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(RE-INVITE During Recording Sessions Leak Memory. (MTN-1317))
 
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[[Category: Release notes]]
 
[[Category: Release notes]]
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Previous Release - [[AlphaCom 11.05 - Release Notes]]
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Next Release - [[AlphaCom 12.05 - Release Notes]]
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This document provides the release notes for AlphaCom 12.3 with incremental bug fix releases. The release notes for AlphaCom 12.3 describe new features, improvements and issues fixed after AlphaCom 11.5.
 +
 
'''Software in production:''' AMC 11.2.3.10<br />
 
'''Software in production:''' AMC 11.2.3.10<br />
 
'''Note:'''  For each software version the NVRAM version is listed.   
 
'''Note:'''  For each software version the NVRAM version is listed.   
Line 8: Line 14:
  
  
 
+
=AMC 12.3.3.4 (LA)=
= AMC 12.3.3.4 =
+
{{Note| INCA based Intercom stations with software version 01.x.x.x are not supported from this AMC version onwwards. Any such station must be upgraded to version 02.x.x.x
 +
}}
 
  Release: Official, available on request
 
  Release: Official, available on request
 
  NVRAM version 12.3
 
  NVRAM version 12.3
  Date: x th of November 2017
+
  Date: 29th of November 2017
  
 
'''System upgrade file: bsp_32.00.3.1_AS_12.3.3.4.apkg'''<br />
 
'''System upgrade file: bsp_32.00.3.1_AS_12.3.3.4.apkg'''<br />
Line 22: Line 29:
  
 
==Errors Corrected==
 
==Errors Corrected==
=== SIP: Support of Early Media "183" (MTN 1371)===  
+
===SIP: Support of Early Media "183" (MTN 1371)===  
 
Incoming Calls from SIP to AlphaCom did not allow Early Media.
 
Incoming Calls from SIP to AlphaCom did not allow Early Media.
  
=== Camp on Busy (MTN-1373)===
+
===Camp on Busy (MTN-1373)===
 
In some camp on busy situations a system reset could occur.
 
In some camp on busy situations a system reset could occur.
  
=== Silence Call not Working (MTN-1390, MTN-1395)===
+
===Silence Call not Working (MTN-1390, MTN-1395)===
 
The 11.5 data commands and parameters for silence call setup using $CALL and $SC are now included in the 12.3. software.
 
The 11.5 data commands and parameters for silence call setup using $CALL and $SC are now included in the 12.3. software.
  
=== Issues with two Simultaneous ReCall Features (MTN-1359)===
+
===Issues with two Simultaneous ReCall Features (MTN-1359)===
 
Reset could occur if dispatching two simultaneous ReCall playbacks to groups, where groups contains overlapping members and AlphaPro missing members configuration > 0.
 
Reset could occur if dispatching two simultaneous ReCall playbacks to groups, where groups contains overlapping members and AlphaPro missing members configuration > 0.
  
=== BUFFER_TOO_SMALL SysLog Messages (MTN-1405) ===
+
===BUFFER_TOO_SMALL SysLog Messages (MTN-1405)===
 
When reporting absence status a "Buffer too small" debug SysLog message could occur. Previously fixed for the 11.3 software.
 
When reporting absence status a "Buffer too small" debug SysLog message could occur. Previously fixed for the 11.3 software.
  
=== Avoid Tone Test and Scheduler During IPHA Standby Mode (MTN-1256) ===
+
===Avoid Tone Test and Scheduler During IPHA Standby Mode (MTN-1256)===
 
Both the scheduler and the Tone test was running in an AlphaCom in standby mode. <br>
 
Both the scheduler and the Tone test was running in an AlphaCom in standby mode. <br>
 
This would generate syslog messages.<br> Now Tone test and Event Scheduler is stopped during standby mode.
 
This would generate syslog messages.<br> Now Tone test and Event Scheduler is stopped during standby mode.
  
=== SIP Station Timeout on Collect Digit Features (MTN-1410)===
+
===SIP Station Timeout on Collect Digit Features (MTN-1410)===
 
SIP stations dialing feature 81 / 83 configured with collection of digits would timeout before the digits where sendt to the SIP trunk.
 
SIP stations dialing feature 81 / 83 configured with collection of digits would timeout before the digits where sendt to the SIP trunk.
  
=== Remove Directory Numbers from the 614 Menu (MTN-1389)===
+
===Remove Directory Numbers from the 614 Menu (MTN-1389)===
 
These numbers are removed in Autoload: 54, 55, 0, 627, 628, 8270, 8260-8266.
 
These numbers are removed in Autoload: 54, 55, 0, 627, 628, 8270, 8260-8266.
  
=== Ringing Group from SIP Trunk (MTN-1331) ===
+
===Ringing Group from SIP Trunk (MTN-1331)===
 
It was not possible to use ringing group calls from SIP in combination with node flag "180 Ringing".
 
It was not possible to use ringing group calls from SIP in combination with node flag "180 Ringing".
  
=== Busy Override to Ringing Group (MTN-1357) ===
+
===Busy Override to Ringing Group (MTN-1357)===
 
It was not possible to do busy override of stations already in call from answering Ringing Group call.<br>
 
It was not possible to do busy override of stations already in call from answering Ringing Group call.<br>
 +
 
*If A is answering Ringing Group call from B. <br>
 
*If A is answering Ringing Group call from B. <br>
 
*C is not able to do busy override to A.
 
*C is not able to do busy override to A.
  
=== Call Forwarding (71/72) Fails for SIP Station (MTN-1361) ===
+
===Call Forwarding (71/72) Fails for SIP Station (MTN-1361)===
 
When a SIP user calls a station (ASLT or IP) which is forwarded, the call will not connect with the target station.
 
When a SIP user calls a station (ASLT or IP) which is forwarded, the call will not connect with the target station.
  
=== Dummy Stations as Default SX Speaker (MTN-1353) ===
+
===Dummy Stations as Default SX Speaker (MTN-1353)===
 
Sometimes the audio was not connected correctly from a Dummy IP talker of a SX conference.
 
Sometimes the audio was not connected correctly from a Dummy IP talker of a SX conference.
  
=== Wrong Default CRM Notification Tone (MTN-1327) ===
+
===Wrong Default CRM Notification Tone (MTN-1327)===
 
CRM notification tone is now default setting.
 
CRM notification tone is now default setting.
  
=== Avoid Overlapping Talkback UDP Ports for Mixer Station (MTN-1350) ===
+
===Avoid Overlapping Talkback UDP Ports for Mixer Station (MTN-1350)===
 
Sometimes there could be conflicting use of UDP ports for talkback and multicast SX conference streaming when using the Mixer Station. <br>
 
Sometimes there could be conflicting use of UDP ports for talkback and multicast SX conference streaming when using the Mixer Station. <br>
 
This would result in no talkback audio from the Mixer Station.
 
This would result in no talkback audio from the Mixer Station.
  
=== Send Private Info to IP Station when Using Feature 80 (MTN-1348) ===
+
===Send Private Info to IP Station when Using Feature 80 (MTN-1348)===
 
Using the feature 80 to control station private setting did not update the display of the INCA stations.
 
Using the feature 80 to control station private setting did not update the display of the INCA stations.
  
=== Missing B subscriber info, Voice Recording in Ring-Master Installations (MTN-1349) ===
+
===Missing B subscriber info, Voice Recording in Ring-Master Installations (MTN-1349)===
 
When answering call request from remote nodes, remote stations number and name  was not updated in the active proxy stations list. <br>
 
When answering call request from remote nodes, remote stations number and name  was not updated in the active proxy stations list. <br>
 
This resulted in missing info for Recording sessions and event handler using %dir/%nam.
 
This resulted in missing info for Recording sessions and event handler using %dir/%nam.
  
=== Configuration not Synchronized in IPHA  (MTN-1332) ===
+
===Configuration not Synchronized in IPHA  (MTN-1332)===
 
In situations where the configuration slave has ex_data waiting to be written to flash, the synchronization of configuration data from config master will trigger writing to flash thus overwrite new configuration. <br>
 
In situations where the configuration slave has ex_data waiting to be written to flash, the synchronization of configuration data from config master will trigger writing to flash thus overwrite new configuration. <br>
 
Now configuration slave will inhibit writing local changes to flash file.
 
Now configuration slave will inhibit writing local changes to flash file.
  
=== Issues with Ringing Group using SIP Station Members (MTN-1340) ===
+
===Issues with Ringing Group using SIP Station Members (MTN-1340)===
 
Simultaneous answer from two SIP stations of a ringing group call could in some situations resulting in Exchange reset.
 
Simultaneous answer from two SIP stations of a ringing group call could in some situations resulting in Exchange reset.
  
=== Issues with Answering Ringing Group Call Early (MTN-1023) ===
+
===Issues with Answering Ringing Group Call Early (MTN-1023)===
 
If a ringing group call was answered before first ringing sequence is finished, the ringing will continue at the other receivers of the call.
 
If a ringing group call was answered before first ringing sequence is finished, the ringing will continue at the other receivers of the call.
  
=== AlphaCom Ignores "486 Busy Here" from Connected SIP Phones (MTN-1425) ===
+
===AlphaCom Ignores "486 Busy Here" from Connected SIP Phones (MTN-1425)===
 
The "Reject"  key of the ITSV-1 SIP station would not cancel the incoming call.
 
The "Reject"  key of the ITSV-1 SIP station would not cancel the incoming call.
  
== Enhancement ==
+
==Enhancement==
=== Configuration of Common Codec for all Program/GroupCall/Conference (MTN-1374) ===  
+
===Configuration of Common Codec for all Program/GroupCall/Conference (MTN-1374)===  
 
A common setting of codec for "one to many" call types can be configured in NVRAM.<br>
 
A common setting of codec for "one to many" call types can be configured in NVRAM.<br>
 
  .ex_profile.glob_const.CP_codec = 0  
 
  .ex_profile.glob_const.CP_codec = 0  
Line 95: Line 103:
 
Using AlphaPro configuration of individual codecs for Group/SX will overrule the common setting.
 
Using AlphaPro configuration of individual codecs for Group/SX will overrule the common setting.
  
=== Added Configuration for Changing SIP Reply Codes (MTN-1331)===
+
===Added Configuration for Changing SIP Reply Codes (MTN-1331)===
 
For SIP trunk station calls, AlphaCom will default reply with code 480 in fault situation and 182 for early media. <br>
 
For SIP trunk station calls, AlphaCom will default reply with code 480 in fault situation and 182 for early media. <br>
 
  .ex_profile.ip_config.SIP_REP_480 = 0 (0x0000)
 
  .ex_profile.ip_config.SIP_REP_480 = 0 (0x0000)
Line 101: Line 109:
 
Configuring other SIP codes can be done by changing the parameters to legal SIP codes. (0 = use default)
 
Configuring other SIP codes can be done by changing the parameters to legal SIP codes. (0 = use default)
  
=== Feature for Mixer Station WorkChannel Talkback from DAK keys ===
+
===Feature for Mixer Station WorkChannel Talkback from DAK keys===
 
New feature 120, used for WorkChannel talkback from Mixer station.
 
New feature 120, used for WorkChannel talkback from Mixer station.
  
=== Support of TFIE-6 Display Update(MTN-1370)===
+
===Support of TFIE-6 Display Update(MTN-1370)===
 
The new TFIE-6 requires a special display protocol to operate.
 
The new TFIE-6 requires a special display protocol to operate.
  
=== If only one member of the age multicast bitmap, then send age message as tcp DIP (MTN-1399)===  
+
===If only one member of the age multicast bitmap, then send age message as tcp DIP (MTN-1399)===  
 
Optimization: using TCP to single stations avoiding multicast repetitions to groups of only one station.
 
Optimization: using TCP to single stations avoiding multicast repetitions to groups of only one station.
  
=== Moved line volume from state to nvram (MTN-1401) ===
+
===Moved line volume from state to nvram (MTN-1401)===
 
Turbine line output volume is now configurable from AlphaPro 11.6.3.12.
 
Turbine line output volume is now configurable from AlphaPro 11.6.3.12.
  
=== Support Event Handler DAK as RCI for function keys in CRM mode ===
+
===Support Event Handler DAK as RCI for function keys in CRM mode===
 
The Navigation keys in CRM mode available on IP desktop/flush and DualDisplay are now reported to Event handler as DAK as RCI sub event 101->104.
 
The Navigation keys in CRM mode available on IP desktop/flush and DualDisplay are now reported to Event handler as DAK as RCI sub event 101->104.
  
=== Automatic Work Channel Implementation (MTN-1311) ===
+
===Automatic Work Channel Implementation (MTN-1311)===
 
The Mixer station control master can now be configured for using work channel. <br>
 
The Mixer station control master can now be configured for using work channel. <br>
 
Configurable from AlphaPro 11.6.3.12 <br>
 
Configurable from AlphaPro 11.6.3.12 <br>
  
= AMC 12.3.3.3K2 =
+
=AMC 12.3.3.3K2=
 
  Release: Official, available on request
 
  Release: Official, available on request
 
  NVRAM version 12.3
 
  NVRAM version 12.3
Line 131: Line 139:
  
 
==Errors Corrected==
 
==Errors Corrected==
=== Talkback Issue if Mixer Station Down During Multi Talkback (MTN-1384) ===
+
===Talkback Issue if Mixer Station Down During Multi Talkback (MTN-1384)===
 
If the Mixer Station is reset/offline while active in multi conference talkback, the Mixer Station is not able to do talkback after re-registration. <br>
 
If the Mixer Station is reset/offline while active in multi conference talkback, the Mixer Station is not able to do talkback after re-registration. <br>
 
Fixed with proper cleanup of talkback states during station down.
 
Fixed with proper cleanup of talkback states during station down.
  
=== Issues with the Automatic Static/Dynamic Multicast Port Allocation System (MTN-1387) ===
+
===Issues with the Automatic Static/Dynamic Multicast Port Allocation System (MTN-1387)===
 
For 12.3.3.1 software a backward compatibility algorithm was introduced to support older IP station software in systems without Mixer Stations. <br>  
 
For 12.3.3.1 software a backward compatibility algorithm was introduced to support older IP station software in systems without Mixer Stations. <br>  
 
In a Mixer Station system, if one or more SX-conferences was started before the first Mixer Station was registered, these first conferences would use static multicast UDP ports, inhibiting correct SX-conference mixing.  
 
In a Mixer Station system, if one or more SX-conferences was started before the first Mixer Station was registered, these first conferences would use static multicast UDP ports, inhibiting correct SX-conference mixing.  
Line 143: Line 151:
  
  
= AMC 12.3.3.3 =
+
=AMC 12.3.3.3=
 
  Release: Official, available on request
 
  Release: Official, available on request
 
  NVRAM version 12.3
 
  NVRAM version 12.3
Line 154: Line 162:
  
 
==Errors Corrected==
 
==Errors Corrected==
=== Reset due to Internal Message Handing (MTN-1305)===
+
===Reset due to Internal Message Handing (MTN-1305)===
 
Reset occurred due to multi thread variable usage, fixed.  
 
Reset occurred due to multi thread variable usage, fixed.  
  
Line 160: Line 168:
 
Several error messages observed during Mixer Station usage are corrected.
 
Several error messages observed during Mixer Station usage are corrected.
  
=== Memory Leak when Using the AlphaWeb Station Configuration Page (MTN-1319)===
+
===Memory Leak when Using the AlphaWeb Station Configuration Page (MTN-1319)===
 
Using AlphaWeb Station Configuration would create a memory leak for each apply, fixed.
 
Using AlphaWeb Station Configuration would create a memory leak for each apply, fixed.
  
=== IPHA Config Master Volume Settings was not Updated in Standby Controller (MTN-1324)===
+
===IPHA Config Master Volume Settings was not Updated in Standby Controller (MTN-1324)===
 
The configured volume setting received from Configuration Master was overwritten in the Standby Controller from local data with default values, fixed.
 
The configured volume setting received from Configuration Master was overwritten in the Standby Controller from local data with default values, fixed.
  
=== Clean & Factory IPHA License Removal (MTN-1333) ===
+
===Clean & Factory IPHA License Removal (MTN-1333)===
 
Related to systems used as IPHA, the takeover license inherted from Configuration Master was not removed using Clean & Factory reset.
 
Related to systems used as IPHA, the takeover license inherted from Configuration Master was not removed using Clean & Factory reset.
 
Resulting in a faulty "free license" behavior. Fixed.
 
Resulting in a faulty "free license" behavior. Fixed.
  
=== Restarting of Terminated Recording Sessions (MTN-1316) ===
+
===Restarting of Terminated Recording Sessions (MTN-1316)===
 
In a system running with recording of 24/7 SX-conferences, the recording sessions would not be restarted in case of network issues or Recorder PC reset. <br>
 
In a system running with recording of 24/7 SX-conferences, the recording sessions would not be restarted in case of network issues or Recorder PC reset. <br>
 
AlphaCom will now monitor and restart faulty recording session.
 
AlphaCom will now monitor and restart faulty recording session.
  
=== RE-INVITE During Recording Sessions Leak Memory. (MTN-1317) ===
+
===RE-INVITE During Recording Sessions Leak Memory. (MTN-1317)===
 
The Recorder PC uses RE-INVITE messages for monitoring of active recording sessions. Each RE-INVITE will allocate memory in SIPD. <br>
 
The Recorder PC uses RE-INVITE messages for monitoring of active recording sessions. Each RE-INVITE will allocate memory in SIPD. <br>
 
Memory will be free when Recording session ends. <br>
 
Memory will be free when Recording session ends. <br>
 
For system using 24/7 conference recording the session should ideal never end, thus result in overuse of memory over time. Fixed.
 
For system using 24/7 conference recording the session should ideal never end, thus result in overuse of memory over time. Fixed.
  
= AMC 12.3.3.2 =
+
<br>
 +
 
 +
=AMC 12.3.3.2=
 
  Release: Official, available on request
 
  Release: Official, available on request
 
  NVRAM version 12.3
 
  NVRAM version 12.3
Line 188: Line 198:
 
''AlphaSys 12.3.3.x software only runs on AMC 11 hardware and require use of STIC'' <br>
 
''AlphaSys 12.3.3.x software only runs on AMC 11 hardware and require use of STIC'' <br>
  
== Error Corrected ==
+
==Error Corrected==
=== Issues with SIP Stations using ODX Conferences in AlphaNet (MTN-1264) ===
+
===Issues with SIP Stations using ODX Conferences in AlphaNet (MTN-1264)===
 
SIP stations where rejected if doing remote ODX conferencing in AlphaNet. <br>
 
SIP stations where rejected if doing remote ODX conferencing in AlphaNet. <br>
  
=== Mixer station DAK LED Update after Registration (MTN-1240) ===
+
===Mixer station DAK LED Update after Registration (MTN-1240)===
 
The build in LED status of active conferences on the Mixer station is now updated correctly after station/system reset.
 
The build in LED status of active conferences on the Mixer station is now updated correctly after station/system reset.
  
=== $DISCON_STATUS_BC after Cancellation of Call to Busy Station (MTN-1260) ===
+
===$DISCON_STATUS_BC after Cancellation of Call to Busy Station (MTN-1260)===
 
System now generate a $DISCON_STATUS_BC for terminated busy calls.
 
System now generate a $DISCON_STATUS_BC for terminated busy calls.
  
=== Reconnect Recording (MTN-1284) ===
+
===Reconnect Recording (MTN-1284)===
 
If a ongoing Recording session is terminated due to network issues or reset of the Recording PC, the Recording session is re-established when the Recorder is reconnected to AlphaCom.
 
If a ongoing Recording session is terminated due to network issues or reset of the Recording PC, the Recording session is re-established when the Recorder is reconnected to AlphaCom.
  
Line 204: Line 214:
  
  
= AMC 12.3.3.1B =
+
=AMC 12.3.3.1B=
 
  Release: Official, available on request
 
  Release: Official, available on request
 
  NVRAM version 12.3
 
  NVRAM version 12.3
Line 213: Line 223:
 
''AlphaSys 12.3.3.x software only runs on AMC 11 hardware and require use of STIC'' <br>
 
''AlphaSys 12.3.3.x software only runs on AMC 11 hardware and require use of STIC'' <br>
  
== Error Corrected ==
+
==Error Corrected==
=== Recording of Multiple Simplex Conferences (MTN-1251) ===
+
===Recording of Multiple Simplex Conferences (MTN-1251)===
 
Issues with the recorded audio during multiple SX conferences. Compatibility issues with RTCP messages. <br>
 
Issues with the recorded audio during multiple SX conferences. Compatibility issues with RTCP messages. <br>
 
Distribution of RTCP messages removed from AlphaCom.
 
Distribution of RTCP messages removed from AlphaCom.
  
  
= AMC 12.3.3.1 =
+
=AMC 12.3.3.1=
 
  Release: Official, available on request
 
  Release: Official, available on request
 
  NVRAM version 12.3
 
  NVRAM version 12.3
Line 233: Line 243:
 
''This software is intended for InterCom usage. For Exigo software please use official releases for Exigo'' <br>
 
''This software is intended for InterCom usage. For Exigo software please use official releases for Exigo'' <br>
  
''The IP configuration in AlphaWeb is different from 11.2.x, see AlphaWiki for description: [[AlphaWeb#Interfaces | Interfaces ]]
+
''The IP configuration in AlphaWeb is different from 11.2.x, see AlphaWiki for description: [[AlphaWeb#Interfaces | Interfaces]]''
  
  
 
+
==Error Corrected==
== Error Corrected ==
+
===Queuing for Group Call Gong (MTN-1229)===
=== Queuing for Group Call Gong (MTN-1229) ===
 
 
Multiple simultaneous startup of group calls with gong could make the group call function hang.
 
Multiple simultaneous startup of group calls with gong could make the group call function hang.
  
=== Issues with group call to slave module, and when SX conf Speaker is in slave module (MTN-1204) ===
+
===Issues with group call to slave module, and when SX conf Speaker is in slave module (MTN-1204)===
 
Audio distribution was depending on an active link connection on <u>both</u> Eth ports, also unused/unconfigured Eth port had to be connected to a network switch.
 
Audio distribution was depending on an active link connection on <u>both</u> Eth ports, also unused/unconfigured Eth port had to be connected to a network switch.
  
=== Group Call with Max Missing Members (MTN-438) ===
+
===Group Call with Max Missing Members (MTN-438)===
 
Group call with configured max missing members > 0 did not start at all.
 
Group call with configured max missing members > 0 did not start at all.
  
=== Restrictions of C-key as Station Private Mode Switch (MTN-1192) ===
+
===Restrictions of C-key as Station Private Mode Switch (MTN-1192)===
 
The use of C-key-Private is now only allowed when privacy mode is set to "Station Controlled" in AlphaPro. That is the same behavior as analogue stations.<br>
 
The use of C-key-Private is now only allowed when privacy mode is set to "Station Controlled" in AlphaPro. That is the same behavior as analogue stations.<br>
 
Now the C-key is not working for private mode setting if station is configured with "Always private/open"
 
Now the C-key is not working for private mode setting if station is configured with "Always private/open"
  
===Not Possible to Delete or Add Custom Scripts in AlphaWeb (MTN-1096) ===
+
===Not Possible to Delete or Add Custom Scripts in AlphaWeb (MTN-1096)===
 
There was a permission problem of the files after the security upgrade of AlphaSys 11.5
 
There was a permission problem of the files after the security upgrade of AlphaSys 11.5
  
===Sending NUL Character from EDO (MTN-502) ===
+
===Sending NUL Character from EDO (MTN-502)===
 
Using hex encoding the character 0 was not possible to send on the EDO port. (All other values was okay)  
 
Using hex encoding the character 0 was not possible to send on the EDO port. (All other values was okay)  
 
  edo "\x00"
 
  edo "\x00"
  
  
=== Memory leak in SIPD (AL-2658)===
+
===Memory leak in SIPD (AL-2658)===
 
Memory leak in sipd found during Exigo testing. New SIPD 2.09.
 
Memory leak in sipd found during Exigo testing. New SIPD 2.09.
  
===Call Status Broadcast DP Messages Should Use Current User Directory Number (MTN-1069) ===
+
===Call Status Broadcast DP Messages Should Use Current User Directory Number (MTN-1069)===
 
In some customer systems several users are configured for a single physical station. <br>
 
In some customer systems several users are configured for a single physical station. <br>
 
Connected 3rd party equipment then needs information of user currently called. <br>
 
Connected 3rd party equipment then needs information of user currently called. <br>
Line 275: Line 284:
 
Temprature alarm will no longer be generated in the XE1 exchange.
 
Temprature alarm will no longer be generated in the XE1 exchange.
  
===Avoid Overlapping ATLB12 and IP Station Physical Numbers (MTN-610) ===
+
===Avoid Overlapping ATLB12 and IP Station Physical Numbers (MTN-610)===
 
IP stations will have priority above ATLB12 station if configured on same physical number. <br>  
 
IP stations will have priority above ATLB12 station if configured on same physical number. <br>  
 
Same functionality as for ASLT. Need reset to take effect.
 
Same functionality as for ASLT. Need reset to take effect.
Line 286: Line 295:
 
The $ST_STATE message now contain the active conference of the station.
 
The $ST_STATE message now contain the active conference of the station.
  
=== Clear All DAK LEDs on Turbine (MTN-1074)===
+
===Clear All DAK LEDs on Turbine (MTN-1074)===
 
The Event Handler IND command for clearing all DAK leds now also work for Turbine ECPIR with DAK module extentions.
 
The Event Handler IND command for clearing all DAK leds now also work for Turbine ECPIR with DAK module extentions.
  
  
== Enhancement SIP==
+
==Enhancement SIP==
  
=== Class Of Service Configuration for Each SIP Trunk ===
+
===Class Of Service Configuration for Each SIP Trunk===
 
It is now possible to configure COS for each SIP trunk.
 
It is now possible to configure COS for each SIP trunk.
 
   .module_profile.node_profile[1].class_serv  
 
   .module_profile.node_profile[1].class_serv  
 
If set to 0, default COS will be used (15 - Outside Telephones).
 
If set to 0, default COS will be used (15 - Outside Telephones).
  
=== SIP as members of Conference and Group Calls ===
+
===SIP as members of Conference and Group Calls===
*SIP stations and SIP trunk stations can be included in the group call groups.  
+
 
*SIP stations can be set as default member and speaker of a SX conference.  
+
*SIP stations and SIP trunk stations can be included in the group call groups.
 +
*SIP stations can be set as default member and speaker of a SX conference.
 +
 
 
  Limitation: SIP stations can not dial them self into conferences by dialing 8201.  
 
  Limitation: SIP stations can not dial them self into conferences by dialing 8201.  
 
             SIP stations can only be included using $DP messages or features codes for conference to group/station.
 
             SIP stations can only be included using $DP messages or features codes for conference to group/station.
 
             SIP stations can not signal M-key to talk. Must use $CONF_TALK.  
 
             SIP stations can not signal M-key to talk. Must use $CONF_TALK.  
 
             All members of the group call or SX conference must support the same Codec.
 
             All members of the group call or SX conference must support the same Codec.
 +
 
*New flag for Allowing SIP Stations as Members of Group Call and Default Speaker of SX Conference
 
*New flag for Allowing SIP Stations as Members of Group Call and Default Speaker of SX Conference
 +
 
  Flag is default false for backward compatibility. Available in latest AlphaPro 11.6.
 
  Flag is default false for backward compatibility. Available in latest AlphaPro 11.6.
 
  &000260 .ex_profile.flags.cp_allow_sip = 1 (0x00000001)
 
  &000260 .ex_profile.flags.cp_allow_sip = 1 (0x00000001)
  
===SIP Parsing "P-Asserted-Identity" in the 200 OK header (MTN-1182) ===
+
===SIP Parsing "P-Asserted-Identity" in the 200 OK header (MTN-1182)===
 
If a "P-Asserted-Identity" is found in the reply for an outgoing INVITE, the new user name and/or display name are updated in AlphaCom.
 
If a "P-Asserted-Identity" is found in the reply for an outgoing INVITE, the new user name and/or display name are updated in AlphaCom.
  
=== Option for Silence RTP Audio to SIP Stations ===
+
===Option for Silence RTP Audio to SIP Stations===
 
When SIP stations are used as default speakers in conference rtp audio can be sent back to the SIP station to avoid timeout on SIP side due to missing RTP audio.
 
When SIP stations are used as default speakers in conference rtp audio can be sent back to the SIP station to avoid timeout on SIP side due to missing RTP audio.
 
  52 &000256 .ex_profile.flags.cp_bp_silence = 1 (0x00000001)  
 
  52 &000256 .ex_profile.flags.cp_bp_silence = 1 (0x00000001)  
 
'''Node: This flag is intended for "IP only systems". If this flag is used for a system with ASLT/Multimodule there will be no audio distributed to analog stations.'''
 
'''Node: This flag is intended for "IP only systems". If this flag is used for a system with ASLT/Multimodule there will be no audio distributed to analog stations.'''
 
  
 
   
 
   
== Enhancement Mixer Station ==
+
==Enhancement Mixer Station==
=== Enhanced Turbine Mix Station Support ===
+
===Enhanced Turbine Mix Station Support===
 
Turbine Mixer stations was first introduced in the limited X-version software 11.2.3.12x. <br>
 
Turbine Mixer stations was first introduced in the limited X-version software 11.2.3.12x. <br>
 
Feature Description: [[Conference_Mixer_Station | Multi Conference]] <br>
 
Feature Description: [[Conference_Mixer_Station | Multi Conference]] <br>
  
=== Multi Talkback on M-key ===
+
===Multi Talkback on M-key===
 
The mixer station has default talkback to all active conferences in the mix when pressing M-key from idle.
 
The mixer station has default talkback to all active conferences in the mix when pressing M-key from idle.
 
  The functionality can be turned off using the station flag 3: NO_MULTI_TALKBACK
 
  The functionality can be turned off using the station flag 3: NO_MULTI_TALKBACK
 
The new feature 119, multi talkback can be used for DAK key activation.
 
The new feature 119, multi talkback can be used for DAK key activation.
  
=== Automatic DAK LED update of Mix Controller Station ===
+
===Automatic DAK LED update of Mix Controller Station===
 
Based on param3 of the Conference Feature 65, the LEDs of the DAK keys for toggling conferences in the Mixer can be auto updated without use of event handler.  
 
Based on param3 of the Conference Feature 65, the LEDs of the DAK keys for toggling conferences in the Mixer can be auto updated without use of event handler.  
 
   
 
   
=== Feature 116 "Volume and Control Mixer Station" ===
+
===Feature 116 "Volume and Control Mixer Station"===
 
[[Conference_Mixer_Station | Conference Mixer Station Feature 116 added to COS 1]]. <br>
 
[[Conference_Mixer_Station | Conference Mixer Station Feature 116 added to COS 1]]. <br>
 
Auto-loaded directory numbers: <br>
 
Auto-loaded directory numbers: <br>
Line 337: Line 349:
 
  8266 Toggle conference
 
  8266 Toggle conference
  
=== Feature 119 "Multi Talkback" ===
+
===Feature 119 "Multi Talkback"===
 
[[Multi_Talkback | Multi Talkback Feature 119 added to COS 1]] <br>
 
[[Multi_Talkback | Multi Talkback Feature 119 added to COS 1]] <br>
 
Auto-loaded directory number:
 
Auto-loaded directory number:
 
  8270 Multi talkback
 
  8270 Multi talkback
  
== Enhancement ==
+
==Enhancement==
=== Backward Compatible Multicast Group Call ===
+
===Backward Compatible Multicast Group Call===
 
Previous X-versions of 12.3 used dynamic allocation of udp ports for multicast group call. This is required for the new Mixer Station software. <br>
 
Previous X-versions of 12.3 used dynamic allocation of udp ports for multicast group call. This is required for the new Mixer Station software. <br>
 
Not all IP station software versions did support dynamic udp port usages thus audio was missing during group functions. <br>
 
Not all IP station software versions did support dynamic udp port usages thus audio was missing during group functions. <br>
 
Now the Multicast system of 11.5 and earlier versions are used as long as no Mixer Stations are configured in the system.
 
Now the Multicast system of 11.5 and earlier versions are used as long as no Mixer Stations are configured in the system.
  
=== Enhanced TST commands ===
+
===Enhanced TST commands===
 
  cnf - list all active conferences and its members.
 
  cnf - list all active conferences and its members.
 
  res - Now show IP resources currently in use. (CC 01 Count:32, Type:(01) PCMC, ''Used: 3'')
 
  res - Now show IP resources currently in use. (CC 01 Count:32, Type:(01) PCMC, ''Used: 3'')
  
=== Direct IP Audio Enhancements in "One to Many" Calls===
+
===Direct IP Audio Enhancements in "One to Many" Calls===
 
For previous versions of AlphaSys audio in SX conferences, Group Call and Program used the AlphaCom backplane for audio distribution.<br>
 
For previous versions of AlphaSys audio in SX conferences, Group Call and Program used the AlphaCom backplane for audio distribution.<br>
 
Use of the backplane require audio trans-coding, the result is additional audio delay, distortion and limitation of the frequency response. <br>
 
Use of the backplane require audio trans-coding, the result is additional audio delay, distortion and limitation of the frequency response. <br>
 
Optimizations done:
 
Optimizations done:
*Program, Group Call and SX conference with IP audio source, will distribute the received audio source to IP listeners without re-coding the audio.  
+
 
 +
*Program, Group Call and SX conference with IP audio source, will distribute the received audio source to IP listeners without re-coding the audio.
 
*Global SX conference and global group call will connect other nodes as IP audio into the group call or conference. Resulting in less delay and audio IP-resource usage.
 
*Global SX conference and global group call will connect other nodes as IP audio into the group call or conference. Resulting in less delay and audio IP-resource usage.
  
Line 365: Line 378:
 
Global SX conference and Group Call only supported G722, now the setup is based on the SX/VP Codec configuration of the Initiator node
 
Global SX conference and Group Call only supported G722, now the setup is based on the SX/VP Codec configuration of the Initiator node
  
=== Increased DAK string length from 19 to 25 bytes (MTN-1050)===
+
===Increased DAK string length from 19 to 25 bytes (MTN-1050)===
 
The length of the DAK string is increased from 19 to 25 bytes.
 
The length of the DAK string is increased from 19 to 25 bytes.
  
Line 371: Line 384:
 
VOIP parameters for the IP dummy station using the $VAC command is now permanently stored, thus will survive system reset.
 
VOIP parameters for the IP dummy station using the $VAC command is now permanently stored, thus will survive system reset.
  
=== Key Press and Release Option on $DIAL_DAK ===
+
===Key Press and Release Option on $DIAL_DAK===
 
It is now an optional parameter to specify DAK press or release of the $DIAL_DAK message. Default is press/release as the original $DIAL_DAK message.
 
It is now an optional parameter to specify DAK press or release of the $DIAL_DAK message. Default is press/release as the original $DIAL_DAK message.
  
=== Support of 10 OPC Servers ===
+
===Support of 10 OPC Servers===
 
Support of a total of 10 OPC servers in AlphaNet. Limited to two servers connected to each node.
 
Support of a total of 10 OPC servers in AlphaNet. Limited to two servers connected to each node.
  

Latest revision as of 04:52, 8 March 2022

Previous Release - AlphaCom 11.05 - Release Notes

Next Release - AlphaCom 12.05 - Release Notes

This document provides the release notes for AlphaCom 12.3 with incremental bug fix releases. The release notes for AlphaCom 12.3 describe new features, improvements and issues fixed after AlphaCom 11.5.

Software in production: AMC 11.2.3.10
Note: For each software version the NVRAM version is listed. If the NVRAM version is different, the AlphaCom InterCom configuration will get default configuration, and then you must do a SendAll from AlphaPro to restore the configuration.
All AlphaWeb configuration will be kept.



Contents

AMC 12.3.3.4 (LA)

Note icon INCA based Intercom stations with software version 01.x.x.x are not supported from this AMC version onwwards. Any such station must be upgraded to version 02.x.x.x


Release: Official, available on request
NVRAM version 12.3
Date: 29th of November 2017

System upgrade file: bsp_32.00.3.1_AS_12.3.3.4.apkg
This software includes both AlphaSys and BSP 32.0.3.1
AlphaSys 12.3.3.x software only runs on AMC 11 hardware and require use of STIC

Security Scan

bsp_32.00.3.1_AS_12.3.3.4.apkg with default configuration passed the Nessus Vulnerability Scanner run 30th of November 2017

Errors Corrected

SIP: Support of Early Media "183" (MTN 1371)

Incoming Calls from SIP to AlphaCom did not allow Early Media.

Camp on Busy (MTN-1373)

In some camp on busy situations a system reset could occur.

Silence Call not Working (MTN-1390, MTN-1395)

The 11.5 data commands and parameters for silence call setup using $CALL and $SC are now included in the 12.3. software.

Issues with two Simultaneous ReCall Features (MTN-1359)

Reset could occur if dispatching two simultaneous ReCall playbacks to groups, where groups contains overlapping members and AlphaPro missing members configuration > 0.

BUFFER_TOO_SMALL SysLog Messages (MTN-1405)

When reporting absence status a "Buffer too small" debug SysLog message could occur. Previously fixed for the 11.3 software.

Avoid Tone Test and Scheduler During IPHA Standby Mode (MTN-1256)

Both the scheduler and the Tone test was running in an AlphaCom in standby mode.
This would generate syslog messages.
Now Tone test and Event Scheduler is stopped during standby mode.

SIP Station Timeout on Collect Digit Features (MTN-1410)

SIP stations dialing feature 81 / 83 configured with collection of digits would timeout before the digits where sendt to the SIP trunk.

Remove Directory Numbers from the 614 Menu (MTN-1389)

These numbers are removed in Autoload: 54, 55, 0, 627, 628, 8270, 8260-8266.

Ringing Group from SIP Trunk (MTN-1331)

It was not possible to use ringing group calls from SIP in combination with node flag "180 Ringing".

Busy Override to Ringing Group (MTN-1357)

It was not possible to do busy override of stations already in call from answering Ringing Group call.

  • If A is answering Ringing Group call from B.
  • C is not able to do busy override to A.

Call Forwarding (71/72) Fails for SIP Station (MTN-1361)

When a SIP user calls a station (ASLT or IP) which is forwarded, the call will not connect with the target station.

Dummy Stations as Default SX Speaker (MTN-1353)

Sometimes the audio was not connected correctly from a Dummy IP talker of a SX conference.

Wrong Default CRM Notification Tone (MTN-1327)

CRM notification tone is now default setting.

Avoid Overlapping Talkback UDP Ports for Mixer Station (MTN-1350)

Sometimes there could be conflicting use of UDP ports for talkback and multicast SX conference streaming when using the Mixer Station.
This would result in no talkback audio from the Mixer Station.

Send Private Info to IP Station when Using Feature 80 (MTN-1348)

Using the feature 80 to control station private setting did not update the display of the INCA stations.

Missing B subscriber info, Voice Recording in Ring-Master Installations (MTN-1349)

When answering call request from remote nodes, remote stations number and name was not updated in the active proxy stations list.
This resulted in missing info for Recording sessions and event handler using %dir/%nam.

Configuration not Synchronized in IPHA (MTN-1332)

In situations where the configuration slave has ex_data waiting to be written to flash, the synchronization of configuration data from config master will trigger writing to flash thus overwrite new configuration.
Now configuration slave will inhibit writing local changes to flash file.

Issues with Ringing Group using SIP Station Members (MTN-1340)

Simultaneous answer from two SIP stations of a ringing group call could in some situations resulting in Exchange reset.

Issues with Answering Ringing Group Call Early (MTN-1023)

If a ringing group call was answered before first ringing sequence is finished, the ringing will continue at the other receivers of the call.

AlphaCom Ignores "486 Busy Here" from Connected SIP Phones (MTN-1425)

The "Reject" key of the ITSV-1 SIP station would not cancel the incoming call.

Enhancement

Configuration of Common Codec for all Program/GroupCall/Conference (MTN-1374)

A common setting of codec for "one to many" call types can be configured in NVRAM.

.ex_profile.glob_const.CP_codec = 0 

Default 0 = G722. Codecs values are the same as for codec selection of the $VAC command. Using AlphaPro configuration of individual codecs for Group/SX will overrule the common setting.

Added Configuration for Changing SIP Reply Codes (MTN-1331)

For SIP trunk station calls, AlphaCom will default reply with code 480 in fault situation and 182 for early media.

.ex_profile.ip_config.SIP_REP_480 = 0 (0x0000)
.ex_profile.ip_config.SIP_REP_182 = 0 (0x0000)

Configuring other SIP codes can be done by changing the parameters to legal SIP codes. (0 = use default)

Feature for Mixer Station WorkChannel Talkback from DAK keys

New feature 120, used for WorkChannel talkback from Mixer station.

Support of TFIE-6 Display Update(MTN-1370)

The new TFIE-6 requires a special display protocol to operate.

If only one member of the age multicast bitmap, then send age message as tcp DIP (MTN-1399)

Optimization: using TCP to single stations avoiding multicast repetitions to groups of only one station.

Moved line volume from state to nvram (MTN-1401)

Turbine line output volume is now configurable from AlphaPro 11.6.3.12.

Support Event Handler DAK as RCI for function keys in CRM mode

The Navigation keys in CRM mode available on IP desktop/flush and DualDisplay are now reported to Event handler as DAK as RCI sub event 101->104.

Automatic Work Channel Implementation (MTN-1311)

The Mixer station control master can now be configured for using work channel.
Configurable from AlphaPro 11.6.3.12

AMC 12.3.3.3K2

Release: Official, available on request
NVRAM version 12.3
Date: 6th of October 2017

System upgrade file: bsp_32.00.3.1_AS_12.3.3.3_K2.apkg
This software includes both AlphaSys and BSP 32.0.3.1
AlphaSys 12.3.3.x software only runs on AMC 11 hardware and require use of STIC


Errors Corrected

Talkback Issue if Mixer Station Down During Multi Talkback (MTN-1384)

If the Mixer Station is reset/offline while active in multi conference talkback, the Mixer Station is not able to do talkback after re-registration.
Fixed with proper cleanup of talkback states during station down.

Issues with the Automatic Static/Dynamic Multicast Port Allocation System (MTN-1387)

For 12.3.3.1 software a backward compatibility algorithm was introduced to support older IP station software in systems without Mixer Stations.
In a Mixer Station system, if one or more SX-conferences was started before the first Mixer Station was registered, these first conferences would use static multicast UDP ports, inhibiting correct SX-conference mixing. Fixed by checking all configured stations, not only registered station.

There is also corrected an issue related to overlapping use of Multicast UDP port and Mixer Station talkback UDP port, inhibiting talkback from Mixer Station for some combinations of SX mixing.


AMC 12.3.3.3

Release: Official, available on request
NVRAM version 12.3
Date: 8th of August 2017

System upgrade file: bsp_32.00.3.1_AS_12.3.3.3.apkg
This software includes both AlphaSys and BSP 32.0.3.1
AlphaSys 12.3.3.x software only runs on AMC 11 hardware and require use of STIC


Errors Corrected

Reset due to Internal Message Handing (MTN-1305)

Reset occurred due to multi thread variable usage, fixed.

SysLog Error Messages related to Mixer Station SX-Conferences Behavior (MTN-1320)

Several error messages observed during Mixer Station usage are corrected.

Memory Leak when Using the AlphaWeb Station Configuration Page (MTN-1319)

Using AlphaWeb Station Configuration would create a memory leak for each apply, fixed.

IPHA Config Master Volume Settings was not Updated in Standby Controller (MTN-1324)

The configured volume setting received from Configuration Master was overwritten in the Standby Controller from local data with default values, fixed.

Clean & Factory IPHA License Removal (MTN-1333)

Related to systems used as IPHA, the takeover license inherted from Configuration Master was not removed using Clean & Factory reset. Resulting in a faulty "free license" behavior. Fixed.

Restarting of Terminated Recording Sessions (MTN-1316)

In a system running with recording of 24/7 SX-conferences, the recording sessions would not be restarted in case of network issues or Recorder PC reset.
AlphaCom will now monitor and restart faulty recording session.

RE-INVITE During Recording Sessions Leak Memory. (MTN-1317)

The Recorder PC uses RE-INVITE messages for monitoring of active recording sessions. Each RE-INVITE will allocate memory in SIPD.
Memory will be free when Recording session ends.
For system using 24/7 conference recording the session should ideal never end, thus result in overuse of memory over time. Fixed.


AMC 12.3.3.2

Release: Official, available on request
NVRAM version 12.3
Date: 20th of June 2017

System upgrade file: bsp_32.00.3.1_AS_12.3.3.2.apkg
This software includes both AlphaSys and BSP 32.0.3.1
AlphaSys 12.3.3.x software only runs on AMC 11 hardware and require use of STIC

Error Corrected

Issues with SIP Stations using ODX Conferences in AlphaNet (MTN-1264)

SIP stations where rejected if doing remote ODX conferencing in AlphaNet.

Mixer station DAK LED Update after Registration (MTN-1240)

The build in LED status of active conferences on the Mixer station is now updated correctly after station/system reset.

$DISCON_STATUS_BC after Cancellation of Call to Busy Station (MTN-1260)

System now generate a $DISCON_STATUS_BC for terminated busy calls.

Reconnect Recording (MTN-1284)

If a ongoing Recording session is terminated due to network issues or reset of the Recording PC, the Recording session is re-established when the Recorder is reconnected to AlphaCom.



AMC 12.3.3.1B

Release: Official, available on request
NVRAM version 12.3
Date: 16th of May 2017

System upgrade file: bsp_32.00.3.1_AS_12.3.3.1B.apkg
This software includes both AlphaSys and BSP 32.0.3.1
AlphaSys 12.3.3.x software only runs on AMC 11 hardware and require use of STIC

Error Corrected

Recording of Multiple Simplex Conferences (MTN-1251)

Issues with the recorded audio during multiple SX conferences. Compatibility issues with RTCP messages.
Distribution of RTCP messages removed from AlphaCom.


AMC 12.3.3.1

Release: Official, available on request
NVRAM version 12.3
Date: 26th of April 2017

System upgrade file: bsp_32.00.3.1_AS_12.3.3.1.apkg
This software includes both AlphaSys and BSP 32.0.3.1
AlphaSys 12.3.3.x software only runs on AMC 11 hardware and require use of STIC

AlphaSys 12.3 includes software developed for Exigo and InterCom functionality from the 11.5 branch and functionality from the obsolete 11.3 branch.

Node: Dynamic group call from 11.3.3.3 is not included in 12.3  

This release note consist of bug fixes and enhancements compared to AlphaSys release note 11.5.3.4 and 11.3.3.3
This software is intended for InterCom usage. For Exigo software please use official releases for Exigo

The IP configuration in AlphaWeb is different from 11.2.x, see AlphaWiki for description: Interfaces


Error Corrected

Queuing for Group Call Gong (MTN-1229)

Multiple simultaneous startup of group calls with gong could make the group call function hang.

Issues with group call to slave module, and when SX conf Speaker is in slave module (MTN-1204)

Audio distribution was depending on an active link connection on both Eth ports, also unused/unconfigured Eth port had to be connected to a network switch.

Group Call with Max Missing Members (MTN-438)

Group call with configured max missing members > 0 did not start at all.

Restrictions of C-key as Station Private Mode Switch (MTN-1192)

The use of C-key-Private is now only allowed when privacy mode is set to "Station Controlled" in AlphaPro. That is the same behavior as analogue stations.
Now the C-key is not working for private mode setting if station is configured with "Always private/open"

Not Possible to Delete or Add Custom Scripts in AlphaWeb (MTN-1096)

There was a permission problem of the files after the security upgrade of AlphaSys 11.5

Sending NUL Character from EDO (MTN-502)

Using hex encoding the character 0 was not possible to send on the EDO port. (All other values was okay)

edo "\x00"


Memory leak in SIPD (AL-2658)

Memory leak in sipd found during Exigo testing. New SIPD 2.09.

Call Status Broadcast DP Messages Should Use Current User Directory Number (MTN-1069)

In some customer systems several users are configured for a single physical station.
Connected 3rd party equipment then needs information of user currently called.
The call status messages CALL_STATUS_BC, CONN_STATUS_BC and DISCON_STATUS_BC was using the default user directory number for the B-station of the call, now the current user is used.
For backward compatibility flag ".ex_profile.flags.AMC10_bc = 1 " revert to default user.

Incorrect OPC Handling for Incoming SIP Trunk Call to Ringing Group (MTN-847)

The original Q_ELEM_ADDED message contained the physical number of the SIP trunk proxy as mail sender.
Now the message contains the directory number of the SIP caller thus making it possible for OPC to show some readable information.

Avoid temprature alarm in XE1 (MTN-498)

Temprature alarm will no longer be generated in the XE1 exchange.

Avoid Overlapping ATLB12 and IP Station Physical Numbers (MTN-610)

IP stations will have priority above ATLB12 station if configured on same physical number.
Same functionality as for ASLT. Need reset to take effect.

CRM Backlight Update (MTN-654)

CRM backlight will now be turned on after restart.

Issues with Event and SysLog Reporting of Global SX Speaker (MTN-1065)

Speaker of global SX conferences was not reported correctly to SysLog. The $ST_STATE message now contain the active conference of the station.

Clear All DAK LEDs on Turbine (MTN-1074)

The Event Handler IND command for clearing all DAK leds now also work for Turbine ECPIR with DAK module extentions.


Enhancement SIP

Class Of Service Configuration for Each SIP Trunk

It is now possible to configure COS for each SIP trunk.

 .module_profile.node_profile[1].class_serv 

If set to 0, default COS will be used (15 - Outside Telephones).

SIP as members of Conference and Group Calls

  • SIP stations and SIP trunk stations can be included in the group call groups.
  • SIP stations can be set as default member and speaker of a SX conference.
Limitation: SIP stations can not dial them self into conferences by dialing 8201. 
            SIP stations can only be included using $DP messages or features codes for conference to group/station.
            SIP stations can not signal M-key to talk. Must use $CONF_TALK. 
            All members of the group call or SX conference must support the same Codec.
  • New flag for Allowing SIP Stations as Members of Group Call and Default Speaker of SX Conference
Flag is default false for backward compatibility. Available in latest AlphaPro 11.6.
&000260 .ex_profile.flags.cp_allow_sip = 1 (0x00000001)

SIP Parsing "P-Asserted-Identity" in the 200 OK header (MTN-1182)

If a "P-Asserted-Identity" is found in the reply for an outgoing INVITE, the new user name and/or display name are updated in AlphaCom.

Option for Silence RTP Audio to SIP Stations

When SIP stations are used as default speakers in conference rtp audio can be sent back to the SIP station to avoid timeout on SIP side due to missing RTP audio.

52 &000256 .ex_profile.flags.cp_bp_silence = 1 (0x00000001) 

Node: This flag is intended for "IP only systems". If this flag is used for a system with ASLT/Multimodule there will be no audio distributed to analog stations.


Enhancement Mixer Station

Enhanced Turbine Mix Station Support

Turbine Mixer stations was first introduced in the limited X-version software 11.2.3.12x.
Feature Description: Multi Conference

Multi Talkback on M-key

The mixer station has default talkback to all active conferences in the mix when pressing M-key from idle.

The functionality can be turned off using the station flag 3: NO_MULTI_TALKBACK

The new feature 119, multi talkback can be used for DAK key activation.

Automatic DAK LED update of Mix Controller Station

Based on param3 of the Conference Feature 65, the LEDs of the DAK keys for toggling conferences in the Mixer can be auto updated without use of event handler.

Feature 116 "Volume and Control Mixer Station"

Conference Mixer Station Feature 116 added to COS 1.
Auto-loaded directory numbers:

8260-8265 used for individual channel, line and mixer volume control
8266 Toggle conference

Feature 119 "Multi Talkback"

Multi Talkback Feature 119 added to COS 1
Auto-loaded directory number:

8270 Multi talkback

Enhancement

Backward Compatible Multicast Group Call

Previous X-versions of 12.3 used dynamic allocation of udp ports for multicast group call. This is required for the new Mixer Station software.
Not all IP station software versions did support dynamic udp port usages thus audio was missing during group functions.
Now the Multicast system of 11.5 and earlier versions are used as long as no Mixer Stations are configured in the system.

Enhanced TST commands

cnf - list all active conferences and its members.
res - Now show IP resources currently in use. (CC 01 Count:32, Type:(01) PCMC, Used: 3)

Direct IP Audio Enhancements in "One to Many" Calls

For previous versions of AlphaSys audio in SX conferences, Group Call and Program used the AlphaCom backplane for audio distribution.
Use of the backplane require audio trans-coding, the result is additional audio delay, distortion and limitation of the frequency response.
Optimizations done:

  • Program, Group Call and SX conference with IP audio source, will distribute the received audio source to IP listeners without re-coding the audio.
  • Global SX conference and global group call will connect other nodes as IP audio into the group call or conference. Resulting in less delay and audio IP-resource usage.

Audio is still copied to the back plane for inclusion of ASLT/AGA types of stations in calls.
No optimization is done when MultiModule is used.

Codec use in Global Conference and Group Call (MTN-1148)

Global SX conference and Group Call only supported G722, now the setup is based on the SX/VP Codec configuration of the Initiator node

Increased DAK string length from 19 to 25 bytes (MTN-1050)

The length of the DAK string is increased from 19 to 25 bytes.

Store $VAC Config in Flash

VOIP parameters for the IP dummy station using the $VAC command is now permanently stored, thus will survive system reset.

Key Press and Release Option on $DIAL_DAK

It is now an optional parameter to specify DAK press or release of the $DIAL_DAK message. Default is press/release as the original $DIAL_DAK message.

Support of 10 OPC Servers

Support of a total of 10 OPC servers in AlphaNet. Limited to two servers connected to each node.

Configuration of Syslog info (MNT-676)

Added configuration of: SysLocation, SysContact and SysName setting. Configuration found under AlphaWeb->Configuration->Logging->SysLog Info.

Feature 118 "Station as Program Source"

New feature enabling an IP station to dial a feature code to become audio source of a program. Will only work if the program is without existing program source.