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)
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
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
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.
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
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
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.
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 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
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
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.
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
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.
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
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
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
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
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 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
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)
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
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
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)
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
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
Keep registration state on "UP" even if timeout (no-response) on call signaling. Kicking registration is only required if alternative registrar address is available.
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
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
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
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.
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
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 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
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
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
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
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
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
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
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
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.
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
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).
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.
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
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
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
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
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
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
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
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.
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
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
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
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
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.
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.
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.
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
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
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).
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
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
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
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
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 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
The SDP "s=" line conveys the subject of the session, which is reasonably defined for multicast, but ill defined for unicast. For unicast sessions, it is RECOMMENDED that it consist of a single space character (0x20) or a dash (-).
PBX Waiting: Trap on collision of call disconnect and transfer of the call from Waiting using SOAP
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.
IP-DECT: Display release cause of rerouted and rejected call
Partner keys with presence subscription enabled receive the partner's display name from PBX. Label text can be omitted. I omitted partner's display name is used as label text.
SIP: Interworking of Call-ID and conferenceID of H.323
Network settings like DHCP, interface IP addresses and the VLAN config was auto-saved after a short timeout and triggered the "Reboot now ?" question. This is not suitable of course. The settings should only be saved on return from the submenu.
The background noise level was pretty high at the handset speaker due to a high gain in the amplifier stages independent of the volume setting and due to the related quantisation noise at the DAC for low volumes. This cannot easily be changed because the LEC requires a constant loop condition. But it could be improved by 7 dB through a higher gain at the equalizer filter that allows a respective lower gain at the DAC output. The only thing is that specific equalizer filters had to be created for the maximum and max -3dB volume steps that sacrifice equalisation in favour of not exhibiting saturation.
H.323: Calls failed if many calls received at the same time
Since the optimization of the timer system the factory reset never completed because it enqueued a timer with duration 0xffffffff and this was not sorted correctly into the btrees. Lowered the duration to 2000000000 but also prepared the platform part for durations above 0x80000000.
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.
IP222/111/112: Could not enter '_' via numeric keypad
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.
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.
IP222/232/111/112: Could not enter 'Q', 'W', 'X' when phone language was Cestina
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.
Dont delete registration on server side when endpoint is not subscribed for "reg". Endpoint will not know about end of registration. In case PBX object has changed.
http: update HTTP-PORT and HTTPS-PORT variables on each change of port number or 'Force HTTPS' flag
Config option "Hide Complete Configuration at Phone" removes all apps from Phone-UI. Only Phone-App was left on display. This config option must only hide some entries from Config-App.
myPBX Android/iOS: Didn't show the dial pad for consultation call
If a consultation call was started by putting a call on hold and just pressing a digit myPBX switched to the directory screen right but didn't show the dial pad.
Phone UI: Long press on call forwarding target didn't forward
When behind a symmetric NAT router (no full-cone) it's better to write private ports into SIP messages (Contact-URI) than mapped ports. Because mapped ports cannot be used by remote side in case of symmetric NAT. When remote side sees private ports, it can workaround NAT by sending to source port.
Fixed inactive Chat/Phone icon in office applications
Handling of incoming re-INVITE after REFER has been sent: In case of attended transfer (two calls) it is accepted as usual. In case of unattended transfer (one call) it is taken as error (failed transfer).
Wait until the problem happens and do a pcap trace. Afterward you can reset
The second way is to reset the box in case of a DSP queue problem. In this case pcap traces are lost, only text messages are available after the reset: 1) Disable PCAP 2) Configure the DSP driver to write text messages 2) Configure the DSP trace-off option http://172.16.9.48/!config change AC-DSP0 /trace on /ctrace on%20/dtrace on%20/txt-trace on 3) Write and activate http://172.16.9.48/!config write http://172.16.9.48/!config active
Wait until the problem happens. Then the box resets and you can use the browser to read the trace. Note that the txt-trace option causes some overhead, so disable the tracing once the problem is solved.