If a participant starts a 3rd party conference while it is already receiving a video conference stream, it has to adapt the conference stream to fit the encoder size. We have to downscale the image.
H.323: Automatically connect signaling TCP if NAT router is detected
When regestering an endpoint from a private network to a PBX within the public network, the signaling TCP connection must be established and maintained by the endpoint. Otherwise calls to the endpoint are not possible.
The QSIG standard defines to use Channel numbers (1-30) instead of timeslot (1-15, 16-31) as it is defined for EDSS1. There are many 'old' QSIG implementations around, which do it wrong. The QSIG-ECMA1 protocol setting is used for these 'old' implementations and the QSIG-ECMA2 setting for standard conform inplementations.
With the QSIG-ECMA1 also 'old' facility coding is used. There is also the combination of standard facility coding and timeslots for channels around so an independent mechanism to configure the channel numbering is needed.
PBX: Support of long user-user-informations by SOAP
Support of long user-user-informations (UUI) for SOAP sessions added. A long UUI is split into multiple short UUIs supported by Q.931. It is required by the FAX interface.
Relay: Support of long user-user-informations by FAX
The passive Layer2 interop flag is needed if the other side (e.g. the ISDN network) does not support to a have an active Layer2 (LAPD) on point to point links all the time, but controls establishment and release on a per call basis. This could generate an Alarm each time there is no call.
PBX Waiting: Support of RTP-DTMF for two-stage dialing
If the announcement interface for a Waiting queue is not the PBX internal one and the calling endpoint can only do RTP-DTMF, the announcement interface sends back the received DTMF as USER-INFO to the PBX. This feature ist needed for a Hosting Scenario where the PBX is located in a private network and the Media interface on a different system in the public network.
Gateway: Only transparent (clearmode) coder in offer if data call
This could cause several problems: - When the call was sent to a local user with multiple registrations, the call to each registration had a different conferenceID, so myPBX could not match these calls to actually being only a single call, so multiple calls were dissplayed - The CDRs created for this call could not be matched
If the FAX interface is used to receive a FAX document with ECM mode and the transmitting terminal appends additional EOLs, the page counter is wrong and document pages are not written. This is fixed now.
SIP: No T.38 parameter when indicating capabilitity only
If a setting for a user/group was deleted, it could happen that some of the settings (Group, Online, Presence, Dialog, Ids) where copied to the next entry.
Media: Redirecting SRTP streams for NAT clients only after successful SRTP authentication
Media endpoints support NAT. If receiving RTP/SRTP from an address other than negotiated one media endpoints redirecting their media stream towards source of incoming media stream. In case of SRTP, this NAT workaround is only executed if incoming media stream has passed authentication. For securitiy reasons.
Office2010 Integration: Own presence did always show the IM presence
Focusing invisible elements causes an exception in IE8
Zeile: 386 Zeichen: 9 Code: 0 Fehlermeldung: Das Steuerelement kann den Fokus nicht erhalten, da dieses unsichtbar oder nicht aktiviert ist oder keinen Fokus zulässt. URL: http://xxx.xxx.xxx.xxx/PBX0/MY/mypbx10_im.js?lang=de
If the connection between an Office application and myPBX gets lost, the COM Server is now restarted. This forces a reload of the web UI and a reset of the video connection too.
Especially POE-switches with higher supply voltages than 48V lead to a decreased timespan of powering the build-in relays of a ip6010/ip810 gateway. The detection of a power-fail condition is therefore derived from the POE ICs which react earlier and thus increases powering time of the relays.
myPBX: Default group visibility was not displayed correctly
The default group visibility can be configured from v9hotfix17. Regardless of that configuration myPBX showed full visibility in the visibilty settings.
IP0010 IP1060 IP3010 IP6010 IP810 IP22 IP24 IP28 IP302 IP305: DSP Update to Version 680.07
- Deactivate tooltips for presence description when presence drop down is active. - Remove tooltips for prompted input fields. They were always the same as the displayed text. - Remove tooltips for icons of drop down menu items. They were always the same as the displayed text.
Needed for SIP interoperability. Some third party SIP PBXs use addresses of 0.0.0.0 to indicate that they don't receive media. This may happens if an endpoint is put on hold. We did not forward such an offer and thus no Music on Hold was heard.
PBX: Busy On ... Calls at PBX objects did not take into acccount that a call may be routed back to Slave
The busy on ... calls on PBX objects can be used to limit bandwidth usage between a master and a slave to a certain number of calls. Some calls are sent from a slave to the master and back to the slave if the routing decision cannot be done on the slave alone. This happens if escapes are used which overlap other obects (e.g. the local trunk). It is a common configuration the the E.164 routing scheme. With this fix, these calls are not counted for this purpose.
In case an endpoint registers to a PBX from within a private network thru a NAT router, the signaling TCP connection must be maintained in order to be able to receive calls. When the registration is up a dummy call is sent to the PBX to establish the signaling TCP. This TCP connection is maintained after the dummy call is cleared. If this TCP connection is lost (e.g. NAT Router reset), the Registration is cleared and restarted, so that after the re-registration another dummy call is sent.
This is a fix for the previous fix
fix: #89497: H.323: Automatically connect signaling TCP if NAT router is detected
which did not work well
HTTP: Force HTTPs did not work with websocket protocol
When many leaks exist or leak check is done when much tracing is turned on. The leak check itself could cause a watchdog trap, because the collecting of the leaks is done on highest priority so not even the timer interrupt could trigger the watchdog.
myPBX launcher: Possible exception when starting myPBX
Depending on the timing there could be an exception when starting myPBX. This happened because the flow for initializing the video and webcam window was unsafe.
The "Outgoing call restricted" flag on the trunk object to which the call was forked caused the call as a whole to be marked as Calling Line Presentation Restricted.
SIP: Locally configured DNS entries were not used if no DNS server configured
If no DNS server was configured, but DNS names are to be resolved, local DNS entries can be added (Services/DNS/Hosts). SIP stack fails with SRV query and does not try A query which would deliver IP address.
PBX: CDR was missing for a call rejected because of Busy on ... Calls
On the user with Busy on ... calls set no CDR was generate for a call which was rejected because of this. The user would like to see this as a missed call in this case.
PBX: Registration with name was possible even without matching device
This is a security issue, because of this it was not possible to enforce a police which allows registration by hardware id only. No fix in version 9, because of compatibility reasons.
If a remote hold event is received, no RTP data should be sent by the IP-DECT device. A CTI initiated call is established with a call transfer and a "No Media data received" error event can occur. This is fixed now.
PBX: Twin-Phones: On connect from twin-phone endpoint disconnect additional ringing calls
If two calls are ringing at a twin phone user with multiple endpoints, and no call is conneceted, these calls are ringing at all endpoints. When the first call is connected, the remaining call legs ringing at other endpoints should be disconnected.
The launcher can now detect user activity and set the IM status depending on the current state. If the user is inactive the IM status is set to closed (Appear Offline).
The feature can be turned on and off and an idle timeout can be configured using the option "Auto Appear Offline" in the configuration dialog. During installation the number of minutes can be configured using the MSI parameter AUTOAPPEAROFFLINE=[0|1|5|10|15]. 0 turns the feature off.
In some cases it is desireable not to reveal the final destination of a call to a caller. For example a call center agent should not be called directly by the customer.
myPBX launcher: Show webcam configuration at the same location as the video call window
The state of the remote video service is shown on page PBX/Registrations: - available: the phone supports remote video - connected: a remote video client is connected to the phone
The configuration option 'Registration with system password' is added. If ticked, all users are registered with the system password. This is useful, if the PBX users are only allowed to register with the PBX password.
myPBX launcher: Handle power state changes of the computer
Now the VOIP connections between the Master and the Radio use static ports instead of dynamic ones. This is useful if only a few ports should be opened through a firewall. For calls from the Radio to the Master the ports 1716 and 1717 (TLS) are used. For the default Master connection for calls from the Master to the Radio the ports 1718 and 1719 (TLS) are used. For dynamic Radio-Master connections the ports from 1722 are used. Every connection needs two ports.
There is a new option available for the CONF interface now: remote control connect (*82). If a conference call with enabled remote control connect option is made, first an alert event is sent to the caller of this conference room. The call is only connected if the innovaphone remote control connect (0) is received by the CONF interface. See also: http://wiki.innovaphone.com/index.php?title=Reference10:Gateway/Interfaces#Call-Setup_Commands
If a call is answered on the mobile phone, it should look identical to the caller to the case that the call was answered locally. This means a connected number from the mobile phone must not be forwarded.
PBX Waiting: Potential Trap if editing while a call is initiated with SOAP
The Waiting object can be used as outgoing dialing object with SOAP. If this is done and the configuration is changed while an outgoing call was pending, a trap could happen
H.323: Potential Trap in special case which could only happen in version 10
PAI/PPI was processed when receiving UPDATE without SDP offer. PAI/PPI was ignored when receiving UPDATE with SDP offer. Now PAI/PPI is processed when receiving UPDATE with SDP offer.
There was a race condition when switching local media channels (e.g. ISDN channels to conference interfaces), which could cause media not functioning or even a trap
PBX CSV Import: Corrupted objects at buffer boundaries
When doing a reset to manufacturing defaults (long reset) an address of 192.168.1.1 is configured for ETH1. This address was removed when the gateway was configured with the wizard.
Phone: Trap when selecting registration for a directory entry
State 25 is incoming overlap sending. This means a call was received with incomplete dialing information and the caller failed to dial more digits within the timeout of 2min. This is no indication of any malfunction but only a usage problem, so no event should be generated.
State 11 is disconnecting with inband announcement. A timeout happens if a user listens to the announcement for more then 30s. This could be normal.
Media negotiation for video fails if called through waiting queue or multi reg. In this case the PBX has to handle offer/offer-collision. In this case the PBX must select audio and video codec. In this case the PBX must send SDP answers to both endpoints.
Usually a response to a SRV query delivers additional records containing the ip address of any target (hostname). Some DNS servers do not. Additional A querys are required. An A query was issued for the primnary target (most preferred hostname). No A query was issued for the secondary target (less preferred hostname). Fixed now.
A trap in the IP-DECT Radio module occurs if the Mobility Master is used and a duplicate IPEI command is sent to the Master. The Master handles it with a location cancel and an endpoint delete command sent to the radio. If the two commands arrives with no delay, the Radio module traps. This is fixed now.
myPBX: Prompted input fields lost title when logging-out
This happened with H.323 connections without registration when disconnecting a call with inband information (e.g. a call to an ISDN interface). Unnecessary events were generated.
Video: disable stand by/sleep modus during a video call
although we have tested more than 30 Webcams from different vendors we may find a webcam with other output formats or resolutions. Write more traces to have this information available.
In some cases SRTP calls had one-way media because the RTP sequence number wrapped from 65535 to 0 at be beginning of the call before the receiver started receiving and processing packets.
The scope of start sequence numbers for RTP streams is changed from [0;65535] to [0;32767] to make sure that the receiver can always receive packets before the overflow happens.
The calculation of the roll-over counter (ROC) is also improved to be more reliable.
H.323: Unnecessary re-initializing of rtp-channel on incoming calls to phone
This did not create any problems except CPU load and together with another problem in RTP it caused no media on incoming SRTP calls approximately every 1000th call.
IP22,IP24,IP28,IP302,IP305: RTP-DTMF not offered when using a/b interface
The MOH URL Paramter (%l, %h, %n, ...) can be used to use different MOH Files based on the User who is holding the call. In case of a parked call this should refer to the object where the call is parked, not to the user who has initiated the parking.
If SRV query returns 2 hosts with different port, but no IP address in additional records, SIP starts two A queries for the two host names. Both resolved IP addresses are combined with the port of the most preferred host of the SRV answer.
IP-DECT: Hold/Retrieve could result in no media for incoming SIP calls with SRTP
Cipher key index request procedure is changed to pass the test with security test devices. The cipher key index is used for DECT "Early Encryption"(EE).
Some networks e.g. sip carriers behave badly when receiving subscribes for presence/dialog-info, which cannot be handled, so there is an option added to block these.
There are new innovaphone remote controls available for the CONF interface: - 24: Receive on - 25: Receive off - 26: Exclusive listen mode - 27: Normal listen mode
Windows 8 defines new properties for the h264 encoder and decoder like CODECAPI_AVLowLatencyMode which removes the delay added by the windows h264 decoder.
PBX: For calls accepted somewhere else, provide this information within signaling, to allow improved calls lists
A call which is accepted somewhere else, because it was a call to a broadcast group, or a pickup was done, does not show up in the call lists on the phone. It is good that such a call does not show up as missed call, but it could be useful if it could be seen as connected call. To allow this, the PBX provides the information within the signaling (diverting leg4 info).
Bug Fixes
SIP/WLAN: Keep local Contact-URI up-to-date on subscriptions
Reading SIP messages from TCP stream gets confused by huge SIP messages. Presence exchange with external UC was disordered. Increased size limit from 100KByte to 200KByte.
Gateway: Transmitting FAX documents to receiver with polling mode
If a Voicemail license is installed the Voicemail checkmark on the user license page is displayed disabled and checked, to indicate that no VoicemailUser license is needed
The cipher key index table is wrongly updated in the Crypto Master if a entry line yet exists. This is fixed now. The Crypto Master is needed for DECT Security Early Encryption.
IP222 IP232: Handset gains changed to avoid low microphone volume
When tandeming VOIP links for trancoding or other purposes DTFM digits were sometimes duplicated. The RTP carried up to 25ms DTMF remaining DTMF, now its only 16ms.
PBX: Trap if user object is deleted, which is used by other applications (e.g. myPBX)
$_leg2tweak -- Controls <pbx-getcallinfo out-leg2=".."/> true(default): set leg2 to <ext-nr> from <vm-nr>+<ext-nr> false: set leg2 according to received divertingInfoLeg2 facility
$_trailhash -- Controls <pbx-getcallinfo out-cdpn=".."/> true: pass trailing (en-bloc) '#' into cdpn false(default): don't pass trailing (en-bloc) '#' into cdpn
Call transfer with enbloc dailing fails. This is fixed now. This changes also the R-key handling: after dialling a digit for a consultation call the call must disconnect with R-1 like in ring-back state.
SRTP: Remove traces when packet authentication failed
SRTP and SRTCP software encryption produced traces when packet authentication failed. This is not needed, because an event is created anyway, when this happens frequently.
the initialisation fails with CC=5 on first device descriptor read. after restart of host controller serial_irq() traps in reading the done list. Happens mostly with upload DRAM.
IP-DECT/Relay: Blocked calls by hidden feature code *5/*7
Outgoing calls with beginning number *5 or *7 are blocked by the feature codes module because of hidden new service codes for an OEM device (#79028). This is fixed now.
The registration of unknown endpoints worked, but when the number was dialed to assign the registration to a user object, the unknown registration remained in the list.
PBX: Master Slave license update period 10s instead of 10min
Usability optimizations for the current functionality: - Display people only once - Do not display empty lines - Use keyboard to select entries. Dial the selected entry, not the content of the input field - Possibility to edit numbers before dialing.
Video: support for webcams delivering h264 (only Windows 8)
The innovaphone Fax Server is using User-User-Info to send commands to the Fax interface. The standard config is, to register the Fax interface to the local PBX and this worked. If the call to the Fax interface was routed thru multiple PBX, this User-User-Info get lost in Alerting state
Video: allow to change webcam format during the call
If the master was configured with non-standard ports or was a dynPBX, the port or module part of the URL was wrong. With the new boolean argument tls, a https URL can be requested
Fax-server-application: HTTPS support for Slave PBX connection
User-User-Info response of the FAX interface is not forwarded in the alerting state. The problem exists when call to Fax interface was routed through multiple PBX. Now the UUI response is sent in the disconnect event if the response is a error notification.
PBX: Calls page did not show all calls expected, esspecially in mobility context
Clicking the number copies it to the search input field. There it can be edited an additional information (presence, contact data) is retrieved from the PBX and LDAP.
myPBX launcher: Adapt video windows to new UI design
Parameters: /multi - allow starting multiple instances, disable saving configuration /url - URL of the myPBX web application /user - user name for the myPBX web application /password - password for the myPBX web application
In the config menu the user can now choose if myPBX should be started as a normal window or as an AppBar attached to the right or left border of the screen.
PBX Executive: Allow monitoring of availability of secondary secretary, don't treat Exec as secretary
With these two additions a configuration with two executives and two secretaries, each secretary being primary to one executive an secondary to other can be configured with a single group for each secretary and both executives can monitor the availability of both secretaries.
There is a new configuration option 'Display Original Called'. If it is enabled, the original called instead of the diverted party is shown if the call is diverted.
Bug Fixes
myPBX: Redirect to different PBX does not support non-standard ports
These functions are responsible of the displaying part. So far I stopped the connection process if any of these functions failed to complete. Now, I continue with the video call and I try to initialize these functions every second. Video preview and output video will be shown when these functions succeed.
Gateway: CGPN-Maps executed even if the Route did not match in case of enbloc calls
If one tried to delete a large amount of CDRs (e.g. > 100000), the process failed without an error message. This is fixed now, although the process lasts longer now.
LDAP Client: SearchRequest.derefAliases Changed To neverDerefAliases(0)
After a change in the decoded resolution (for example 352x288 -> 176x144) the display area should be reconfigured before displaying the new decoded frames.
If the shown web report is too large, some browsers are crashing. This is now prevented by a limit to the shown calls. The report itself can still be directly downloaded without this limit.
PBX Waiting: Presence set for operator was not cleared, on delete or editing of Waiting Queue object
There are Fax devices sending wrong (too long) message initially after being called. In this case it is best handled by ignoring these message and wait for the retry instead of disconnecting the call.
myPBX: Improve presentation of monitored calls of favourites
- Alerting calls that cannot be picked are displayed using a red bell symbol. - Own calls of the user are now detected more acurately. - The pickup button does not jump on mouse-over any more.
SIP/UDP: Sending response to wrong address and port
The availability state (secretary booked into the exec primary group) was not associated with the correct secretary. A compare of the names only covered the first half of the name.
Fixes for the main window: - Minimum window size - Window appears in the ALT-TAB list - Normal window style instead of tool window style Fixes for the config window: - Hide maximize button - Hide minimize button
Video: packets were drop if they arrived before connect
a customer has problems with LDAP resolution and we guess the problem resides on the % character. Now the filter contains both cases, number with and without percentages.
myPBX: Exclude non-numbers from LDAP search in number attributes
The physical location information is based on the redirection of the registration from the PBX at the physical location to the registration PBX. Some information was not cleared with the logout, so re-registration startet with the registration PBX right away.
phone: if a number to be dialled contains a comma, the digits following the comma are sent as DTMF tones after connect
This applies to all numbers dialed en bloc, i.e. numbers dialed via indirect dialing, a phone directory or a function key. The comma must not be the first character of the number.
The Mobility Master stores also the default cipher key of the handsets now.
This accelerates the access with early encryption in systems with sites connected to a remote Master over slow WAN links by configuring a local Mobility Master. Only the first access of the cipher key is done over WAN, further accesses can be served by the local Mobility Master over the much quicker LAN.
The redirecting number is an old style information element, which contains part of the information as the diverting leg2 facility. Some Fax Servers do not understand the leg2 facility.
The myPBX call list authentication only worked with the Linux Web Server credentials, although it is possible to configure separate Reporting access credentials. Now both credentials will work.
Remote video needs the local IP address for signalling. Some anti virus software sets the local IP address to 127.0.0.1. In this cases the remove video websocket connection should fail.
PBX Trunk: "Outgoing Calls restricted" did not work correctly, Presentation restricted was set, but number could be wrong
For example if an analog Gateway was registered to a PBX user, and this Gateway did not send a Calling Party Number with the call, the call was sent with Presentation restricted, but without digits. This could affect Billing Applications which are based on CDRs from the Gateway.
Video: directx fails if pc does not support hardware vertex processing
CreateDevice function from direct3d library fails if hardware vertex processing is not supported. If this happens, try with software vertex processing afterwards.
phone: ip222,ip232: inbound calls automatically connected to Plantronics Savi W440/740/745 headsets with new firmware Versions
reported for: - Savi W440 with firmware 0118 on USB/DECT Dongle D100 - Savi W740/745 with firmware 0115
reason: the newer firmware versions reject truncated output reports (no trailing 0 bytes) with STALL. The error handling for this case was wrong and caused an autoconnect.
IP-DECT: Wrong GK id of standby Master to Mobility Master
- Deactivate the timeout while the activity drop-down is open. - Submit presence when the input field looses the focus. No timeout needed here. - Do not select text in note input when the activity drop-down is activated.
Video: increase the number of slices for h264 decoding
So if the name of the PBX registering as slave was changed as well, it did not register anymore. The PBX object had to be deleted and created with new name.
A registration may be impossible due to network config (no route to destination). SIP stack must tell the application about this error. ALARM should be set.
Phonesig: Disconnect remote video on CHANNEL_INIT with CHANNEL_CODER_UNDEFINED
The web application should use the same protocol (HTTP or HTTPS) for talking to the PBX and to the reporting. This helps avoiding browser warnings.
- Remove possibility to configure LOCAL-AP-S and REMOTE-AP-S - The PBX sends secure and unsecure URLs with the update-reporting message - The webapplication chooses the right URL depending on the used protocol (HTTP or HTTPS)
myPBX Launcher: Remove "always on top" in docking mode
Esspecially on the second hold within a call the Channel Close was not sent to the party, which put the other on hold. This caused the channel not beeing turned off on this side (the other side receives music on hold in this case)
IP4 did not work anymore when IP6 was disabled via WEB interface
Only the dialed number was available. This was for example a problem for applications with the CONF interface which provides the room number as connected number
phone: ip222,ip232: Plantronics Savi W440 dosn't report Talk-Key events in a call established at phone or by a CTI application
When a call via this headset was initiated/accepted by the Redial-Key, the Headset(Mode:Control) function key or a CTI application, the call could not be disconnected by pressing the Talk-Key at the headset because the Headset did not report this action.
myPBX: Avoid confusion of client.htm and start.htm
The command mod cmd UP1 provision <seconds> sets the poll interval for the next poll cycle after completion of the current script to the given number of seconds (max 60). This interval is duplicated after each failing poll. The command mod cmd UP1 eval <var-name> performs a variable substitution on the value of the variable "UPDATE/USER/<var-name>" and writes back the new value
The command mod cmd UP1 provision <seconds> sets the poll interval for the next poll cycle after completion of the current script to the given number of seconds (max 60). This interval is duplicated after each failing poll. The command mod cmd UP1 eval <var-name> performs a variable substitution on the value of the variable "UPDATE/USER/<var-name>" and writes back the new value
New config file option /register-interval 60 Problem is too weired to explain. This option can be used to set the REGISTER interval to a fixed value regardless of the negotiation.
New config file option /register-interval 60 Problem is too weired to explain. This option can be used to set the REGISTER interval to a fixed value regardless of the negotiation.
Phone: Removed "Meeting" from default set of presence activities
This behaviour can be enabled via config add PHONE APP /auto-handsfree <digits> where <digits> is the sequence of all digits which shall trigger handsfree mode, for example config add PHONE APP /auto-handsfree 0 To disable this behaviour use config rem PHONE APP /auto-handsfree
phone: old fashioned feature - automatically enter handsfree mode when a certain digit is entered when phone is idle
This behaviour can be enabled via config add PHONE APP /auto-handsfree <digits> where <digits> is the sequence of all digits which shall trigger handsfree mode, for example config add PHONE APP /auto-handsfree 0 To disable this behaviour use config rem PHONE APP /auto-handsfree
PBX Waiting:Operator mobile phones are called thru Mobility
- in the "Destination Number" configured under "Phone/Direct Dialing" in conjunction with a nonzero "Autodial Timeout": the DTMF digits were sent as dial digits - with a nonzereo "Enblock Dialing Timeout" configured under "Phone/User x/General/Options": sending of DTMF digits was delayed by the configured timeout\t
phone: DTMF digits following a comma in a number to be dialed were not handled correctly in some cases
- in the "Destination Number" configured under "Phone/Direct Dialing" in conjunction with a nonzero "Autodial Timeout": the DTMF digits were sent as dial digits - with a nonzereo "Enblock Dialing Timeout" configured under "Phone/User x/General/Options": sending of DTMF digits was delayed by the configured timeout\t
Services/DNS/Hosts: SRV records identified by triple: name,target,port
observed with myPBX (user A): 1. A calls B, B accepts 2. A puts B on hold 3 A calls C, C accepts 4. A sets up a 3pty conference 5a. A puts C on hold, no media 5b. A puts B on hold, OK
phone: ip222,ip232: setting one peer of a 3pty conference on hold via PBX may switch off media for the other peer
observed with myPBX (user A): 1. A calls B, B accepts 2. A puts B on hold 3 A calls C, C accepts 4. A sets up a 3pty conference 5a. A puts C on hold, no media 5b. A puts B on hold, OK
PBX: Name-Id of busy destination was not forwarded to other PBX
If there is for instance a change in the video resolution, the display driver must be reinitialized. During this process, access to this driver must be disallowed.
If there is for instance a change in the video resolution, the display driver must be reinitialized. During this process, access to this driver must be disallowed.
IPVA: Unused ETH1 Could Cause Out-Of-Memory Situation
Observerved with with headsets from various manufacturers. Most probably caused by strong electrostatic discharges to the USB connection cable. In such case also unplugging/plugging of the headset was not noticed by the driver anymore.
phone: ip222,ip232: USB connection sometimes lost until reboot
Observerved with with headsets from various manufacturers. Most probably caused by strong electrostatic discharges to the USB connection cable. In such case also unplugging/plugging of the headset was not noticed by the driver anymore.
Logging: "Alarm and Event Forward Server" address could not be changed anymore once configured
The command mod cmd UP1 provision <seconds> sets the poll interval for the next poll cycle after completion of the current script to the given number of seconds (max 60). This interval is duplicated after each failing poll. The command mod cmd UP1 eval <var-name> performs a variable substitution on the value of the variable "UPDATE/USER/<var-name>" and writes back the new value
The command mod cmd UP1 provision <seconds> sets the poll interval for the next poll cycle after completion of the current script to the given number of seconds (max 60). This interval is duplicated after each failing poll. The command mod cmd UP1 eval <var-name> performs a variable substitution on the value of the variable "UPDATE/USER/<var-name>" and writes back the new value
New config file option /register-interval 60 Problem is too weired to explain. This option can be used to set the REGISTER interval to a fixed value regardless of the negotiation.
New config file option /register-interval 60 Problem is too weired to explain. This option can be used to set the REGISTER interval to a fixed value regardless of the negotiation.
Phone: Removed "Meeting" from default set of presence activities
This behaviour can be enabled via config add PHONE APP /auto-handsfree <digits> where <digits> is the sequence of all digits which shall trigger handsfree mode, for example config add PHONE APP /auto-handsfree 0 To disable this behaviour use config rem PHONE APP /auto-handsfree
phone: old fashioned feature - automatically enter handsfree mode when a certain digit is entered when phone is idle
This behaviour can be enabled via config add PHONE APP /auto-handsfree <digits> where <digits> is the sequence of all digits which shall trigger handsfree mode, for example config add PHONE APP /auto-handsfree 0 To disable this behaviour use config rem PHONE APP /auto-handsfree
PBX Waiting:Operator mobile phones are called thru Mobility
- in the "Destination Number" configured under "Phone/Direct Dialing" in conjunction with a nonzero "Autodial Timeout": the DTMF digits were sent as dial digits - with a nonzereo "Enblock Dialing Timeout" configured under "Phone/User x/General/Options": sending of DTMF digits was delayed by the configured timeout\t
phone: DTMF digits following a comma in a number to be dialed were not handled correctly in some cases
- in the "Destination Number" configured under "Phone/Direct Dialing" in conjunction with a nonzero "Autodial Timeout": the DTMF digits were sent as dial digits - with a nonzereo "Enblock Dialing Timeout" configured under "Phone/User x/General/Options": sending of DTMF digits was delayed by the configured timeout\t
Services/DNS/Hosts: SRV records identified by triple: name,target,port
observed with myPBX (user A): 1. A calls B, B accepts 2. A puts B on hold 3 A calls C, C accepts 4. A sets up a 3pty conference 5a. A puts C on hold, no media 5b. A puts B on hold, OK
phone: ip222,ip232: setting one peer of a 3pty conference on hold via PBX may switch off media for the other peer
observed with myPBX (user A): 1. A calls B, B accepts 2. A puts B on hold 3 A calls C, C accepts 4. A sets up a 3pty conference 5a. A puts C on hold, no media 5b. A puts B on hold, OK
PBX: Name-Id of busy destination was not forwarded to other PBX
If there is for instance a change in the video resolution, the display driver must be reinitialized. During this process, access to this driver must be disallowed.
If there is for instance a change in the video resolution, the display driver must be reinitialized. During this process, access to this driver must be disallowed.
IPVA: Unused ETH1 Could Cause Out-Of-Memory Situation
Observerved with with headsets from various manufacturers. Most probably caused by strong electrostatic discharges to the USB connection cable. In such case also unplugging/plugging of the headset was not noticed by the driver anymore.
phone: ip222,ip232: USB connection sometimes lost until reboot
Observerved with with headsets from various manufacturers. Most probably caused by strong electrostatic discharges to the USB connection cable. In such case also unplugging/plugging of the headset was not noticed by the driver anymore.
Logging: "Alarm and Event Forward Server" address could not be changed anymore once configured