: Mark directory entries with "mobile" symbol if Number Attribute is tagged with 'M'
Mark directory entries with "mobile" symbol if Number Attribute is tagged with 'M'.
E.g. telephoneNumber:D,homePhone:P,mobile:M
For more details see http://wiki.innovaphone.com/index.php?title=Reference10:Phone/User/Directories
myPBX: Possible trap with hidden recording calls
myPBX hides calls to the recording. In this context a trap could occur.
IP232: Backspace is executed before text input control has focus
When touching a text input control that hasn't got the focus yet at the very right end, the last character is deleted.
myPBX: New translations
Translations for the myPBX launcher and the myPBX web application have changed.
SIP: Wrong expires parameter in 200/OK for REGISTER
Wrong expires parameter in Contact header in 200/OK for REGISTER, but only if in case of multiple bindings.
SIP: ctComplete not always interworked into re-INVITE with updated P-Asserted-Identity
ctComplete not always interworked into re-INVITE with updated P-Asserted-Identity.
But only if ctComplete is passed through (in recording scenario).
Fax server: Wrong error correction
The error correction doesn't work if it is necessary. It results in missed document parts or failed connections. This is fixed now.
SIP: Changed trace message text
Changed misleading trace message text from "SIP message too large"
into "End of SIP message not found".
Voicemail: Duplicate Leak Checks
Occurred within regression tests
SIP: Re-negotiation from Audio to CLEARMODE did not work
Re-negotiation from Audio to CLEARMODE did not work.
re-INVITE was rejected with "SIP/2.0 488 Not Acceptable Here".
Refresh the NAT mapping also for packets from outside to inside
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.
SDP: Encoding was wrong due to uninitialized variables
Encoding was wrong due to uninitialized variables
IP222/232/111: Config parameter missing for PARTNER fkey
Config parameter "Aufschalten" missing for PARTNER fkey.
IP222/232/111: Phonenumbers from directory are not normalized
Phonenumbers from directory must be normalized using the dialing location before used.
E.g.
In Directory: +49 7031 73009 0
To be dialed: 00049 7031 73009 0
IPv6: De-fragmentation did not work
De-fragmentation did not work if more than 2 fragments were received.
SIP: Do not send SAVP answer to an AVP offer
Do not send SAVP answer to an AVP offer.
TLS: Overwrite sensitive data before deleting
To avoid leaving sensitive data in free memory space.
SIP: Try to handle offer/offer-collision
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.
SIP: CANCEL rejected when From-URI contains "epid" parameter
CANCEL rejected when From-URI contains "epid" parameter.
IP222/232/111: Phone-UI: Change page on 'key-press' or 'touch-on'
Change page on 'key-press' or 'touch-on' (Not on 'key-release' or 'touch-off').
And handle long-press.
PBX SOAP: TAPI could not assign users to correct PBX in setups with many PBXs
Group handling has changed due to dialog subscriptions accross PBXs
IP222/232/111: Display Alerting Partners on Pickup Key too
Display Alerting Partners on Pickup Key too.
SIP: Keep registration state on "UP" even if timeout on call signaling
Keep registration state on "UP" even if timeout (no-response) on call signaling.
Kicking registration is only required if alternative registrar address is available.
IP222/232/111: Trap in display rendering
Trap in display rendering.
Status:
http://inno-social.innovaphone.sifi/microblog/global/portal/topics/dvl/notes/41042
PBX Waiting: Set Operator presence did not work correctly
Presence was sometimes reset before the configured timeout
PBX: CFB on Trunk or Gateway did not work if the call was cleared with DISC
This happend for example on ISDN interfaces with in-band busy tones
PBX Boolean: Access rights (visibility) made configurable
Needed for the boolean function key
Linux: Deleted device DNS
If the Linux IP address is renewed, the actual device DNS is cleared. This is fixed now.
Network: Device's secondary DNS is cleared on dynamical route change
The actual secondary DNS of the device is cleared if a dynamical route is changed (a PPP connection or the Linux IP address). This is fixed now.
PBX Broadcast Conference: Call to WQ not closed
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
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
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
The operator was then regarded as busy
Session Border Registrations were lost, if a "License only" registration at the master was re-established
Happend for example when the license master was restarted
phone: ip222/232/241: accept packets from PC-link immediately after physical link-up
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
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 WebRTC: Unvisible hanging calls when terminating a WebRTC call by disallowing access to Audio/Video devices
Presence of the user indicated on-the-phone and idle reset did not work in such a case.
ISDN: Calls to NT Point to Multipoint terminated if a single endpoint responds with RELEASE_COMPLETE
A RELEASE_COMPLETE should be ignored as long as other endpoints could still accept the call
Favourite cannot be added if Fav App is not activated once
Favourite cannot be added if Fav App is not activated once.
Problems on login (bad encoding)
0:0041:104:1 - str::to_latin1(3) - caller 9443dd58 - bad encoding
SIP: Memory leak when receiving more than one 180 Ringing with name info
Memory leak when receiving more than one 180 Ringing with name info.
IP232/222/111: Change app when using keys LEFT or RIGHT
Change app when using keys LEFT or RIGHT.
Use SHA256 for automatically created certificates
Certificates that are created without any user interaction were created using SHA1. Now SHA256 is used.
TLS/DTLS: Support for Diffie-Hellman key agreement
Add the following cipher suites to DTLS:
* TLS_DHE_RSA_WITH_AES_128_CBC_SHA
* TLS_DHE_RSA_WITH_AES_256_CBC_SHA
TLS: Config options for disabling individual cipher groups
- TLS0 /no-rsa on
- disable RSA key exchange
;TLS0 /no-dhe on: disable DHE key exchange
;TLS0 /no-ecdhe on: disable ECDHE key exchange
;TLS0 /des on: enable DES cipher suites
Note that the cipher suite TLS_RSA_WITH_3DES_EDE_CBC_SHA is no longer used unless configured.
TLS/DTLS: Support for ECDHE key agreement
- Research how Diffie-Hellman works with elliptic courves and if we can do it with reasonable effort
* Port EC library
* Implement ECDHE handshake
Secure freeing of bufman buffers
New function bufman::free_secure that overwrites the memory before freeing.
myPBX: New translations
New translations for the myPBX launcher and the myPBX web application.
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.
11r2 Service Release 4 (113260)
Changes included in Version 11r2 Service Release 4
Definition
PBX: Hide Calls page
config option to hide the PBX calls page for privacy
IP222/232/111/112: Wrong melody played in ringtone configurator
Wrong melody played in ringtone configurator,
but only if ring-melody still configured as "Default".
PBX SOAP: LocationUrl broken, if standby slave takes over
The URL contained the expession (NULL).
SIP: Accepting call from myPBX doesn't work
Accepting call from myPBX doesn't work.
NOTIFY(talk) from PBX was rejected with "Bad Event".
myPBX: New translations
New translations for web application and launcher.
register for notification on changed variables only once
phone: ip222/232/111/112: CSV export of local directory left name column empty
phone: an inbound call arriving early after boot was 'automagically' rejected sometimes
Update of Presence Info on Extension Module does not work if Extension Module and Phone have the same list displayed
Update of Presence Info on Extension Module does not work if Extension Module and Phone have the same list displayed
PBX: Registrations not counted correctly, when registering with MAC address to old-style HW-ID
When user config was changed the existing such registrations were not matched correctly to the configured devices. This could cause all kind of problems. It was detected when TAPI lines disappeared.
Logging of PBX SOAP Admin requests resulted in broken log messages
The text contained NUL characters und no XML data as it should
PBX Number Map; Call was forwarded with diverting leg2 info
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.
PBX SOAP: Struct item tag name changed from v10 to v11
It should be possible that this tag name is chosen freely, but there are applications which depend on a specific name.
Reduced the sidetone gain on IP222
The sidetone was perceived as too strong on IP222. Reduced it by 6 dB through different balancing of analog and digital mic gain.
SIP: NOTIFY on a subscription was rejected with "481 Call Leg/Transaction Does Not Exist"
NOTIFY on a subscription was rejected with "481 Call Leg/Transaction Does Not Exist"
Collateral damage from
#152586: SIP: Accepting call from myPBX doesn't work
(v12r1 / v11r2sr4 / v11r1sr6)
IP232: Phone-UI: Paging left-ward did not work since v11r2sr3
IP232: Phone-UI: Paging left-ward did not work since v11r2sr3.
IPVA: Number Of Available Vars Segments 11
was 1. Now 11: twice as much as for an IP6010.
IP222/232/111: Subscription started by call list app used display name as destination user-id
Subscription started by call list app used display name as destination user-id.
H.323: Potential restart on incoming H.245 TLS connection
This is something which happens with very specials configurations only.
Gateway: A route with the matching number terminated with '!' should cut off any following digits
This worked fine for enblock calls, but not for overlap dialing.
PBX Waiting: Disconnect was signaled with SOAP when announcement changed
This caused applications to show a wrong state
IP-DECT: H.323 user registration with RAS ber TCP/TLS (H.460.17) added
Now it is possible to register the users with H.323 with RAS ber TCP/TLS (H.460.17).
Gateway: Some protocol settings got lost, when changing interface maps
The settings, which got lost are:
* H.323/TCP or H.323/TLS - was changed back to H.323
* SIP No registration
* SIP transport tcp or tls
Web-UI: Font-family of input, select, textarea, button did not inherit body style
Font-family of input, select, textarea, button did not inherit body style.
Using now "font-family:inherit" to have same font-familiy all over.
IP-DECT: Reset required notification for OEM version fixed
-
IP-DECT: SRTP default value for OEM version changed
-
when the system time was derived from an ISDN trunk the current timezone offset was not taken into account
Happens when no NTP server is available and "Set Date/Time" is checked under "Gateway/Interfaces/Interface/TELx"
IP6: setting the "Default Gateway" of an IP6 interface with "Address Configuration:Static" did not work
SIP: Trouble handling SDP offer with "vbd=yes"
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
myPBX: Allow non-breaking-spaces in phone numbers
Phone numbers from Outlook can contain non-breaking-spaces. For example this happens with contacts that are synchronized from an iPhone.
IP222/232/111: Attribute "No Pickup" could not be configured for Partner fkey
Attribute "No Pickup" cannot be configured on Partner fkey when configuring on the Phone.
Was only available on Web config before.
PBX: New visibility flag Pickup and ON_THE_PHONE independent of presence
This solves two issues:
- customers would like to allow pickup, without providing full dialog info with all numbers
- customers would like to give visibility to on-the-phone, without revealing presence
IP222/232/111: Some call list entries could not be called back
Some call list entries could not be called back since v11r2sr3.
Collateral damage from fix #151637.
myPBX: New translations
-
Show calls with CFNR to another user as missed call in the myPBX call list
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.
CF Call Lists: Counting of missed calls wrong, in case of multiple registrations or mobility
Call was counted as missed even if accepted on other device
SIP: Wrong local media address selected in some cases
Wrong local media address selected in some cases.
When sending INVITE to inbound registrations (e.g. PBX clients).
SIP: Trap on calls with very long phone number
Trap on calls with very long phone number.
Status:
Fixed in 10.00, 11.00, 11r2, 12r1
PBX Waiting: SOAP UserRedirect of operator call did not work as expected
The operator was re-called after 3s
Video/Collab: libraries were not started if socket->bind call failed for a single port.
we just allowed WSAEADDRINUSE to happen as error for bind calls but it does not matter if an error ocurrs, maybe following port does not return an error.
IP222/232/111: Pending inbound call-completion requests were not displayed in call lists
If someone calls you and gives up before answering, a missed call is placed into call-list.
If caller activates call-completion, this 'missed call' entry is now replaced by a 'call-completion' entry instead of getting deleted from call-list.
IP222/232/111: Some fkey config parameters got lost when re-configuring fkey on the phone
Some fkey config parameters got lost when re-configuring fkey on the phone.
E.g. "Send as Control Call" on 'Dial' fkey.
Multiple invokes of user config screen possible
Showing only the first Screen with viable Information
Config dir and files were not created for a different user than the installing user
11r2 Service Release 5 (113289)
Changes included in Version 11r2 Service Release 5
Definition
Use CN as file name for certificate downloads
Previously the file name for all certificate downloads was "certificate.crt". Now the CN (or the next available name component) is used, like "IP800-06-11-ac.crt".
PBX Waiting: Remote number wrong after round robin recall, if transfer had happend on incoming call
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
IP-DECT: Registration facility for OEM PBX changed
The facility for user registrations to an OEM PBX is changed.
HTTP request to <domain>/drive/... accessed the local CF
HTTP request to a URL <domain>/drive/... e.g. from a voicemail script accessed the local CF instead of the remote resource.
PBX CDRs: No info-from, info-to events after conn
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
Alarm on CF removal without previous unmount
If the CF card is physically removed or by a false internal card detection (e.g. due to a card issue), an alarm is now raised.
Gatekeeper: Protocol name SIP changed to SIP/UDP
The protocol name in the drop down menu for registrations is changed from SIP to SIP/UDP.
IPVA: Trap when activating IDE tracing
Trying to activate IDE tracing caused a crash.
Gateway: On calls between Exclusive Coder interfaces and non-Exclusive Coder interfaces set both sides to exxclusive coder
Also the coder is set to the coder on the original exclusive coder interface
Gateway: Logging for Gateway routing showed maps, which were not executed
Output was wrong/confusing
PBX Waiting: Pickup an alerting call from queue did not work anymore
Call was disconnected
PBX: Default visibility settings including Calls and Calls with number
Additionaly use terms visibility and calls instead of access and dialog
Issue event and prevention of stackoverflow under high CF load
An event is now issued if too many files/directories are open.
A stackoverflow under such a condition has been fixed.
SIP: Bug on media negotiation
Bug on media negotiation.
Second provisional response contains an SDP offer instead of previously sent SDP answer.
Discovered in automated fax test (media/fax).
allow dsp trace to be switched on/off during operation
set softwarephone path without quotes
myPBX: Docking did not work correctly in Windows 10 with scaled desktop
If the scaling of text, apps and other elements was set to 125% or more, myPBX did not dock to the right position on the screen.
PBX Conference: Room number length specific announcement
The PBX conference object searches for the room number input announcement a room number length specific file first now.
Trap while reading kerberos config after upgrade from v9
Boxes with version 10 or higher could trap while starting after upgrade from version 9, if kerberos was configured.
H.323: Checking of old IP240-1000 device certificates did not work
There are 'old' device certificates with a name of IP240-1000-<mac4>-<mac5>-<mac6>. These have to be matched to registrations using the mac address as hardware id. The algorythm doing this, did not take the -1000 into account.
PBX: Voicemal: Wrong connected number sent, in case VM was 'local' object
The caller got a display of the VMs node number, which is not desired for 'local' objects.
Faxserver: Receiving G.711 fax aborted sometimes with ECM
Error decoding T.38 data from DSP
Softwarephone doesn't react anymore when 2 headsets are plugged in
swphone Loops endlessly during Startup when more than one HID telephony device is present
SIP: No audio due to bug in media negotiation (Interop with Openstage phones)
No audio due to bug in media negotiation.
Occurs when called SIP client returns an SDP answer with more than one selected audio codec:
\tv=0
\to=OpenStage-Line_0 513161446 1351641410 IN IP4 10.253.7.13
\ts=SIP Call
\tc=IN IP4 10.253.7.13
\tt=0 0
\tm=audio 5010 RTP/AVP 18 8 0 101
\ta=rtpmap:18 G729/8000
\ta=rtpmap:8 PCMA/8000
\ta=rtpmap:0 PCMU/8000
\ta=rtpmap:101 telephone-event/8000
\ta=silenceSupp:off - - - -
\ta=fmtp:18 annexb=no
\ta=fmtp:101 0-15
\ta=sendrecv
Fix for trap in SIP stack when using TCP or TLS as transport
Fix for trap in SIP stack when using TCP or TLS as transport.
When registering via mypbx and using passwords with special characters, registering fails
SIP: Trap when parsing presence XML with many presence/tuple elements
Trap when parsing presence XML with more than 5 presence/tuple elements.
ip28: incorrect measurement of pulse dial pulse length
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
The number was displayed containing node prefixes
802.1X: Display "Proxy-Logoff"-Checkmark Only Where Applicable
Display the checkmark only on phones with a "PC"-port
SIP: Local IP address in Contact-URI not updated when using SIP/TCP or SIP/TLS
Local IP address in Contact-URI not updated when using SIP/TCP or SIP/TLS.
For the very rare case that the local IP address changes at runtime.
IP222/232/111: Disconnected consultation call not dropped when retrieving held party
Disconnected consultation call not dropped when retrieving held party.
PBX SOAP: Call Intrusion calls could not be initiated with SOAP
The rc in UserCall argument needs to be sent to the local phone as well.
myPBX: New translations
New polish translations for v12 and v11
When installation package is deleted a new user cannot run softwarephone
IP222/232/111: Blind transfer with <redial>12345<redial> was not possible
Blind transfer with <redial>12345<redial> was not possible, as it was possible on old telephones.
Call-Info header is used to request automatic call answer.
Now writing IP address into SIP-URI instead of "domain".
Call-Info: <sip:1.2.3.4>;answer-after=0;rc=0
instead of
Call-Info: <sip:domain>;answer-after=0;rc=0
IP222/232/111: Favorites not sorted different compared to myPBX
Greek letters are sorted behind latin letters on myPBX, but before latin letters on phones.
Now greek letters are sorted behind latin letters on phones too.
HTTP: IP V6 support for allowed station
-
H.323: Send ICE candidates in an answer only if there were ICE candidates in the offer
Most endpoints ignore these candidates, but it does not conform to the standard to send them and some endpoints treat it as protocol error
Jabra 410: When going offhook via mypbx the onhook button on the device didn't work
Mobility: CFNR after timeout back to another user who has diverted the call to the Mobility User did not work
A loop detection was prohibited this unnecessarily
CF Call Lists: Read next mechanism did not work
When scolling down the call list in myPBX an pressing the button to read more entries, the same entries were added again to the end of the list.
PBX/Quick Dial: Consider General Checkmark "Hide Connected Endpoint"
Alpha display information was erronously generated, regardless of the setting of the checkmark named "Hide Connected Endpoint.
Phones: Presence not updated on phone display
Sometimes the self subscription for Presence is terminated by the phone.
IP222/232/111/112: Bug in key handling
Bug in key handling.
Every key event after R key press was discarded.
E.g. Going onhook quick after pressing R key does not disconnect.
IP-DECT: Local ring-back tone after call transfer and early media
If a call in alerting state and with early media is transferred and the new call hasn't early media, no ring-back tone is heard. This is fixed now.
IP-DECT: MWI update with handset change (login feature)
If the handset is changed with the login feature, the MWI isn't updated correctly. This is fixed now.
SIP: Do not interwork an early channels answer into 183 Session Progress with SDP if no PROGRESS indicator
Ein Fall von Interworking zwischen SIP-Carrier und einer ASCOM-IP-DECT-Base-Station.
ASCOM sendet ALERT mit Early-Answer (aber ohne PI).
Wenn wir da ein 183 Session Progress mit SDP-Answer zum SIP-Carrier geben, denkt dieser, wir spielen Early-Media ein.
Tut die ASCOM-IP-DECT-Base-Station aber gar nicht.
Der entfernte Anrufer hrt dann Stille bis zum Connect.
configured DNS adresses sometimes lost after reconfiguration of Linux-AP
SIP: Multiple SIP proxies for registration can cause high load
Proxies are tried round robin without delay.
No connection to webrtc
phone: ip241 - manual configuration of IP address, netmask and default-gw directly at phone did not work
Subscriptions of favourites have not been destroyed after leaving fav app
Subscriptions of favourites have not been destroyed after leaving fav app
Do not refresh myPBX if communication with an office application crashed
Sometimes the COM connection between myPBX and an Office application fails.
In this case, myPBX is now not completely refreshed anymore, just the COM part.
You'll still have to restart your Office application (hinted in the myPBX trace).
SIP: ICE on Overlap-Dialing and Early-Media
Only one side restarts ICE after re-routing.
Automated test project media/pbx fails.
SIP: Media negotiation fails in some call scenarios
Media negotiation fails in some call scenarios with Waiting Queue involved.
New visibility setting for dialog info without IDs
Allow seeing dialog info without IDs. This is needed for pickup without exposing the numbers of calls.
Additionally we did small improvements for the user interface in myPBX.
myPBX: Redirect to another PBX using HTTPS
myPBX always redirected to an HTTP URI. Now the redirect keeps the current protocol.
make filetrace hotconfigurable
Enabling/Disabling file trace does not require a restart anymore
rpcap IP trace implemented
SIP/SDP: Cisco does not support SRTP lifetime
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/security/8_6_1/secsipvideo861.pdf
"The lifetime parameter in the SDP must not be populated."
New config file option for Cisco interoperability:
config change SIP /no-master-key-lifetime
11r2 Service Release 6 (113306)
Changes included in Version 11r2 Service Release 6
Definition
H.323: Audio not switched to correct coder, when connecting a DSP to an active VOIP channel
Happened on IP-DECT when putting a call on hold, the putting the same call from the other side on hold and then retrieving it again from the original side, if different coders were used for the call end to end and for the MOH.
PBX Waiting: Operator call transfered by SOAP was not shown as transfer call
This function is needed in the innovaphone Operator application
PBX: For subscription calls to other locations, cpu for media relay was reserved, if media relay was enabled
The subscription calls never use media, so no reservation should be made
IP112: USB headset speaker volume lower than on IP222
The volume control for the USB headset was calibrated to a gain of 0 on maximum setting but should instead match the IP222 which has gain 0 on a mid level setting and allows adding actual gain.
CF Call Lists: Leak when configuring an invalid WebDav destination
The files in the local WebDav client were not closed, when writing failed
H.323: No DTLS after hold/retrieve on media-relay
This caused a WebRTC call to be disconnected after hold/retrieve
SIP: ICE information of remote channels got lost
ICE information of remote channels got lost.
Also honor CHANNEL_FLAG_OFF.
H.323: Calls from SIP media-relay/exclusive coder to Conference via H.323 did not work
In this case the no media answer was generated with the connect, which caused SIP to timeout
PBX Waiting:CDR did not show the transfer to the operator anymore
This was a collateral damage from fix #138649: PBX Waiting: CDRs in mode 'Operator connect for SOAP' should reflect the time a caller is waiting in queue
phone: function keys "Create Registration" and "Switch" could not be configured for H323/TCP and H323/TLS
IP-DECT: SRTP configuration for OEM version
After changing the default value for SRTP the configuration couldn't be saved in the OEM device version. This is fixed now.
Phone: STUN server configured at function key "Create Registration" was not used
STUN server configured at function key "Create Registration" was not used
Regards SIP registrations only.
SIP: Bug when handling REGISTER from same addr/port for different users with same Contact-URI
Bug when handling REGISTER from same addr/port for different users with same Contact-URI.
Seconds REGISTER just got 200/OK without any processing.
IP111: Fix for a trap
Fix for a trap
IP-DECT: Release code for unconnected calls to radio
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
Intermediate crash when early media was recieved
SIP: Support for RFC-4904 (Trunk Groups in tel/sip URIs)
Extracting CDPN from Request-URI's like this:
sip:+390249499001;trunk-context=3902494990.ims.vf.it;tgrp=tg3902494990@3902494990.corporate.vodafone.it
Phone: STUN server config missing at function key "Create Registration"
STUN server config missing at function key "Create Registration" for H323 regs.
IP-DECT: Release string added for IP1202
The release string is missed on the IP1202. This is fixed now.
SIP: Wrong local RTP address selected on interfaces without registration
Wrong local RTP address selected on interfaces without registration.
But only if "Proxy" option was not configured.
Phones: Fix for a trap
myPBX: Visibility settings for presence note had no effect in search result
The presence note was displayed in the search result, event if it was configured to be invisible.
IP-DECT: Release code forwarded to DECT
The release code isn't forwarded to the DECT system in some cases. This fixed now.
SIP: Digest authentication sometimes fail due to wrong calculation
Digest authentication sometimes fail.
Wrong method is used to calculate digest response on client side.
H323/tls does not register anonymously
SIP: Fix for SIP clients not supporting RTP/SAVP (media encryption)
Fix for SIP clients not supporting RTP/SAVP (media encryption).
If INVITE with RTP/SAVP is rejected with "406 Not Acceptable",
the INVITE is re-tried without media encryption (RTP/AVP).
This workaround already worked for 408 or 415 responses.
Now it works for 406 also.
Admin UI: Truncated Kerberos host name after config changes in CMD0
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: Bad SDP answer for re-INVITE after rejected t38
1. Call setup with offer/answer exchange for coder "A".
2. re-INVITE for "T38" is rejected with 488.
3. re-INVITE for coder "A" is accepted but bad SDP answer is sent.
TLS: Verifying of RSA signatures didn't always work
If the signature of a certificate started with a null byte the verification could fail in some special cases.
Possible trap in DTLS
Trap with DTLS-SRTP with Firefox 42
SIP: Support for CTI using REFER without preceding INVITE
Support for Third Party Call Control (3pcc) using REFER outside a dialog.
SIP: Interop with Jitsi client
Adding "Jitsi-Conference-Room: xxx" to INVITE.
11r2 Service Release 7 (113345)
Changes included in Version 11r2 Service Release 7
Definition
myPBX: Use newest registration instead of oldest for call control
When a phone looses the registration without closing the TCP connection gracefully (trap, network change), after the re-registration the PBX sees both the old dead and the new registration for a short time.
myPBX used always the first registration for call control. In this case this would be the old one that doesn't work anymore. So it's better to always use the newest registration.
RemoteMedia: Always use primary local address of phone for connection from launcher to phone
Using the local address of the registration caused problems with some VPN settings. So now we use always the primary address. That means that the phone and the computer must be in the same network or the networks must be routed.
SIP: UPDATE with SDP during early-media was rejected
UPDATE request with SDP was received before call was connected.
UPDATE request with SDP was rejected with "403 Forbidden".
IP112 USB headset microphone not sensitive enough
From the field we got notification that sometimes the peer listeners complained about too low volume. Therefore added 7.5 dB gain which includes a level limiting feature. On the IP222 there seems to be 8 dB gain in this path. Let's try if it's OK now.
IP232/222/111: Make Phone-UI return to last user-activated app after blind-transfer has been initiated
Make phone UI return to last user-activated app after blind-transfer has been initiated.
E.g. HOME app is active
- call comes in (phone jumps to PHONE app)
- user accepts the call, talks and presses REDIAL key (phone jumps to DIR app)
- user enters transfer destination and presses REDIAL key (call is transferred)
Now phone automatically returns to HOME app.
SIP: Interface goes down when STUN server changes it's IP address
Updated DNS information is fetched, but new IP address is not used.
PBX Trunk: Option to discard Diverting info received with incoming calls
Diverting Info from a provider is sometimes not desired
myPBX MSI parameter VIDEOACTIVE didn't work
If the paramter has been given as false, the registry value hasn't been written.
IPXX10: Flash Directory Space Increased To 16MB
Was 8MB
IP-DECT: Avoid busy treatment during call transfer
If feature codes are enabled on the DECT Master, call waiting is disabled and there are pending calls for call transfers (with SIP), further calls are rejected as busy instead of forward them to the idle handset. This is fixed now.
Implement ECDSA algorithm
- Update micro-ecc library in order to support arbitrary hash sizes
* Implement ECDSA using the library
Gateway: No busy tone was played with MOH Mode to MOH Source on disconnect
So a Music on Hold source connected to an FXS interface could not detect that the call was disconnected and did not release the line, so it was busy for further calls
PBX: Partnerkeys with Group Indications, did not show outgoing number in case of block dialing
With overlap dialing it was ok.
IP232/222/111: Bug when starting a call while phone is idle but handset is lifted
Bug when starting a call while phone is idle but Handset is lifted.
If headset was connected, Headset was activated.
If no headset was connected, Speaker was activated.
Better activate Handset when handset is lifted.
SIP: Don't escape pound sign in SIP-URI's when "user=phone" is added
No need to escape pound sign (#) in userpart of SIP-URI's when "user=phone" is added as URI parameter.
E.g.
INVITE sip:103#@IP800-PBX;user=phone SIP/2.0
Instead of
INVITE sip:103%23@IP800-PBX;user=phone SIP/2.0
IP222/232/111: Unable to add favorites for users in other PBX nodes
Unable to add favorites for users in other PBX nodes.
Node prefix was missing.
Sometimes video and app sharing icons do not appear in mypbx
PBX: It could happen that subscription calls were sent from a license only slave to the master
A license only slave should not exchange any information with the master except licenses.
PBX Wakeup: Call Retry too fast, Wakeup not executed anymore if call failed because of Busy On 1 call
The retry timeout is 10s now. Wakeup does not stop anymore.
PBX: Wrong connected number for subscriptions to another node on same PBX
The node prefixes were missing
IP-DECT: Support multiple call transfers/reroutes
Now it is possible to transfer or reroute a call by the call transfer or reroute destination.
SIP: Fixed Mapping of ISDN/QSIG Cause Values to SIP
Mapping of ISDN/QSIG Cause Values to SIP did not follow RFC-4497.
H.323/ISDN: Slowstart Call from H.323 to ISDN could fail under special conditions
PROGRESS message received after SETUP_ACK
Gateway: Cut off trailing # (a.k.a. hash or poundsign) from CDPN on routes with "Force enblock" option
Cut off trailing # (a.k.a. hash or poundsign) from CDPN on routes with "Force enblock" option.
Already done on calls with overlap dialing.
Also done on calls with sending-complete indication.
PBX Mobility: Potential trap on unexpected disconnect
If this collides with some user actions
WebDAV: Reducing memory usage of WebDAV client
Reducing memory usage of WebDAV client allocated for XML parsing.
web folder cleanup
Removed some folders (jsxsl, jscharts, styles, style, font).
Also removed the ui js files inside 12r1/web/js.
SIP: Don't return SDES key in SDP answer if option "Keying" is set to "No Encryption"
Don't return SDES key in SDP answer if option "Keying" is set to "No Encryption".
SIP: Bad SDP offer sent in re-INVITE when Media-Relay is activated and T.38 deactivated
Bad SDP offer sent in re-INVITE when Media-Relay is activated and T.38 deactivated.
E.g.
\tv=0
\to=- 6 2 IN IP4 172.16.161.16
\ts=session
\tt=0 0
\tm=image 0 udptl t38
\tc=IN IP4 172.16.161.16
myPBX Android: For H.323/TLS one way audio with peers that do not support ICE
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.
SIP: Picked wrong destination URI as call destination
An INVITE received on a registered device may not contain the called AOR.
To-URI contains the originally called AOR.
Request-URI contains the Contact-URI.
SIP: Add Session-Expires to 200/OK when /session-expires <seconds> is configured
Add Session-Expires to 200/OK even if received INVITE did not contain Session-Expires.
ip38: possible trap if received FSK CallerID information corrupt
if the lenghth field of a FSK CallerID has values above 128, an internal counter may overflow and cause an endless loop.
SIP: Bug in media negotiation on DECT gateways
Bug in media negotiation on DECT gateways.
Phones: Entered number not displayed during overlap dialing
Entered number not displayed during overlap dialing.
But only after PROGRESS has been received.
E.g. from TONE interface.
Bug exists since 11r2sr6.
SIP: Mapped IP address in Contact-URI not refreshed
After NAT router has changed its public IP address,
the new public IP address must be sent in Contact-URI of next REGISTER request.
SNMP: Encoded Trap Agent Address Always 0.0.0.0
IP-DECT: Reroute result with alert response
Now the reroute result is sent if an alert is received from the reroute destination call instead of the connect.
IP232/222/111/112: Fix for a trap
Fix for a trap.
Collateral damage of fix #151637: IP222/232/111: Phonenumbers from directory are not normalized
Since v11r1sr6 / v11r2sr3
Fixed tooltip for STUN server config
Fixed tooltip on page IP4/General/STUN
Voicemail: <store-del> Didn't Escape '*'-Character in File Name
Filenames containing '*'-character must be escaped.
myPBX for Android: Hook button of cable headsets not taking effect
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.
IP222/232/111/112: Memory leaks on LDAP search
Memory leaks on LDAP search.
Audio packets delay every 3. Packet
Sometimes, the softwarephone sent inconsistent RTP data
CF Call Lists: There should be no entry if no number/name was dialed
Just going off-hook created an entry in the call list
DTLS: ClientHelloExtensions were added twice after HelloVerify
After receiving a HelloVerify the ClientHello contained two times the same extensions Elliptic Curves and EC Point Formats.
PBX SOAP: Use latest registration for call if there are devices with multiple registrations
Multiple registrations could happen, because a device restarts and creates a new registration, while the old is not removed yet. In this case using the latest one is better.
IP222/232/111/112: Cannot cancel hotdesking registration attempt
If hotdesking attempt fails (e.g. wrong password) the phone
stays in "Registering" state for a very long time (45 seconds)
until finally all comes to an end with "Operation failed".
If user wants to cancel this process with ESC key,
the popup disapears but the registration attempt goes on.
One-way audio on phone_inca with DTLS-SRTP
The key derivation needs to be computed on a low priority to avoid problems with the DSP.
phone: ip222,ip232: USB headset mute when a call was released by remote peer and a new call was signalled imediately thereafter
H.323: Unexpected restart, when forwarding a call without ICE but with DTLS to an interface with no encryption
This results in empty ICE data, which was not handled well.
FAX: Judged training failure in some cases where TCF was well acceptable
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.
SIP: Request-URI and History-Info header could contain wrong information when re-trying INVITE without encryption
But only if 180/Ringing was received before with Contact-URI.
SIP/TCP: Keep client-initiated connection open
Keep client-initiated TCP connection permanentely open
to allow server to send SIP requests through it to the client.
Not not only if behind NAT.
Fix for verification of ECDSA signatures
Sigatures that need addition of leading zeros were not verified correctly.
myPBX: Remove restriction for length of details from LDAP directory
The length of detail values was restricted to 127 characters. This restriction is now removed.
Fax server: Mode bit check removed
The mode bit check is removed because of non-compliant remote devices.
Fax server: Maximum frame timeout increased
The maximum frame timeout is increased for compatibility issues.
IP232/222/111: No need to press "OK" after entering a PIN
No need to press "OK" after entering a PIN.
PIN is now checked while typing.
If PIN is correct things go on automatically.
DTLS: Support for ECDSA cipher suites
Support for client-side
* TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA
* TLS_ECDHE_ECDSA_WITH_AES_256_CBC_SHA
Phones: Config option "Protect Configuration at Phone" now locks ringer volume
User can still change ringer volume during ringing (keys left/right)
if config option "Protect Configuration at Phone" is activated,
but changed value is not written into persistent user config.
Config option "Allow User Settings at Phone" allows persistent volume setting.
11r2 Service Release 8 (113355)
Changes included in Version 11r2 Service Release 8
Definition
SIP: Presence interoperability with ESTOS UC server
SIP: Presence interoperability with ESTOS UC server
DHCP: on a change from disabled to client mode without reboot the received lease parameters were not propagated to IP stack
PBX: Validate XML phone config before writing to config
Validate XML phone config before writing to config, since bad XML causes bad trouble.
IP222/232/111/112: Dial fkey with "prepare" and "announce" did not work
Call was started with "announce" option.
SDP: Master Key Lifetime could be missing in a=crypto line
Master Key Lifetime could be missing in a=crypto line.
Trap when option "Outgoing Calls No Name" is set on PBX object
Trap when option "Outgoing Calls No Name" is set on PBX object.
Available on objects of type "Gateway".
IP232/222/111/112: Change app using keys LEFT or RIGHT does not work when in empty LIST app
Change app using keys LEFT or RIGHT does not work.
But only if in LIST app without any entries (empty).
phone: a cc-exec-possible sent to a busy phone got lost when the active call was relased by myPBX
-
SIP: Fix for bug in media negotiation on outbound calls without offer
Fix for bug in media negotiation when INVITE without SDP offer is sent.
Phones: Highlight "Partner" fkey if partner's presence is 'busy', 'on the phone' or 'do not disturb'
Highlight "Partner" fkey if partner's presence is either 'busy', 'on the phone' or 'do not disturb'.
phone: ip222,ip232,ip112: Plantronics VOYAGER FOCUS UC BT Headset support
myPBX: Harmonize LDAP number resolution with phones
The search filter on the phones contains also the international number without wildcards. This part of the filter is now also used in myPBX.
TCP/IP: Increase default keep-alive interval on TCP connections
To keep network load and CPU load on a low level on large PBX scenarios:
- Increased default keep-alive interval on TCP connections from 20 seconds to 120 seconds; 6 re-transmission are done with a distance of 20 seconds
- Avoid both sides of a connection sending keep-alive packets by running a slightly bigger default keep-alive interval on server side (121 secs) than on client side (120 secs).
"LDAP/Server/Allowed Networks" Didn't Work
The matching method was outdated
"SNMP/Allowed Networks" Didn't Work
The matching method was outdated
softwarephone: added phonesig interface for setting RAS license state
Phones: Show same symbol for presence activity 'appointment' as used for 'meeting'
Show same symbol for presence activity 'appointment' as used for 'meeting'.
Instead of symbol for 'unknown'.
11r2 Service Release 9
Changes included in Version 11r2 Service Release 9
Definition
phone: ip111,ip112,softwarephone: any registration must query the PBX for g729 licenses
SIP: MESSAGE request from pjsip client was rejected with 407
MESSAGE request was rejected with 407 if Contact header was missing.
Phones: When using fkey "Boolean Object" the user's presence was set to "on-the-phone" for a moment
When using fkey "Boolean Object" the user's presence was set to "on-the-phone" for a moment.
All monitors get spurious presence updates.
SDP: Write c-line before t-line in session description
Write c-line before t-line in session description.
Whitespace in CN of application certificates didn't work
Missing URL-Decode
IP232/222/111/112: Show more entries of local phonebook
Show up to 100 entries (instead of only 40) of local phonebook.
UDP-NAT: (re)configuration of UDP-NAT port range must not be applied without reset
IP222/232/111/112: Problems with fkey type 'Boolean'
Fkey did not subscribe at PBX for monitoring the boolean object.
IP222/232/111/112: Problems with fkey type 'Message'
Did not work if message text was pre-defined but destination not.
Should open the composer with the pre-defined message text in order to enter the missing destination.
IPv6: PING from a box with a 6t04 interface to another box over this 6t04 interface did not work
IP222/232/111/112: Problems switching character input mode
Problems switching character input mode from small letters to capital letters.
But only on first character.
H.323: v12 Registration with H.323/TCP or H.323/TLS did not work at v11 PBX
In v12 a H.460-17 non-conformity was fixed and in v11 we were not compatible to this.
SIP: 200/OK was sent without SDP offer
200/OK was sent without SDP offer.
200/OK must carry SDP offer if INVITE had no SDP.
Gateways RSTP: permit to select type of ARP packets sent to notify about changed IPv4/ethernet address assignment
License invalidation: Display warning if invalidation was not accepted
License invalidation: Display warning if invalidation was not accepted.
Trap on SOAP initiated transfer of calls connected to a Waiting Queue
The tested szenario was a transfer of an incoming call to a WQ to an outgoing call from the same WQ
PBX SOAP: Remote number update missing on blind transfer on another PBX
The CT-COMPLETE facility used to transmit the new number, was not used to update SOAP call
H.323: Don't forward invalid G.711 channel with rate=16000
Such a channel is sent by some SIP equipment
SIP: Wrong ptime negotiated sometimes
SIP: Wrong ptime negotiated on offer/offer collision.
CDRs: Forward information missing on CDRs generate for a call which was diverted to the user and then diverted to the next
In this case the CDR at the user did not show that the call was already diverted to this user.
Phones: Fix for fkey "Bool Objekt"
Toggle between the "Manual Override Off" and "Manual Override On" states only, if fkey has been configured for these 2 states only.
According to documentation:
If there are no Text, Icon and LED settings for both the Automatic Off State and Automatic On State state, the function key will toggle between the Manual Override Off and Manual Override On states only.
Gateway/H.323: Local signaling port configuration caused regsitration to be restarted on any config change
Even if only a Route was changed, the registration was restarted and any call disconnected
myPBX: Chat messages sent while the destination has not responded, got lost if call to different PBX
Problem in the PBX to PBX signaling
SIP: Config option /session-expires does not overwrite value received with INVITE
Config option /session-expires does not overwrite value received with INVITE.
Leak on certain CF errors
A leak has been fixed. The error itself now contains some more usefull data.
IP-DECT: Forced logout does not store CKI
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.
SIP: Do not send re-INVITE for T.38 if T.38 is not enabled on interface
Do not send re-INVITE for T.38 if T.38 is not enabled on interface.
For Local-Media and Media-Relay it already worked this way.
Now even on Remote-Media (Transit) interfaces a switch to T.38 is blocked.
PBX: After Export/Import Objects without devices had a default device
A no-dev Attribute was missing in the export
PBX: Max Call Duration setting did not work for call, with all legs incoming
The assumption that there is always an outgoing call leg, was wrong.
myPBX Android should not turn to restricted mode during a call
During a call myPBX Android should not apply restricted mode even if the keyguard became active. It's because the keyguard may also come in place while the phone is held to the ear since we allow Android to switch the screen off then and this in turn activates the keyguard. The user expects full functionality after removing the smartphone from the ear.
PBX Waiting: Sometimes not all members of primary group were called, when blocked because of presence
Calls need to be retried to operators which have been blocked because of presence once.
SIP: re-INVITE for T.38 must be rejected if T.38 is not enabled on this interface
Was not rejected on Media-Relay if SRTP was active.
PBX: Reporting license counting wrong
If a user confiuguration was changed while calls were active, it could happen that an additional reporting license was acquired, which was never released.
PBX Waiting: No inband call progess indication on DTMF forwarded calls
The media channel was not connected to the outgoing call
CF/SATA driver: Disturbs Linux SATA driver at start-up
The innovaphone CF/SATA driver can disturb the Linux SATA driver at Linux start-up, Linux recognizes a spurious interrupt and disables wrongly the SATA interrupt. The SATA device doesn't work or works slowly. This is fixed now.
myPBX: TLS connections for Remote Media blocking sometimes
Error in the usage of the Schannel API.