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Difference between revisions of "AlphaCom 10.xx - Release Notes"

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(New DP command $VOL)
 
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[[Category: Release notes]]
 
[[Category: Release notes]]
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Previous Release - [[AlphaCom 9.xx - Release Notes]]
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Next Release - [[AlphaCom 10.4x - Release Notes]]
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This document provides the release notes for AlphaCom 10 with incremental bug fix releases. The release notes for AlphaCom 10 describe new features, improvements and issues fixed after AlphaCom 9.
 +
 
'''Software in production:''' AMC 10.20<br>
 
'''Software in production:''' AMC 10.20<br>
 
'''Software released date:''' 2007-04-12<br>
 
'''Software released date:''' 2007-04-12<br>
Line 23: Line 29:
  
  
== Errors Corrected ==
+
==Errors Corrected==
=== Issue 3213 Reset when 7820 backup ===
+
===Issue 3213 Reset when 7820 backup===
 
Backup could crash due to overuse of system resources. The backup/restore function is optimized.
 
Backup could crash due to overuse of system resources. The backup/restore function is optimized.
=== Issue 3274 and 3320 IP Program distribution and mixed Multi Module with E7/E20/E26 ===
+
===Issue 3274 and 3320 IP Program distribution and mixed Multi Module with E7/E20/E26===
 
Combination of E7 / E20 / E26 program distribution will only work with prog.dist over IP.<br>
 
Combination of E7 / E20 / E26 program distribution will only work with prog.dist over IP.<br>
 
Remember to open the RTP voip ports in AlphaWeb IP filters !
 
Remember to open the RTP voip ports in AlphaWeb IP filters !
=== Issue 3278 Missing first digit in display when node number is 15 ===
+
===Issue 3278 Missing first digit in display when node number is 15===
 
Low level function bug related to node numbers ending in 0x0F (HEX), 15,31,47, 63 etc.  
 
Low level function bug related to node numbers ending in 0x0F (HEX), 15,31,47, 63 etc.  
=== Issue 3324 $ALRM to IP station ===
+
===Issue 3324 $ALRM to IP station===
 
Assert when disconnecting $ALRM to IP stations.
 
Assert when disconnecting $ALRM to IP stations.
=== Issue 3338 Time setting "785" and AlphaPro ===
+
===Issue 3338 Time setting "785" and AlphaPro===
 
Local time setting is now changing local time and not UTC time.
 
Local time setting is now changing local time and not UTC time.
=== Issue 2758 Serial communication via Eth ===
+
===Issue 2758 Serial communication via Eth===
 
Using 485 multidrop protocol can cause communication errors both on serial adapters and when connected directly to the AMC serial port.
 
Using 485 multidrop protocol can cause communication errors both on serial adapters and when connected directly to the AMC serial port.
  
== Functional changes / Enhancement ==
+
==Functional changes / Enhancement==
=== Treble preemphasis adustment of all IP audio ===
+
===Treble preemphasis adustment of all IP audio===
 
IP audio connections in AlphaNet are not compatible with earlier versions of AMCD SW. This is because  treble preemphasis of AlphaCom audio is now removed when converted to IP/RTP. This change is done in order to improve audio compatibility with IP stations and third party devices. When calling between AlphaCom with old and new SW, audio is connected, but the treble will be too low in one direction, and too high i the opposite direction. Beside not sounding right, the voice switched duplex failes to work properly, requiring extensive use of M-key.
 
IP audio connections in AlphaNet are not compatible with earlier versions of AMCD SW. This is because  treble preemphasis of AlphaCom audio is now removed when converted to IP/RTP. This change is done in order to improve audio compatibility with IP stations and third party devices. When calling between AlphaCom with old and new SW, audio is connected, but the treble will be too low in one direction, and too high i the opposite direction. Beside not sounding right, the voice switched duplex failes to work properly, requiring extensive use of M-key.
  
=== External VoIP audio gain adjustment ===
+
===External VoIP audio gain adjustment===
 
[[IP audio Gain adjustment]] options. Gain towards SIP devices is now by default +18db, giving more appropriate audio level towards external exquipment.
 
[[IP audio Gain adjustment]] options. Gain towards SIP devices is now by default +18db, giving more appropriate audio level towards external exquipment.
=== Multicast reachability test ===
+
===Multicast reachability test===
* IP station multicast reachability tests (mping), with reporting to syslog.  
+
 
* IP station not receiving RTP audio after connect, error forwarded to syslog.
+
*IP station multicast reachability tests (mping), with reporting to syslog.
=== Fader support in IP only system ===
+
*IP station not receiving RTP audio after connect, error forwarded to syslog.
 +
 
 +
===Fader support in IP only system===
 
Use AMC-IP FPGA for [[Fader Resource]]. Allowing Groupcall and Simplex conference in "IP only" system.
 
Use AMC-IP FPGA for [[Fader Resource]]. Allowing Groupcall and Simplex conference in "IP only" system.
=== Extended SIP-DTMP signaling ===
+
===Extended SIP-DTMP signaling===
* Outgoing call to SIP: DAK 2 - 5 sends DTMF signals A - D.
+
 
=== Pager support for 8 Bit Ascom ESPA protocol ===
+
*Outgoing call to SIP: DAK 2 - 5 sends DTMF signals A - D.
 +
 
 +
===Pager support for 8 Bit Ascom ESPA protocol===
 
Support Norwegian character to paging. [[ASCOM_Norwegian_Character_Mapping | Ascom Norwegian Characters]]
 
Support Norwegian character to paging. [[ASCOM_Norwegian_Character_Mapping | Ascom Norwegian Characters]]
=== OPC server license support ===
+
===OPC server license support===
 
License for connection of OPC server supported.
 
License for connection of OPC server supported.
=== Global Conference Lockup Situations ===
+
===Global Conference Lockup Situations===
 
Bug fix related to global conference, node reset, default speaker conflicts and resource lockup.
 
Bug fix related to global conference, node reset, default speaker conflicts and resource lockup.
=== DECT station lockup ===
+
===DECT station lockup===
 
Robust handling of call termination to avoid lockup of ATLB stations.
 
Robust handling of call termination to avoid lockup of ATLB stations.
=== Improved update of Watchdog ===
+
===Improved update of Watchdog===
 
Reset due to false watchdog timeout fixed.
 
Reset due to false watchdog timeout fixed.
  
Line 76: Line 86:
  
  
== Errors Corrected ==
+
==Errors Corrected==
=== Issue 3052: $TPROG causes AMC reset: ===
+
===Issue 3052: $TPROG causes AMC reset:===
 
$TPROG L%1.d U1 is used in the standard PrisCom database when the key returns to "0" position. Fixed(X10.23(1127))<br />
 
$TPROG L%1.d U1 is used in the standard PrisCom database when the key returns to "0" position. Fixed(X10.23(1127))<br />
=== Issue 3108: SIP trunk, short number not possible: ===
+
===Issue 3108: SIP trunk, short number not possible:===
 
[[Extended_Short_Number_Usage|Short number usage]]<br>
 
[[Extended_Short_Number_Usage|Short number usage]]<br>
 
With PNCI you can use the "Phone" function to define 3- or 4 -digit short numbers for easy access to the most frequently used phone numbers. You could also do 71 transfer to short numbers. A short number should show up in the directory list of the display.<br> Short number system for SIP is now implemented.(X10.23(1127))
 
With PNCI you can use the "Phone" function to define 3- or 4 -digit short numbers for easy access to the most frequently used phone numbers. You could also do 71 transfer to short numbers. A short number should show up in the directory list of the display.<br> Short number system for SIP is now implemented.(X10.23(1127))
 
(AlphaPro is currently (1028) not supporting the configuration)
 
(AlphaPro is currently (1028) not supporting the configuration)
  
=== Issue 3201 SIP stations currently used ===
+
===Issue 3201 SIP stations currently used===
 
When configuring new SIP stations the license use is now updated without the need of a reset.
 
When configuring new SIP stations the license use is now updated without the need of a reset.
=== Issue 3206: Serial Communication Problems (1021) ===
+
===Issue 3206: Serial Communication Problems (1021)===
 
Reset of AlphaCom in relations to Serial Port use. Serial port buffer made better.(X10.23(1127))
 
Reset of AlphaCom in relations to Serial Port use. Serial port buffer made better.(X10.23(1127))
=== Issue 3235: Audio Event In AlphaNet ===
+
===Issue 3235: Audio Event In AlphaNet===
 
When calling a station in a different node, event type audio and always programmed in the called node is not triggered.
 
When calling a station in a different node, event type audio and always programmed in the called node is not triggered.
 
This was due to the move of duplex from end node to start node. The default behavior of IP AlphaNet is now back to end node duplex.(X10.23(1127))
 
This was due to the move of duplex from end node to start node. The default behavior of IP AlphaNet is now back to end node duplex.(X10.23(1127))
=== Issue 3242: SIP RE-INVITE ===
+
===Issue 3242: SIP RE-INVITE===
 
SIP implementation now does support RE-INVITE(X10.23(1127))
 
SIP implementation now does support RE-INVITE(X10.23(1127))
=== Issue 3269 SIP rtp audio on different UDP ports ===
+
===Issue 3269 SIP rtp audio on different UDP ports===
 
Some SIP equipment sends RTP packets from different UDP port than the one they are receiving on. This is now supported by AlphaCom.
 
Some SIP equipment sends RTP packets from different UDP port than the one they are receiving on. This is now supported by AlphaCom.
=== Issue 3270 SIP Use CANCEL, not BYE ===
+
===Issue 3270 SIP Use CANCEL, not BYE===
 
After a ringing timeout the AlphaCom terminates the call with SIP command ‘BYE’  instead ‘CANCEL’, fixed.
 
After a ringing timeout the AlphaCom terminates the call with SIP command ‘BYE’  instead ‘CANCEL’, fixed.
=== Issue 3277 CR notification is missing/changed ===
+
===Issue 3277 CR notification is missing/changed===
 
Some Issues with Call request notification after ringing group implementation fixed.
 
Some Issues with Call request notification after ringing group implementation fixed.
=== Issue 3279 Event Handler discon group call ===
+
===Issue 3279 Event Handler discon group call===
 
If group call is disconnected in event handler "8 Conversation Outgoing" related to the same group call in some situations the AlphaCom asserts. This is fixed.
 
If group call is disconnected in event handler "8 Conversation Outgoing" related to the same group call in some situations the AlphaCom asserts. This is fixed.
=== Issue 3280 SIP registrar number starting with "0"===
+
===Issue 3280 SIP registrar number starting with "0"===
 
Fix of SIP reqistrar problem when numbers starting with Zero.
 
Fix of SIP reqistrar problem when numbers starting with Zero.
=== Issue 3281 ESPA444 stops ===
+
===Issue 3281 ESPA444 stops===
 
Protocol issues with the ESPA444 can get the pager protocol in a dead locked state. Fixed.
 
Protocol issues with the ESPA444 can get the pager protocol in a dead locked state. Fixed.
=== Issue 3283 Syslog report of "tst" warnings missing.===
+
===Issue 3283 Syslog report of "tst" warnings missing.===
 
System Warning is now also reported to Syslog,  
 
System Warning is now also reported to Syslog,  
=== Issue 3284 Ringing Grp from handset lifted station ===
+
===Issue 3284 Ringing Grp from handset lifted station===
 
A call to a ringing group from a station with lifted handset will lockup exchange on acceptance. Exchange needs to be reset, fixed.
 
A call to a ringing group from a station with lifted handset will lockup exchange on acceptance. Exchange needs to be reset, fixed.
=== Issue 3285 Ringing group call back ===
+
===Issue 3285 Ringing group call back===
 
The M-key needs to be pressed on the calling station if the calling station is in private.    Fixed.
 
The M-key needs to be pressed on the calling station if the calling station is in private.    Fixed.
=== Issue 3301 Crash when Call Req to IP station ===
+
===Issue 3301 Crash when Call Req to IP station===
 
Call request to IP stations will generate a reset of the exchange, fixed.
 
Call request to IP stations will generate a reset of the exchange, fixed.
=== Issue 3302 Event 10: %2.dir missing when SIP===
+
===Issue 3302 Event 10: %2.dir missing when SIP===
 
Mail events related to ringing group fixed.
 
Mail events related to ringing group fixed.
=== Issue 3316 Global grp call w IP station - reset ===
+
===Issue 3316 Global grp call w IP station - reset===
 
Reset problems related to IP station group calls fixed.
 
Reset problems related to IP station group calls fixed.
=== IP DECT hanging when canceling from AlphaCom ===
+
===IP DECT hanging when canceling from AlphaCom===
 
Problems of canceling early in audio setup from AlphaCom to Asacom IP DECT is solved
 
Problems of canceling early in audio setup from AlphaCom to Asacom IP DECT is solved
=== Search to Call Request ===
+
===Search to Call Request===
 
Fixes of preforming a Call Request in a search string when caller is located in a remote exchange.
 
Fixes of preforming a Call Request in a search string when caller is located in a remote exchange.
=== Full E26 rack problems ===
+
===Full E26 rack problems===
 
Add workaround of faulty read of own board type when full E26 rack.  
 
Add workaround of faulty read of own board type when full E26 rack.  
=== Pocket paging faults ===
+
===Pocket paging faults===
 
Use of Alarm priority for pocket paging could reset the exchange. <br>
 
Use of Alarm priority for pocket paging could reset the exchange. <br>
 
Bleep priority 0 is now legal to use.
 
Bleep priority 0 is now legal to use.
 
'''RedBoot Fix'''
 
'''RedBoot Fix'''
=== Issue 3268: Serial data in/out during boot ===
+
===Issue 3268: Serial data in/out during boot===
 
The serial data transmitted during reset is removed.
 
The serial data transmitted during reset is removed.
  
== Functional changes / Enhancement ==
+
==Functional changes / Enhancement==
=== Hostname support ===  
+
===Hostname support===  
 
Hostname can be stored for SIP destinations instead of ip address.
 
Hostname can be stored for SIP destinations instead of ip address.
=== Mutual exclusion groups increased ===
+
===Mutual exclusion groups increased===
 
Mutual exclusion groups increased from 4 to 8
 
Mutual exclusion groups increased from 4 to 8
=== Number of groupcalls increased ===
+
===Number of groupcalls increased===
 
The number of groupcalls is increased from 100 to 250
 
The number of groupcalls is increased from 100 to 250
 
(No AlphaPro support 1028)
 
(No AlphaPro support 1028)
  
=== Local Echo Canceling ===
+
===Local Echo Canceling===
 
[[Line_Echo_Cancellation|LEC]]<br />
 
[[Line_Echo_Cancellation|LEC]]<br />
 
10 channels of local echo canceling are available for "open" handset conversation when using SIP, Multimodule IP and AlphaNetIP.<br>
 
10 channels of local echo canceling are available for "open" handset conversation when using SIP, Multimodule IP and AlphaNetIP.<br>
 
The channels are allocated when needed, when no more resourced switched duplex is used.<br>
 
The channels are allocated when needed, when no more resourced switched duplex is used.<br>
=== Faster synchronizing of database to flash ===
+
===Faster synchronizing of database to flash===
 
The speed of synchronizing the configuration database to flash after "SendAll", dak programming etc is increased.
 
The speed of synchronizing the configuration database to flash after "SendAll", dak programming etc is increased.
=== Pocket paging "meet me" during conversation ===
+
===Pocket paging "meet me" during conversation===
 
Feature 25, Pocket paging meet me, alarm priority added to the autoload as dirno "48".
 
Feature 25, Pocket paging meet me, alarm priority added to the autoload as dirno "48".
=== Outgoing conversation related to SIP ===
+
===Outgoing conversation related to SIP===
 
[[Private_Ringing_Outgoing%28Event_Type%29|Private Ringing Outgoing]]<br>
 
[[Private_Ringing_Outgoing%28Event_Type%29|Private Ringing Outgoing]]<br>
 
Event type 33, used for state of outgoing conversation to SIP.
 
Event type 33, used for state of outgoing conversation to SIP.
  
=== Event handler %1.pag ===
+
===Event handler %1.pag===
 
New event handler operator %x.pag (x = 1 or 2) added for getting pager related to a station.
 
New event handler operator %x.pag (x = 1 or 2) added for getting pager related to a station.
=== New DP command $VOL ===
+
===New DP command $VOL===
 
[[VOLUME|VOL]] <br>
 
[[VOLUME|VOL]] <br>
 
New DP command $VOL, adjust station volume 0-9.
 
New DP command $VOL, adjust station volume 0-9.
  
=== Restriction of syslog reporting ===
+
===Restriction of syslog reporting===
 
Syslog system and debug reports are restricted to 60 messages each hour to prevent overload because of faulty configuration or other active syslogging problems. <br>
 
Syslog system and debug reports are restricted to 60 messages each hour to prevent overload because of faulty configuration or other active syslogging problems. <br>
 
The limit can be adjusted from TST:
 
The limit can be adjusted from TST:
 
  TST>>nvram - ex_profile.glob_const.syslog_block = 6 (6 = 60 messages pr hour)
 
  TST>>nvram - ex_profile.glob_const.syslog_block = 6 (6 = 60 messages pr hour)
=== Enhancement Issue 3266 Selective dialing ===
+
===Enhancement Issue 3266 Selective dialing===
 
[[Extended_Short_Number_Usage|Short number usage]]<br>
 
[[Extended_Short_Number_Usage|Short number usage]]<br>
 
When calling in to the AlphaCom via SIP gateway, it is be possible to automatically connect to an AlphaCom feature which makes it possible to do selective dialing. If no dialing within a preset time, the call is forwarded to a predefined number. The short number configuration system is used (Currently not available in AlphaPro 1028)
 
When calling in to the AlphaCom via SIP gateway, it is be possible to automatically connect to an AlphaCom feature which makes it possible to do selective dialing. If no dialing within a preset time, the call is forwarded to a predefined number. The short number configuration system is used (Currently not available in AlphaPro 1028)
  
=== Enhancement Issue 3272 SIP - DAK8, OnHold and Transfer ===
+
===Enhancement Issue 3272 SIP - DAK8, OnHold and Transfer===
 
[[Transparent_Mode_DAK-Key_Configuration|Transparent mode DAK configuration]]<br>
 
[[Transparent_Mode_DAK-Key_Configuration|Transparent mode DAK configuration]]<br>
 
Key used for activating transparent modus during SIP calls can now be change in NVRAM. <br>
 
Key used for activating transparent modus during SIP calls can now be change in NVRAM. <br>
 
  TST>>nvram - ex_profile.glob_const.trans_mode_dak = 8 (Default key is 8.)
 
  TST>>nvram - ex_profile.glob_const.trans_mode_dak = 8 (Default key is 8.)
  
=== Enhancement Issue 3275 Group call without M-key ===
+
===Enhancement Issue 3275 Group call without M-key===
 
[[Automatic_M-key_In_Group_Call|Automatic M-key in group call]]<br>
 
[[Automatic_M-key_In_Group_Call|Automatic M-key in group call]]<br>
 
A group call can be configured to work as automatic "M-key" when initiated
 
A group call can be configured to work as automatic "M-key" when initiated
 
  TST>>nvram -  ex_profile.group[G].flags = F
 
  TST>>nvram -  ex_profile.group[G].flags = F
 
G
 
G
 +
 
*group number 1 > 250
 
*group number 1 > 250
 +
 
F
 
F
*0 = default (press M key)  
+
 
 +
*0 = default (press M key)
 
*1 = automatic M-key from all stations
 
*1 = automatic M-key from all stations
 
*2 = automatic M-key only from SIP calls.
 
*2 = automatic M-key only from SIP calls.
  
=== Enhancement Issue 3304 Increase AlphaNet data links ===
+
===Enhancement Issue 3304 Increase AlphaNet data links===
 
The number of AlphaNet data links is increased from 20 to 50.
 
The number of AlphaNet data links is increased from 20 to 50.
  
Line 196: Line 209:
 
alpha_sys_10_00.tbz2 must be installed.<br><br>
 
alpha_sys_10_00.tbz2 must be installed.<br><br>
  
== Errors Corrected ==
+
==Errors Corrected==
  
=== Issue 3149: Follow Me(72) and User/Phy number ===
+
===Issue 3149: Follow Me(72) and User/Phy number===
 
There was a mix of user and physical number in the follow me function. Stations with different physical and user number would  not have correct follow me behavior.
 
There was a mix of user and physical number in the follow me function. Stations with different physical and user number would  not have correct follow me behavior.
=== Issue 3157: Call Request Transfer to Phone ===
+
===Issue 3157: Call Request Transfer to Phone===
 
Call request forwarding is solved by the new feature [[Call_Request_forward_by_search_strings|Call request forwarding by search strings]]  
 
Call request forwarding is solved by the new feature [[Call_Request_forward_by_search_strings|Call request forwarding by search strings]]  
=== Issue 3210: AlphaNet- CRM4 - display call request ===
+
===Issue 3210: AlphaNet- CRM4 - display call request===
 
The calling party’s number is not displayed when answering the call request (or MST mail) from a CRM IV type station in another node (AlphaNet). Only the name is shown. The error is only when the answering station has the Station Type set to "CRM 3&4" in software. Fixed.
 
The calling party’s number is not displayed when answering the call request (or MST mail) from a CRM IV type station in another node (AlphaNet). Only the name is shown. The error is only when the answering station has the Station Type set to "CRM 3&4" in software. Fixed.
=== Issue 3214: $CANM with source as global number ===
+
===Issue 3214: $CANM with source as global number===
 
The $CANM can now use global number as source without adding the specific node number (use L2345 in stead of L(2)2345)
 
The $CANM can now use global number as source without adding the specific node number (use L2345 in stead of L(2)2345)
=== Issue 3225: IP substation: Call Req Mode - LED ===
+
===Issue 3225: IP substation: Call Req Mode - LED===
 
If call requester mode is removed from an IP station the Audio to the IP station is now terminated.
 
If call requester mode is removed from an IP station the Audio to the IP station is now terminated.
=== Issue 3229: Formating of EXP not OK ===
+
===Issue 3229: Formating of EXP not OK===
 
Formating of %1.EXP is not working properly
 
Formating of %1.EXP is not working properly
=== Issue 3231: Statistic Log on IP station ===
+
===Issue 3231: Statistic Log on IP station===
 
Calls to IP stations in now logged in the SysLog statistics
 
Calls to IP stations in now logged in the SysLog statistics
=== Issue 3233: No Line error if Hot-line enabled ===
+
===Issue 3233: No Line error if Hot-line enabled===
 
The flag ''ignore_st_down_in_conv'' allow stations to fail during conversation. For hot line station it means that if hot line is trigged during station down it will not be reported. Fixed by setting a short timeout on how long station can be down when active
 
The flag ''ignore_st_down_in_conv'' allow stations to fail during conversation. For hot line station it means that if hot line is trigged during station down it will not be reported. Fixed by setting a short timeout on how long station can be down when active
=== Issue 3238: Hanging audio links to SIP/RMD ===
+
===Issue 3238: Hanging audio links to SIP/RMD===
 
When initiating a call to SIP/RMD from AlphaCom and terminating during setup tone the Audio connection is not disconnected. Fixed.
 
When initiating a call to SIP/RMD from AlphaCom and terminating during setup tone the Audio connection is not disconnected. Fixed.
=== Issue 3240:Ringing group - Loose CR mode when called ===
+
===Issue 3240:Ringing group - Loose CR mode when called===
 
When in call requester mode, you can be called by anybody, which will take you out of call requester mode.  It will be confusing if you expect your call to be answered and all of a sudden somebody else gets through. There should be a mechanism that when in call requester mode you can only be called by a station which has you in the queue. This is fixed by setting the caller in busy state and use automatic call override when answering from a ringing group.
 
When in call requester mode, you can be called by anybody, which will take you out of call requester mode.  It will be confusing if you expect your call to be answered and all of a sudden somebody else gets through. There should be a mechanism that when in call requester mode you can only be called by a station which has you in the queue. This is fixed by setting the caller in busy state and use automatic call override when answering from a ringing group.
=== Issue 3247/3252: AlphaCom uses duplex algorithm when conversation with Ring Master ===
+
===Issue 3247/3252: AlphaCom uses duplex algorithm when conversation with Ring Master===
 
The duplex algorithm is no longer used on AlphaCom side when calling/called by Ring Master.
 
The duplex algorithm is no longer used on AlphaCom side when calling/called by Ring Master.
=== SysLog ===
+
===SysLog===
 
In some cases syslog reporting from AlphaCom was not correctly initiated.
 
In some cases syslog reporting from AlphaCom was not correctly initiated.
  
 
==Functional changes / Enhancement==
 
==Functional changes / Enhancement==
=== DTMF tones during conversation ===
+
===DTMF tones during conversation===
 
Feature for dialing DTMF tones during conversation [[DTMF_During_Connection|DTMF feature]] (107) works for local conversation or towards SIP/RMD. Can Also be used as Data Message [[DTMF_CONN]]
 
Feature for dialing DTMF tones during conversation [[DTMF_During_Connection|DTMF feature]] (107) works for local conversation or towards SIP/RMD. Can Also be used as Data Message [[DTMF_CONN]]
=== Event trigger during conversation ===
+
===Event trigger during conversation===
 
New feature [[Event_During_Connection|Event Trigger During Connection]] (108).
 
New feature [[Event_During_Connection|Event Trigger During Connection]] (108).
=== Parallel Ringing ===
+
===Parallel Ringing===
 
Add stations in [[Parallel_Ringing_feature|Parallel Ringing]] when station is in private ringing mode  
 
Add stations in [[Parallel_Ringing_feature|Parallel Ringing]] when station is in private ringing mode  
=== Ringing Group ===
+
===Ringing Group===
 
New behavior of call request. Private ringing tone on receiving station(s). [[Ringing_Group_Feature| Ringing Group]] (109)<br>
 
New behavior of call request. Private ringing tone on receiving station(s). [[Ringing_Group_Feature| Ringing Group]] (109)<br>
 +
 
*[[SEND_MAIL]] added new flag for "auto delete" functionality and optional ringing tone in use with ringing group.
 
*[[SEND_MAIL]] added new flag for "auto delete" functionality and optional ringing tone in use with ringing group.
 
*[[CANCEL_MAIL]] allowed LV in destination field
 
*[[CANCEL_MAIL]] allowed LV in destination field
=== Global Conference ===
+
 
 +
===Global Conference===
 
New fixes and adjustments  
 
New fixes and adjustments  
* Regular interval Root of conference broadcast message was not sent in all situation. (A safety message for reconnecting if links have been down etc)
+
 
* The interval of Root of conference broadcast message is now decreased from 1 hour to 1 minute. It means reconnecting of missing nodes and cleaning of unused links will be much faster after certain network/reset problems.
+
*Regular interval Root of conference broadcast message was not sent in all situation. (A safety message for reconnecting if links have been down etc)
* Useless single AlphaNet audio links to root nodes with no conference members are cleared.  
+
*The interval of Root of conference broadcast message is now decreased from 1 hour to 1 minute. It means reconnecting of missing nodes and cleaning of unused links will be much faster after certain network/reset problems.
* When Root of conference is AMC 8/9 and there is an IP transit node with members of the conference, leaf IP nodes was not  able to connect to the conference due to CODEC selection failure.  
+
*Useless single AlphaNet audio links to root nodes with no conference members are cleared.
* Better terminating algorithm when a conference only consist of default members (in several nodes).
+
*When Root of conference is AMC 8/9 and there is an IP transit node with members of the conference, leaf IP nodes was not  able to connect to the conference due to CODEC selection failure.
=== HTTPS support added ===
+
*Better terminating algorithm when a conference only consist of default members (in several nodes).
 +
 
 +
===HTTPS support added===
 
Gives the possibility to access AlphaWeb over a secure connection. See [[AlphaWeb#AlphaWeb_Technical | AlphaWeb Technical]]
 
Gives the possibility to access AlphaWeb over a secure connection. See [[AlphaWeb#AlphaWeb_Technical | AlphaWeb Technical]]
=== AlphaWeb Custom Scripting ===
+
===AlphaWeb Custom Scripting===
 
[[AlphaWeb_Custom_Scripts | AlphaWeb Custom Scripts]] lets the end user extend the AlphaWeb functionality. Typically this can be 'Click to Call' type of applications.
 
[[AlphaWeb_Custom_Scripts | AlphaWeb Custom Scripts]] lets the end user extend the AlphaWeb functionality. Typically this can be 'Click to Call' type of applications.
  
Line 258: Line 275:
 
alpha_sys_10_00.tbz2 must be installed.<br><br>
 
alpha_sys_10_00.tbz2 must be installed.<br><br>
  
== Error Corrected ==
+
==Error Corrected==
===Issue 3193: IP station -> SIP: codec problems ===
+
===Issue 3193: IP station -> SIP: codec problems===
 
When an IP Substation is calling a SIP Gateway or SIP Phone, the codec of the IP station must be set to G711, else there is no audio. But when the IP substation codec is G711 there will be a "click" sound in the IP substation on regular intercom calls.
 
When an IP Substation is calling a SIP Gateway or SIP Phone, the codec of the IP station must be set to G711, else there is no audio. But when the IP substation codec is G711 there will be a "click" sound in the IP substation on regular intercom calls.
 
Fixed.
 
Fixed.
Line 280: Line 297:
  
  
 
+
==Errors Corrected==
== Errors Corrected ==
+
===Issue 2899: $SM don't trigger RelTo = UDP===
=== Issue 2899: $SM don't trigger RelTo = UDP ===
 
 
Intelligence added to the code receiving $SM (DP). If UDP group not specified, and mail sender is local, look up the UDP group from NVRAM. Can not be done if mail sender is in a different node.  
 
Intelligence added to the code receiving $SM (DP). If UDP group not specified, and mail sender is local, look up the UDP group from NVRAM. Can not be done if mail sender is in a different node.  
 
%1.udp macro added to the EventHandler, so a complete $SM can be sent in the first place.
 
%1.udp macro added to the EventHandler, so a complete $SM can be sent in the first place.
=== Issue 2952: Dual Display MDF text corrections ===
+
===Issue 2952: Dual Display MDF text corrections===
 
Some display text cleanup regarding during mail sending.
 
Some display text cleanup regarding during mail sending.
=== Issue 3142: Line errors at startup ===
+
===Issue 3142: Line errors at startup===
 
At startup there could be line errors reported at non existing physical numbers. Corrected  
 
At startup there could be line errors reported at non existing physical numbers. Corrected  
=== Issue 3161: Event handler %scutf, comma ===
+
===Issue 3161: Event handler %scutf, comma===
 
AMCD Event handler: The %scutf macro now works properly when using comma as delimiter.  
 
AMCD Event handler: The %scutf macro now works properly when using comma as delimiter.  
=== Issue 3163: Conversation incoming second user. ===
+
===Issue 3163: Conversation incoming second user.===
 
Conversation incoming [07] at B-sub (e.g.2001) which is a second user (not default) to phys. 10 (2000) does not work. The event handler is now updated so that incoming event %dir and %name reflect the actual user dialed.
 
Conversation incoming [07] at B-sub (e.g.2001) which is a second user (not default) to phys. 10 (2000) does not work. The event handler is now updated so that incoming event %dir and %name reflect the actual user dialed.
=== Issue 3164: Call request from data protocol (from other than default user) ===
+
===Issue 3164: Call request from data protocol (from other than default user)===
 
If you simulate by data a call request (sender e.g.2001) to CRM4 2500 the call request message is ok (sender 2001). You are able to call back (locally node2) with 70+8 or 7638 and you are connected to the default user (2000 FBSAR), but the call request message (initiated from 2001, not default) is not removed. If you call back by manual dialing 2001, the message is removed.  
 
If you simulate by data a call request (sender e.g.2001) to CRM4 2500 the call request message is ok (sender 2001). You are able to call back (locally node2) with 70+8 or 7638 and you are connected to the default user (2000 FBSAR), but the call request message (initiated from 2001, not default) is not removed. If you call back by manual dialing 2001, the message is removed.  
 
Correct directory number (other than default user) is now used when canceling mail.
 
Correct directory number (other than default user) is now used when canceling mail.
=== Issue 3176: SIP: Gateway blocked, reset required ===
+
===Issue 3176: SIP: Gateway blocked, reset required===
 
Reports of reset of AlphaCom required due to blocking of audio lines to Mediatrix ISDN gateway. Fixed
 
Reports of reset of AlphaCom required due to blocking of audio lines to Mediatrix ISDN gateway. Fixed
=== Issue 3177: Eventhandler fault when related to > 4 digits ===
+
===Issue 3177: Eventhandler fault when related to > 4 digits===
 
Event handler actions “Related to” more than 4 digits is now working.
 
Event handler actions “Related to” more than 4 digits is now working.
=== Issue 3182: SIP ringing terminates after 30 sec. ===
+
===Issue 3182: SIP ringing terminates after 30 sec.===
 
The SIP ringing length is now following the AlphaPro configuration.
 
The SIP ringing length is now following the AlphaPro configuration.
=== Issue 3188: Off Hook duplex when calling SIP ===
+
===Issue 3188: Off Hook duplex when calling SIP===
 
Handset operation /duplex flag is now tested during SIP setup, thus give DTMF signaling from ATLB stations.
 
Handset operation /duplex flag is now tested during SIP setup, thus give DTMF signaling from ATLB stations.
=== Issue 3195: Intermodule IP audio lockup slave reset ===  
+
===Issue 3195: Intermodule IP audio lockup slave reset===  
 
Improvment to slave module audio availablility during slave reset.
 
Improvment to slave module audio availablility during slave reset.
=== Issue 3196: $ST Lxxx W0 don’t stop timer ===
+
===Issue 3196: $ST Lxxx W0 don’t stop timer===
 
Data command “$ST Lxxx W0” will now stop all timers running on station xxx.
 
Data command “$ST Lxxx W0” will now stop all timers running on station xxx.
=== Issue 3203: SIP: Handytone INFO digits ignored workaround ===
+
===Issue 3203: SIP: Handytone INFO digits ignored workaround===
 
Workaround for problem with Grandstream Handytone 488, firmware 1.0.3.64. Ingoing call to Alphacom, then a lot of keying, for excample of * and # for simplex / groupcall keying. Then after some 10 - 20 incomming INFO messages, SIPD suddenly ignores a INFO message. This in turn locks up Grandstream, so that it does not send a BYE when the handset is replaced.  
 
Workaround for problem with Grandstream Handytone 488, firmware 1.0.3.64. Ingoing call to Alphacom, then a lot of keying, for excample of * and # for simplex / groupcall keying. Then after some 10 - 20 incomming INFO messages, SIPD suddenly ignores a INFO message. This in turn locks up Grandstream, so that it does not send a BYE when the handset is replaced.  
=== Issue 3205: Line test 7872 in multi module ===
+
===Issue 3205: Line test 7872 in multi module===
 
The 7872 line test is now working again for Multi module AGA/AE1
 
The 7872 line test is now working again for Multi module AGA/AE1
=== Issue 3215: Busy/Private override in AlphaNet ===
+
===Issue 3215: Busy/Private override in AlphaNet===
 
Busy and private override is now working over AlphaNet using M-key.
 
Busy and private override is now working over AlphaNet using M-key.
=== Issue 3218: Mail related assert: ===
+
===Issue 3218: Mail related assert:===
 
Improved clean-up system for AlphaCom mail elements.
 
Improved clean-up system for AlphaCom mail elements.
=== Issue 3217: Call from SIP to IP station ===
+
===Issue 3217: Call from SIP to IP station===
 
When calling from SIP G711 codec to IP station G722 codec, transcoding is now correctly initiated.
 
When calling from SIP G711 codec to IP station G722 codec, transcoding is now correctly initiated.
  
= AMC  10.20  (2007-04-12) =
+
=AMC  10.20  (2007-04-12)=
  
 
  Release: Official, available on request
 
  Release: Official, available on request
Line 331: Line 347:
 
alpha_sys_10_00.tbz2 must be installed.<br><br>
 
alpha_sys_10_00.tbz2 must be installed.<br><br>
  
== Functional changes / Enhancement ==
+
==Functional changes / Enhancement==
=== Private ringing tone on Call Request function ===
+
===Private ringing tone on Call Request function===
 
Private ringing tone on mail at receiver station will be active when first mail in queue has the mail priority above 150 (default value of "globel_constant>priv_ring_mail_pri")<br>
 
Private ringing tone on mail at receiver station will be active when first mail in queue has the mail priority above 150 (default value of "globel_constant>priv_ring_mail_pri")<br>
 
===New event FEAT_M_KEY (31 )===
 
===New event FEAT_M_KEY (31 )===
Line 346: Line 362:
 
Only state changes within current state is reported on M_KEY and HOOK, it means if the station goes OFF_HOOK from idle the event OFF_HOOK in busy state will not be reported.<br><br>
 
Only state changes within current state is reported on M_KEY and HOOK, it means if the station goes OFF_HOOK from idle the event OFF_HOOK in busy state will not be reported.<br><br>
  
== Errors Corrected ==
+
==Errors Corrected==
=== Issue 3114: Multi Module IP audio fails after slave reset: ===
+
===Issue 3114: Multi Module IP audio fails after slave reset:===
 
When slave is reset, without the master being reset, the IP ICC audio links does now work
 
When slave is reset, without the master being reset, the IP ICC audio links does now work
=== RTP jitter buffer improvements===  
+
===RTP jitter buffer improvements===  
 
Rtpdeamon version 01.02 improves the jitter buffer handling. Both handle larger difference in clock rate between sender and receiver, at the same more stable delay adaptation at the presence of jitter. Fixes issue 3101 with Xlite.<br>
 
Rtpdeamon version 01.02 improves the jitter buffer handling. Both handle larger difference in clock rate between sender and receiver, at the same more stable delay adaptation at the presence of jitter. Fixes issue 3101 with Xlite.<br>
 
<br>
 
<br>
Line 355: Line 371:
 
<br>
 
<br>
  
= AMC  X10.20  (2007-03-20) =
+
=AMC  X10.20  (2007-03-20)=
 
  Release: Official, available on request
 
  Release: Official, available on request
 
  /opt/amc/bin/amcd
 
  /opt/amc/bin/amcd
Line 366: Line 382:
 
alpha_sys_10_00.tbz2 must be installed.<br><br>
 
alpha_sys_10_00.tbz2 must be installed.<br><br>
  
== Errors Corrected ==
+
==Errors Corrected==
=== Issue 3179: Calling from a slave to an E1 node: ===
+
===Issue 3179: Calling from a slave to an E1 node:===
 
It was not possible to dial global numbers from a slave module in multi module over IP to a different node connected with the master over AE1. (10.20x bug)<br>
 
It was not possible to dial global numbers from a slave module in multi module over IP to a different node connected with the master over AE1. (10.20x bug)<br>
 
<br>
 
<br>
 
<br>
 
<br>
 
<br>
 
<br>
= AMC  X10.20  (2007-03-15) =
+
=AMC  X10.20  (2007-03-15)=
 
  Release: Official, available on request
 
  Release: Official, available on request
 
  /opt/amc/bin/amcd
 
  /opt/amc/bin/amcd
Line 383: Line 399:
 
alpha_sys_10_00.tbz2 must be installed.<br><br>
 
alpha_sys_10_00.tbz2 must be installed.<br><br>
  
== Functional changes / Enhancement ==
+
==Functional changes / Enhancement==
=== Ring Master Daemon is included in the package. ===  
+
===Ring Master Daemon is included in the package.===  
 
See separate documentation.<br>
 
See separate documentation.<br>
  
=== Support of IP stations.<br> ===  
+
===Support of IP stations. ===  
 +
 
 
*- Substation functionality
 
*- Substation functionality
 
*- Group Call is available
 
*- Group Call is available
 
*- Outputs on the stations can be related to RCO in AlphaPro. (RCO type station)
 
*- Outputs on the stations can be related to RCO in AlphaPro. (RCO type station)
 
*- Inputs can be related to DAK or directly to DAK_AS_RCI event in AlphaPro.
 
*- Inputs can be related to DAK or directly to DAK_AS_RCI event in AlphaPro.
 +
 
See separate documentation.<br>
 
See separate documentation.<br>
  
=== IP audio transit capacity is increased from 32 to 64 channels. ===  
+
===IP audio transit capacity is increased from 32 to 64 channels.===  
 
Transit audio is audio not going trough the backplane.
 
Transit audio is audio not going trough the backplane.
 
(SIP to SIP, IP station to IP station, AlphaNet IP transit or a combination of IP audio)<br />
 
(SIP to SIP, IP station to IP station, AlphaNet IP transit or a combination of IP audio)<br />
Line 401: Line 419:
 
Information page now contains node name and node software versions.<br /><br />
 
Information page now contains node name and node software versions.<br /><br />
  
== Errors Corrected ==
+
==Errors Corrected==
=== Issue 2885: Log enabling: ===
+
===Issue 2885: Log enabling:===
 
System log to Syslog is now working even when log port is not enabled in AlphaPro.<br />
 
System log to Syslog is now working even when log port is not enabled in AlphaPro.<br />
  
=== Issue 2935: SIP Softphone ExpressTalk causes AMC reset: ===
+
===Issue 2935: SIP Softphone ExpressTalk causes AMC reset:===
 
SIP Softphone ExpressTalk would reset the exchange when the conversation is cancelled from the phone (both incoming and outgoing calls). <br />
 
SIP Softphone ExpressTalk would reset the exchange when the conversation is cancelled from the phone (both incoming and outgoing calls). <br />
  
=== Issue 3053: Conversation Outgoing event: ===
+
===Issue 3053: Conversation Outgoing event:===
 
When making a call from intercom out on a SIP gateway (Mediatrix 2400) the event Conversation Outgoing is now triggered when the phone answers.<br />
 
When making a call from intercom out on a SIP gateway (Mediatrix 2400) the event Conversation Outgoing is now triggered when the phone answers.<br />
  
=== Issue 3154: $CALL in AlphaNet don't work: ===
+
===Issue 3154: $CALL in AlphaNet don't work:===
 
The data protocol command $CALL is now working in AlphaNet.<br />
 
The data protocol command $CALL is now working in AlphaNet.<br />
  
=== Issue 3169: Serial communication block AMCD: ===
+
===Issue 3169: Serial communication block AMCD:===
 
AMC would not start after power reset if serial communication was active during reset. (Other node sending data to resetting AMC) This will not happen on system running Linux 2.4. <br />
 
AMC would not start after power reset if serial communication was active during reset. (Other node sending data to resetting AMC) This will not happen on system running Linux 2.4. <br />
  
=== Issue 3180: Program 7 has no display txt: ===
+
===Issue 3180: Program 7 has no display txt:===
 
Audio program 7 (feat 5/7): The display text was not shown in the display when activating the audio program 7. For other programs it is ok. (I have seen this long time ago also, so it might be an old bug).
 
Audio program 7 (feat 5/7): The display text was not shown in the display when activating the audio program 7. For other programs it is ok. (I have seen this long time ago also, so it might be an old bug).
 
Static License of 2 AlphaNet audio links does not work correctly. Only one audio link was working.
 
Static License of 2 AlphaNet audio links does not work correctly. Only one audio link was working.
Line 425: Line 443:
 
<br />
 
<br />
 
<br />
 
<br />
= AMC  10.05  (2007-02-07) =
+
=AMC  10.05  (2007-02-07)=
 
  Release: Official, available on request
 
  Release: Official, available on request
 
  /opt/amc/bin/amcd
 
  /opt/amc/bin/amcd
Line 436: Line 454:
 
alpha_sys_10_00.tbz2 must be installed.<br /><br />
 
alpha_sys_10_00.tbz2 must be installed.<br /><br />
  
== Functional changes / Enhancement ==
+
==Functional changes / Enhancement==
=== License keys: ===
+
===License keys:===
 
SIP trunk and SIP stations is now separated in two different licenses.<br />
 
SIP trunk and SIP stations is now separated in two different licenses.<br />
 
SIP trunk and AlphaNet licenses is by default dynamic except for the 2 line AlphaNet license that still is static.<br />
 
SIP trunk and AlphaNet licenses is by default dynamic except for the 2 line AlphaNet license that still is static.<br />
Line 445: Line 463:
 
Priority of audio resources allocation is handled by the priority of the initiator station.<br /><br />
 
Priority of audio resources allocation is handled by the priority of the initiator station.<br /><br />
  
== Errors Corrected ==
+
==Errors Corrected==
=== Issue 2715: AlphaNet Duplex in combination with AMC 8/9: ===
+
===Issue 2715: AlphaNet Duplex in combination with AMC 8/9:===
 
In combination AlphaNet with AlphaCom 8 and 9 and exchanges via IP there will be a problem with delay adjustment of the duplex algorithm when calls are made from IP to an AMC 8/9 exchange.<br />
 
In combination AlphaNet with AlphaCom 8 and 9 and exchanges via IP there will be a problem with delay adjustment of the duplex algorithm when calls are made from IP to an AMC 8/9 exchange.<br />
 
The duplex algorithm is now run in A node when going from an ASLT station to IP AlphaNet thus avoiding duplex in end AMC 8/9 nodes. <br />
 
The duplex algorithm is now run in A node when going from an ASLT station to IP AlphaNet thus avoiding duplex in end AMC 8/9 nodes. <br />
 
In transit systems with certain combinations of AMC- 8/9/IP, SIP, AGA, AE1 and Multi module IP some issues could still occur that needs special configuration. (See separate duplex document). <br />
 
In transit systems with certain combinations of AMC- 8/9/IP, SIP, AGA, AE1 and Multi module IP some issues could still occur that needs special configuration. (See separate duplex document). <br />
  
=== Issue 2887/3113: SIP automatic duplex switching: ===
+
===Issue 2887/3113: SIP automatic duplex switching:===
 
When calling SIP there can be problems with the standard duplex algorithm due to DSP echo cancelling in the SIP station.
 
When calling SIP there can be problems with the standard duplex algorithm due to DSP echo cancelling in the SIP station.
 
A new duplex algorithm is available for duplex towards SIP stations that is speech controlled only from volume of the microphone signal from the SIP link.<br />
 
A new duplex algorithm is available for duplex towards SIP stations that is speech controlled only from volume of the microphone signal from the SIP link.<br />
 
(exchange flag “DSP_duplex = 1”) <br />
 
(exchange flag “DSP_duplex = 1”) <br />
  
=== Issue 2897: X-lite on-hold lockup: ===
+
===Issue 2897: X-lite on-hold lockup:===
 
Problems during on-hold feature in X-lite fixed.<br />
 
Problems during on-hold feature in X-lite fixed.<br />
  
=== Issue 2908: SIP-Ringing if call cancelled before answer: ===
+
===Issue 2908: SIP-Ringing if call cancelled before answer:===
 
Calling from AMC-station to SIP phone. If C-key is pressed on AlphaCom-station before the call is answered, the SIP phone keeps ringing if calling via a transit node. <br />
 
Calling from AMC-station to SIP phone. If C-key is pressed on AlphaCom-station before the call is answered, the SIP phone keeps ringing if calling via a transit node. <br />
  
=== Issue 2960/3119: SIP-Handy-Tone 488 making All Call.(* and # key): ===
+
===Issue 2960/3119: SIP-Handy-Tone 488 making All Call.(* and # key):===
 
SIP now handles * and # for  both Grandstream and Mediatrix.<br />
 
SIP now handles * and # for  both Grandstream and Mediatrix.<br />
  
=== Issue 3011: Duplex switching in mixed environment: ===
+
===Issue 3011: Duplex switching in mixed environment:===
 
When the audio path go transit from IP to analog/E1 delay information to the automatic duplex routine is lost.  
 
When the audio path go transit from IP to analog/E1 delay information to the automatic duplex routine is lost.  
 
Duplex delay is now forwarder/backwarded from transit AGA to IP links to the duplex node.<br />
 
Duplex delay is now forwarder/backwarded from transit AGA to IP links to the duplex node.<br />
  
=== Issue 3012: Echo in SIP handset: ===
+
===Issue 3012: Echo in SIP handset:===
 
When talking with handset conversation between Intergard station and SIP station the SIP station will get echo in the handset due to overhearing in AlphaCom handset (and no echo cancelling).
 
When talking with handset conversation between Intergard station and SIP station the SIP station will get echo in the handset due to overhearing in AlphaCom handset (and no echo cancelling).
 
Handset to handset communication will now be forced in duplex. (Default delay setting for SIP = 30ms. Full duplex can be obtained if parameter “max_off_hook_delay” is adjusted to 40 ms or more)<br />
 
Handset to handset communication will now be forced in duplex. (Default delay setting for SIP = 30ms. Full duplex can be obtained if parameter “max_off_hook_delay” is adjusted to 40 ms or more)<br />
  
=== Issue 3015: AlphaNet: No Camp On Busy: ===
+
===Issue 3015: AlphaNet: No Camp On Busy:===
 
There is no "camp on busy" when all AlphaNet lines are in use. Instead one gets a rejection tone. Same behaviour with AGA line, AE1 and VoIP.  
 
There is no "camp on busy" when all AlphaNet lines are in use. Instead one gets a rejection tone. Same behaviour with AGA line, AE1 and VoIP.  
 
Feature implemented.<br />
 
Feature implemented.<br />
  
=== Issue 3017: Name list: ===
+
===Issue 3017: Name list:===
 
After AMC auto load dirno's 9542 - 9545 are in the name list (614).<br />
 
After AMC auto load dirno's 9542 - 9545 are in the name list (614).<br />
  
=== Issue 3034: Time management: ===
+
===Issue 3034: Time management:===
 
When changing time in AlphaWeb, the new time is now also written to the hardware clock.<br />
 
When changing time in AlphaWeb, the new time is now also written to the hardware clock.<br />
  
=== Issue 3051: IP address with leading 0's: ===
+
===Issue 3051: IP address with leading 0's:===
 
Interpretation of leading 0 in AlphaWeb is fixed. <br />
 
Interpretation of leading 0 in AlphaWeb is fixed. <br />
  
=== Issue 3067: AlphaWeb: Same subnet on Eth0 and 1: ===
+
===Issue 3067: AlphaWeb: Same subnet on Eth0 and 1:===
 
Configuration of both Ethernet phys on the same sub net is now tested.<br />
 
Configuration of both Ethernet phys on the same sub net is now tested.<br />
  
=== Issue 3070: SIP: Long phone no on DAK/Substation: ===
+
===Issue 3070: SIP: Long phone no on DAK/Substation:===
 
Up to 16 digit phone number now allowed in DAK "784" and Call-forward "71".<br />
 
Up to 16 digit phone number now allowed in DAK "784" and Call-forward "71".<br />
  
=== Issue 3116: AMC: Speech channel locks up: ===
+
===Issue 3116: AMC: Speech channel locks up:===
 
During global conference and failed SIP calls speech channels could be locked up.<br />
 
During global conference and failed SIP calls speech channels could be locked up.<br />
  
=== Issue 3143: AMC: 99 answer in global group call: ===
+
===Issue 3143: AMC: 99 answer in global group call:===
 
Not possible to answer global group calls from other that initiating node. This feature is fixed.<br />
 
Not possible to answer global group calls from other that initiating node. This feature is fixed.<br />
  
=== Issue 3060: SIP & AlphaNet licenses: ===
+
===Issue 3060: SIP & AlphaNet licenses:===
 
It is now possible to install 30 AlphaNet and 20 SIP trunk licenses. <br />
 
It is now possible to install 30 AlphaNet and 20 SIP trunk licenses. <br />
  
=== Issue 3069: SIP: Phone number in display: ===
+
===Issue 3069: SIP: Phone number in display:===
 
For outgoing/incoming calls the number of shown digits is 16 (including event handler).<br />
 
For outgoing/incoming calls the number of shown digits is 16 (including event handler).<br />
  
=== Issue 3087: SIP: Mediatrix 1204 - Dial Out: ===
+
===Issue 3087: SIP: Mediatrix 1204 - Dial Out:===
 
Transmit digit by digit as dialled now supports:
 
Transmit digit by digit as dialled now supports:
* -  Programming of phone number on DAK from AlphaPro
+
 
* -  DAK, "784" from station
+
*-  Programming of phone number on DAK from AlphaPro
* -  Call forwarding from station "71"
+
*-  DAK, "784" from station
 +
*-  Call forwarding from station "71"
 +
 
 
Both for Mediatrix and AudioCodes<br />
 
Both for Mediatrix and AudioCodes<br />
  
=== Issue 3105: SIP Trunk: DAK call fails (AudioCodes): ===
+
===Issue 3105: SIP Trunk: DAK call fails (AudioCodes):===
 
The first or the two first DTMF digits of the phone number is not transmitted.  
 
The first or the two first DTMF digits of the phone number is not transmitted.  
 
400 ms delay before sending digits. Delay can be extended with exchange timeout "sip_dial_dly".<br />
 
400 ms delay before sending digits. Delay can be extended with exchange timeout "sip_dial_dly".<br />
  
=== Issue 3107: SIP Trunk: Call Forward (71): ===
+
===Issue 3107: SIP Trunk: Call Forward (71):===
 
Manual transfer to phone using 71 don't work (PNCI you could 71 + 0 + phone + M, or 71 + <shortnumber>). Fixed<br />
 
Manual transfer to phone using 71 don't work (PNCI you could 71 + 0 + phone + M, or 71 + <shortnumber>). Fixed<br />
  
=== Issue 3136: AlphaNet: Global SX Conference: ===
+
===Issue 3136: AlphaNet: Global SX Conference:===
 
AlphaNet: Global SX Conference. Problems after node reset and issues of reconnect are fixed.<br />
 
AlphaNet: Global SX Conference. Problems after node reset and issues of reconnect are fixed.<br />
  
=== Issue 3141: Call to unregistered SIP stations: ===
+
===Issue 3141: Call to unregistered SIP stations:===
 
Call from AlphaCom to a SIP phone, SIP phone is configured at registrar node, but the phone is not registered: Then SIP sends the INVITE to its own IP address, which is processed, and forwarded in a loop until all RTP resources are used up.
 
Call from AlphaCom to a SIP phone, SIP phone is configured at registrar node, but the phone is not registered: Then SIP sends the INVITE to its own IP address, which is processed, and forwarded in a loop until all RTP resources are used up.
 
IP address check implemented.<br />
 
IP address check implemented.<br />
  
=== Issue 3156: AlphaWeb show no licence: ===
+
===Issue 3156: AlphaWeb show no licence:===
 
AMC now generate correct  infor in the license info file.<br />
 
AMC now generate correct  infor in the license info file.<br />
 
<br />
 
<br />
 
<br />
 
<br />
 
<br />
 
<br />
= AMC  X10.05  (2007-01-18) =
+
=AMC  X10.05  (2007-01-18)=
 
  Release: Beta, available on request
 
  Release: Beta, available on request
 
  /opt/amc/bin/amcd
 
  /opt/amc/bin/amcd
Line 541: Line 561:
 
alpha_sys_10_00.tbz2 must be installed.<br /><br />
 
alpha_sys_10_00.tbz2 must be installed.<br /><br />
  
== Errors Corrected ==
+
==Errors Corrected==
=== Issue 3072: $CPYM removes two char in name: ===
+
===Issue 3072: $CPYM removes two char in name:===
 
When a mail is copied to another station using $CPYM L%1.dir W%2.tag L<dirno>, the two first characters in the name of the sender is removed. E.g. if the sender is "Donald Duck" it will appear as "nald Duck".<br>
 
When a mail is copied to another station using $CPYM L%1.dir W%2.tag L<dirno>, the two first characters in the name of the sender is removed. E.g. if the sender is "Donald Duck" it will appear as "nald Duck".<br>
  
=== Issue 3074: 626 cancel call request: ===
+
===Issue 3074: 626 cancel call request:===
 
The problem is related to 626. This code blocks the station for a few seconds when you hang up.<br>
 
The problem is related to 626. This code blocks the station for a few seconds when you hang up.<br>
  
=== Issue 3088: Transfer of outgoing calls: ===
+
===Issue 3088: Transfer of outgoing calls:===
 
When doing an outgoing phone call from AlphaCom via SIP Gateway (tested with Mediatrix 1204 (analogue) and Mediatrix ISDN) you cannot transfer the call to another intercom station.<br>
 
When doing an outgoing phone call from AlphaCom via SIP Gateway (tested with Mediatrix 1204 (analogue) and Mediatrix ISDN) you cannot transfer the call to another intercom station.<br>
  
=== Issue 3115: AlphaNet Global SX Conference: ===
+
===Issue 3115: AlphaNet Global SX Conference:===
 
Cancelling and reinitiate a global conference do not distribute audio. Also problems when resetting member nodes of a conference.<br>
 
Cancelling and reinitiate a global conference do not distribute audio. Also problems when resetting member nodes of a conference.<br>
 
<br>
 
<br>
Line 557: Line 577:
 
<br>
 
<br>
  
= AMC  X10.05  (2006-12-21) =
+
=AMC  X10.05  (2006-12-21)=
 
  Release: Beta, available on request
 
  Release: Beta, available on request
 
  /opt/amc/bin/amcd
 
  /opt/amc/bin/amcd
Line 568: Line 588:
 
alpha_sys_10_00.tbz2 must be installed.<br /><br />
 
alpha_sys_10_00.tbz2 must be installed.<br /><br />
  
== Errors Corrected ==
+
==Errors Corrected==
=== Issue 3111: Multi Module Group Call Block: ===
+
===Issue 3111: Multi Module Group Call Block:===
 
MultiModule and group call could block. Fixed one reproduced case.<br />
 
MultiModule and group call could block. Fixed one reproduced case.<br />
  
=== Issue 3112: Reset after cancel of call setup by the $CALL command: ===
+
===Issue 3112: Reset after cancel of call setup by the $CALL command:===
 
When resetting A-station in a call established by the $CALL the exchange did a reset. Fixed<br />
 
When resetting A-station in a call established by the $CALL the exchange did a reset. Fixed<br />
  
=== Issue 3121: In hotline call off-hook action do not work: ===
+
===Issue 3121: In hotline call off-hook action do not work:===
 
Fixed.<br />
 
Fixed.<br />
 
<br />
 
<br />
 
<br />
 
<br />
 
<br />
 
<br />
= AMC  10.04  (2006-12-12) =
+
=AMC  10.04  (2006-12-12)=
 
  Release: Official, available on request
 
  Release: Official, available on request
 
  /opt/amc/bin/amcd
 
  /opt/amc/bin/amcd
Line 591: Line 611:
 
alpha_sys_10_00.tbz2 must be installed.<br /><br />
 
alpha_sys_10_00.tbz2 must be installed.<br /><br />
  
== Functional changes / Enhancement ==
+
==Functional changes / Enhancement==
=== AlphaPro Password: ===
+
===AlphaPro Password:===
 
The AlphaPro password is now the same as the administrator password for AlphaWeb (default user: admin, password alphaadmin)<br />
 
The AlphaPro password is now the same as the administrator password for AlphaWeb (default user: admin, password alphaadmin)<br />
  
=== Issue 3031: Ignor station down as default auto load: ===
+
===Issue 3031: Ignor station down as default auto load:===
 
The flag is enalbled as default from AMC 10.04<br />
 
The flag is enalbled as default from AMC 10.04<br />
  
Line 601: Line 621:
 
  TST->Nvram “ex_profile.timeouts.dig_col_timeout” configures the inter digit timeout in 100 ms steps during collection of digits on feature 81 and 83. Default is 3 seconds.<br /><br />
 
  TST->Nvram “ex_profile.timeouts.dig_col_timeout” configures the inter digit timeout in 100 ms steps during collection of digits on feature 81 and 83. Default is 3 seconds.<br /><br />
  
== Errors Corrected ==
+
==Errors Corrected==
=== Issue 3024/3039: AlphaNet & MultiModule IP - 7872 causes reset: ===
+
===Issue 3024/3039: AlphaNet & MultiModule IP - 7872 causes reset:===
 
MultiModule over IP: Dialling 7872 on station 101 in master causes reset.<br />
 
MultiModule over IP: Dialling 7872 on station 101 in master causes reset.<br />
  
=== Issue 3038: EDIO - %sscan: ===
+
===Issue 3038: EDIO - %sscan:===
 
The macro %sscan(string, search-string)  returned wrong result when there was no match with the search-string.<br />
 
The macro %sscan(string, search-string)  returned wrong result when there was no match with the search-string.<br />
  
=== Issue 3076/3085: Special characters in AlphaCom display text: ===
+
===Issue 3076/3085: Special characters in AlphaCom display text:===
 
AlphaCom display text could not include /-,. æøå etc. when making an SIP call.<br />
 
AlphaCom display text could not include /-,. æøå etc. when making an SIP call.<br />
  
=== Issue 3086: TST Error messages during SIP call: ===
+
===Issue 3086: TST Error messages during SIP call:===
 
Error messages introduced in 10.03 during call SIP reset is now fixed.<br />
 
Error messages introduced in 10.03 during call SIP reset is now fixed.<br />
  
=== Issue 3089:  Events when not "Default User”: ===
+
===Issue 3089:  Events when not "Default User”:===
 
The eventhandler is now working for Conversation event “related to” user that are not default users.<br />
 
The eventhandler is now working for Conversation event “related to” user that are not default users.<br />
  
=== Issue 3091: Mediatrix 1204 - No M-key */#: ===
+
===Issue 3091: Mediatrix 1204 - No M-key */#:===
 
Key signaling of * and # from mediatrix is now implemented. Signaling of * and # out of SIPD is now also using * and # towards SIP equipment<br />
 
Key signaling of * and # from mediatrix is now implemented. Signaling of * and # out of SIPD is now also using * and # towards SIP equipment<br />
  
=== Issue 3093: RCO PULS bug: ===
+
===Issue 3093: RCO PULS bug:===
 
It was possible to activate the RCO puls system by accident from the event handler scripts due to error in the script parser. (The RCO system then got inverted behavior)<br />
 
It was possible to activate the RCO puls system by accident from the event handler scripts due to error in the script parser. (The RCO system then got inverted behavior)<br />
  
=== Issue 3095: Context params in built in cmds, RCO: ===
+
===Issue 3095: Context params in built in cmds, RCO:===
 
The “%chg(on,off)” command did not work correctly for rco.<br />
 
The “%chg(on,off)” command did not work correctly for rco.<br />
  
=== Issue 3096: Mediatrix 1204, direct dial out: ===
+
===Issue 3096: Mediatrix 1204, direct dial out:===
 
SIPD is now hanling collected digits from AlphaCom.<br />
 
SIPD is now hanling collected digits from AlphaCom.<br />
  
=== Issue 3098: Multi Module Group call on AGA lines: ===
+
===Issue 3098: Multi Module Group call on AGA lines:===
 
Group Call with AMC IP master and AGA/AE1 audio could reset the exchange.<br />
 
Group Call with AMC IP master and AGA/AE1 audio could reset the exchange.<br />
  
=== Issue 3103: Eventhandler; left adjustment of %: ===
+
===Issue 3103: Eventhandler; left adjustment of %:===
 
Left adjustment of %macroes did not work.  Example: "%1.exp(3<)"<br />
 
Left adjustment of %macroes did not work.  Example: "%1.exp(3<)"<br />
 
<br />
 
<br />
 
<br />
 
<br />
 
<br />
 
<br />
= AMC  10.03  (2006-20-11) =
+
=AMC  10.03  (2006-20-11)=
 
  Release: Official, available on request
 
  Release: Official, available on request
 
  /opt/amc/bin/amcd
 
  /opt/amc/bin/amcd
Line 649: Line 669:
 
alpha_sys_10_00.tbz2 must be installed.<br /><br />
 
alpha_sys_10_00.tbz2 must be installed.<br /><br />
  
== Functional changes ==
+
==Functional changes==
=== Pulse RCO: ===
+
===Pulse RCO:===
 
Support for generating a pulse of specified length on a “logical” RCO. <br />
 
Support for generating a pulse of specified length on a “logical” RCO. <br />
 
New parameter “Duration” is added to the event handler-built-in command “RCO” and the data protocol message $SLRC. Examples, pulse RCO # 13 on for 1 second: “rco 13 on 10” or “$SLRC W13 1 w10”. <br />
 
New parameter “Duration” is added to the event handler-built-in command “RCO” and the data protocol message $SLRC. Examples, pulse RCO # 13 on for 1 second: “rco 13 on 10” or “$SLRC W13 1 w10”. <br />
 
Duration is in tenths of second. If Duration parameter is present, and non-zero, the RCO state specified in previous parameter lasts for the specified time. After the time has expired, the RCO is toggled to the opposite state. If a new $SLRC/RCO on the same RCO arrives while the pulse timer is running, the timer is cancelled, and the new message determines the new RCO state. Duration = 0 means infinite duration.<br />
 
Duration is in tenths of second. If Duration parameter is present, and non-zero, the RCO state specified in previous parameter lasts for the specified time. After the time has expired, the RCO is toggled to the opposite state. If a new $SLRC/RCO on the same RCO arrives while the pulse timer is running, the timer is cancelled, and the new message determines the new RCO state. Duration = 0 means infinite duration.<br />
  
=== DAK RCI  EVENT: ===
+
===DAK RCI  EVENT:===
 
New event Dak key as RCI (30). Sub event = DAK key number. ON = Dak key press, OFF = dak key release.<br />
 
New event Dak key as RCI (30). Sub event = DAK key number. ON = Dak key press, OFF = dak key release.<br />
  
=== Private ringing for incomming SIP calls: ===
+
===Private ringing for incomming SIP calls:===
 
New tst nvram flag available ”ex_profile.flags.private_ringing_SIP”. If ”1” then calls from SIP will use private ringing mode. If “0” then calls will connect as an intercom call with normal priority.<br /><br />
 
New tst nvram flag available ”ex_profile.flags.private_ringing_SIP”. If ”1” then calls from SIP will use private ringing mode. If “0” then calls will connect as an intercom call with normal priority.<br /><br />
=== Station Bit "no display" ===
+
===Station Bit "no display"===
 
Added support for turning off the automatic display detection of a station. Useful for sub stations etc.
 
Added support for turning off the automatic display detection of a station. Useful for sub stations etc.
  
== Errors Corrected ==
+
==Errors Corrected==
=== Multi module in combination with AMC 8 or 9 reset: ===
+
===Multi module in combination with AMC 8 or 9 reset:===
 
Incompatible tone handling fixed.<br />
 
Incompatible tone handling fixed.<br />
  
=== Issue 2948: Technical Alarm Inputs (RCI) on AC7: ===
+
===Issue 2948: Technical Alarm Inputs (RCI) on AC7:===
 
Amc software RCI supports for AMC hardware 8000/4.<br />
 
Amc software RCI supports for AMC hardware 8000/4.<br />
  
=== Issue 3027: MultiModule and SIP: ===
+
===Issue 3027: MultiModule and SIP:===
 
Combination of Multi Module and SIP is working .<br />
 
Combination of Multi Module and SIP is working .<br />
  
=== Issue 3032: Search string on AlphaNet IP: ===
+
===Issue 3032: Search string on AlphaNet IP:===
 
When doing a call from node A to Node B and the station in node B has a search string active the search will now work.<br />
 
When doing a call from node A to Node B and the station in node B has a search string active the search will now work.<br />
  
=== Issue 3055: Slow call setup when dialling from Master to Slave: ===
+
===Issue 3055: Slow call setup when dialling from Master to Slave:===
 
Software “bottle neck” fixed.<br />
 
Software “bottle neck” fixed.<br />
  
=== Issue 3067: Volume (783) and COS (7873) are not stored in flash: ===
+
===Issue 3067: Volume (783) and COS (7873) are not stored in flash:===
 
Volume and COS setting is now stored in nvram for survival after reset.<br /><br />
 
Volume and COS setting is now stored in nvram for survival after reset.<br /><br />
  
== SIPD 01.02 upgrades and corrections ==
+
==SIPD 01.02 upgrades and corrections==
=== Event “Conversation Outgoing” : ===
+
===Event “Conversation Outgoing” :===
 
Outgoing calls to SIP: Report Event “Conversation Outgoing” (8) ON when receiving SIP 200 OK, reports OFF at disconnect. <br />
 
Outgoing calls to SIP: Report Event “Conversation Outgoing” (8) ON when receiving SIP 200 OK, reports OFF at disconnect. <br />
  
=== CANCEL_MAIL: ===
+
===CANCEL_MAIL:===
 
Outgoing calls to SIP: Send CANCEL_MAIL for outgoing call, which cancels call request from that SIP phone.<br />
 
Outgoing calls to SIP: Send CANCEL_MAIL for outgoing call, which cancels call request from that SIP phone.<br />
  
=== Events for digits from called SIP: ===
+
===Events for digits from called SIP:===
 
Outgoing calls to SIP: If called phone presses digits which are sent to AlphaCom as SIP INFO, "Event Trigger Feature" (15) is reported. The digit is sub event (0-9, * = 10, # = 11). Calling AlphaCom station is Event Owner, and called SIP phone number and node number is related to. This could be used for e.g. Dooropening.<br />
 
Outgoing calls to SIP: If called phone presses digits which are sent to AlphaCom as SIP INFO, "Event Trigger Feature" (15) is reported. The digit is sub event (0-9, * = 10, # = 11). Calling AlphaCom station is Event Owner, and called SIP phone number and node number is related to. This could be used for e.g. Dooropening.<br />
  
=== Dialing after initial connect sent as SIP INFO: ===
+
===Dialing after initial connect sent as SIP INFO:===
 
Outgoing calls to SIP: Digits dialed after initial connect are forwarded as SIP INFO digit messages to the remote SIP device. This can be used for two step dialing from AlphaCom to an external system, or to the PSTN.<br />
 
Outgoing calls to SIP: Digits dialed after initial connect are forwarded as SIP INFO digit messages to the remote SIP device. This can be used for two step dialing from AlphaCom to an external system, or to the PSTN.<br />
  
=== Register from port different than 5060 allowed: ===
+
===Register from port different than 5060 allowed:===
 
Call to SIP: Use port-number stored in registry for contact. Allows SIP device to register from any (random) port, not just 5060. Fixes issue with XLITE 3.0.<br />
 
Call to SIP: Use port-number stored in registry for contact. Allows SIP device to register from any (random) port, not just 5060. Fixes issue with XLITE 3.0.<br />
  
=== Registrar, remove all bindings: ===
+
===Registrar, remove all bindings:===
 
Store only one binding for each user. Remove binding if expires==0. Fixes failure if SIP device registered again from different address/port<br />
 
Store only one binding for each user. Remove binding if expires==0. Fixes failure if SIP device registered again from different address/port<br />
  
=== SDP media description: ===
+
===SDP media description:===
 
Call from SIP: handle that SDP a= missing, use m=... Fixes issue with XLITE 3.0.<br />
 
Call from SIP: handle that SDP a= missing, use m=... Fixes issue with XLITE 3.0.<br />
  
=== Issue 3059: SIP response handling, call to SIP: ===
+
===Issue 3059: SIP response handling, call to SIP:===
 
Call to SIP: Improved handling of non-successful SIP responses 300->699: Forward response to AMCD followed before proper disconnect of call towards AMCD. Finally SIP ACK is now sent.<br />
 
Call to SIP: Improved handling of non-successful SIP responses 300->699: Forward response to AMCD followed before proper disconnect of call towards AMCD. Finally SIP ACK is now sent.<br />
 
This allows AMCD to handle busy SIP phone properly, and avoids retransmissions from SIP phones due to missing the ACK.
 
This allows AMCD to handle busy SIP phone properly, and avoids retransmissions from SIP phones due to missing the ACK.
 
Forward 1xx invite responses to AMCD, instead of sending a faked 180 ringing directly on AudioPathSetup from AMCD. Fixes issue with short ring tone when two-step dialing to a SIP gateway.<br />
 
Forward 1xx invite responses to AMCD, instead of sending a faked 180 ringing directly on AudioPathSetup from AMCD. Fixes issue with short ring tone when two-step dialing to a SIP gateway.<br />
  
=== Call on Hold from SIP: ===
+
===Call on Hold from SIP:===
 
When receiving a re-INVITE, respond with 488 not acceptable here.  
 
When receiving a re-INVITE, respond with 488 not acceptable here.  
 
Quick fix of problem that X-lite was not able to disconnect a call if the call was put on hold by pressing other line button.<br />
 
Quick fix of problem that X-lite was not able to disconnect a call if the call was put on hold by pressing other line button.<br />
  
=== Display name: ===
+
===Display name:===
 
Outgoing calls to SIP: don’t send STATION_INFO to AMCD if called name not received from SIP. Fixes issue with blank upper display line for outgoing calls.<br />
 
Outgoing calls to SIP: don’t send STATION_INFO to AMCD if called name not received from SIP. Fixes issue with blank upper display line for outgoing calls.<br />
  
=== Issue 3058: Default route was required: ===
+
===Issue 3058: Default route was required:===
 
Fixed failure in IP address lookup if default route not defined for the system, even if no routing was required.<br />
 
Fixed failure in IP address lookup if default route not defined for the system, even if no routing was required.<br />
  
=== Crash fixes: ===
+
===Crash fixes:===
 
Fixed memory crash if SIP 1xx response received out of context. Fixed crash if receiving REGISTER with more than 9 digits in user name<br />
 
Fixed memory crash if SIP 1xx response received out of context. Fixed crash if receiving REGISTER with more than 9 digits in user name<br />
  
=== SIP Debug/Trace: ===
+
===SIP Debug/Trace:===
 
SIP trace/debug console on UNIX socket /tmp/sipd_trace. Connect using "tst -s /tmp/sipd_trace" from local shell. Possible to change trace level during runtime, press digits 0-7. Default level is 4, which prints SIP and AlphaNet messages. Press 0 to stop tracing. Formatting of message trace improved.<br />
 
SIP trace/debug console on UNIX socket /tmp/sipd_trace. Connect using "tst -s /tmp/sipd_trace" from local shell. Possible to change trace level during runtime, press digits 0-7. Default level is 4, which prints SIP and AlphaNet messages. Press 0 to stop tracing. Formatting of message trace improved.<br />
  
=== SIPd crash debug: ===
+
===SIPd crash debug:===
 
If SIPD crashes due to segmentation violation, a short debug message is printed to “/var/log/sipd_crashes”. Only the last crash is recorded, to avoid filling up the file system.<br />
 
If SIPD crashes due to segmentation violation, a short debug message is printed to “/var/log/sipd_crashes”. Only the last crash is recorded, to avoid filling up the file system.<br />
  
=== Issue 3007: SIP - digit during connection: ===
+
===Issue 3007: SIP - digit during connection:===
 
It is now possible to send digits from AlphaCom to SIP during a SIP connection.<br /><br />
 
It is now possible to send digits from AlphaCom to SIP during a SIP connection.<br /><br />
  
=== SIPD 01.02 known limitations ===  
+
===SIPD 01.02 known limitations===  
 
----
 
----
 
SIPD do not remove expired  REGISTERs. Can cause problems if changing numbers and node numbers of SIP phones. Workaround: Log in, and delete "/var/opt/amc/nvram/sip_registrar", reset exchange.<br />
 
SIPD do not remove expired  REGISTERs. Can cause problems if changing numbers and node numbers of SIP phones. Workaround: Log in, and delete "/var/opt/amc/nvram/sip_registrar", reset exchange.<br />
Line 739: Line 759:
 
<br />
 
<br />
 
<br />
 
<br />
= AMC  10.02  (2006-08-22) =
+
=AMC  10.02  (2006-08-22)=
 
  Release: Official, available on request  
 
  Release: Official, available on request  
 
  /opt/amc/bin/amcd
 
  /opt/amc/bin/amcd
Line 752: Line 772:
 
Only FPGA for the backplane issue is functionally changed. (FPGA 1.66) AMCD is functionally equal to 10.01.<br />
 
Only FPGA for the backplane issue is functionally changed. (FPGA 1.66) AMCD is functionally equal to 10.01.<br />
  
== Errors Corrected ==
+
==Errors Corrected==
=== Issue 2742: FPGA problems with backplanes: ===
+
===Issue 2742: FPGA problems with backplanes:===
 
In AlphaCom E20 and E26 with green backplanes there could be an audio issue with approx. more than 10 cards in the rack. (22 cards with FPGA 1.60),<br />
 
In AlphaCom E20 and E26 with green backplanes there could be an audio issue with approx. more than 10 cards in the rack. (22 cards with FPGA 1.60),<br />
 
<br />
 
<br />
 
<br />
 
<br />
 
<br />
 
<br />
= AMC  10.01  (2006-08-09) =
+
=AMC  10.01  (2006-08-09)=
 
  Release: Official, available on request  
 
  Release: Official, available on request  
 
  /opt/amc/bin/amcd
 
  /opt/amc/bin/amcd
Line 769: Line 789:
 
alpha_sys_10_00.tbz2 must be installed.<br /><br />
 
alpha_sys_10_00.tbz2 must be installed.<br /><br />
  
== Errors Corrected ==
+
==Errors Corrected==
=== Issue 3019: Global group call: ===
+
===Issue 3019: Global group call:===
 
AlphaNet VoIP channel lock up when receiving GGC from AMC 09.xx<br />
 
AlphaNet VoIP channel lock up when receiving GGC from AMC 09.xx<br />
  
=== Issue 3021: Global group call, multi module/IP: ===
+
===Issue 3021: Global group call, multi module/IP:===
 
Build global group call (7879) was needed after reset <br />
 
Build global group call (7879) was needed after reset <br />
  
=== Issue 3022: Group call: ===
+
===Issue 3022: Group call:===
 
Multi module over IP. M-press during Gong was ignored  <br />
 
Multi module over IP. M-press during Gong was ignored  <br />
 
    
 
    
=== Issue 3023: Global group call: ===
+
===Issue 3023: Global group call:===
 
Multi module/IP; Reset when receiving GGC. <br />
 
Multi module/IP; Reset when receiving GGC. <br />
  
=== Claim 3219: Report EVH_CONV_OUTGO OFF: ===
+
===Claim 3219: Report EVH_CONV_OUTGO OFF:===
 
EVH_CONV_OUTGO_OFF is now reported if disconnect during connection tone.<br />
 
EVH_CONV_OUTGO_OFF is now reported if disconnect during connection tone.<br />
  
=== Issue 2881: AlphaWeb- AMC config backup page: ===
+
===Issue 2881: AlphaWeb- AMC config backup page:===
 
Restore backup: AMCD respond with "OK" to AlphaWeb _before_ reset, so AlphaWeb returns to backup page.<br />
 
Restore backup: AMCD respond with "OK" to AlphaWeb _before_ reset, so AlphaWeb returns to backup page.<br />
  
=== Issue 3025: Multi Module serial port: ===
+
===Issue 3025: Multi Module serial port:===
 
When the serial port is set for multi module the data message “SET MODE = 66" was not transmitted, thus the slave would never become a slave.<br />
 
When the serial port is set for multi module the data message “SET MODE = 66" was not transmitted, thus the slave would never become a slave.<br />
  
=== Issue 3027: Multi Module and SIP: ===
+
===Issue 3027: Multi Module and SIP:===
 
SIP was not working in combination with IP multi module.<br />
 
SIP was not working in combination with IP multi module.<br />
  
=== Issue 3030: RS485: ===
+
===Issue 3030: RS485:===
 
RS485 was not working because mismatch in configuration handling. <br />
 
RS485 was not working because mismatch in configuration handling. <br />
  
=== Issue 3016 AlphaNet Call Priority: ===
+
===Issue 3016 AlphaNet Call Priority:===
 
Station with Call Setup Priority = Alarm did not get through when all AlphaNet lines are busy due to conversation, global group call or SX conference.<br />
 
Station with Call Setup Priority = Alarm did not get through when all AlphaNet lines are busy due to conversation, global group call or SX conference.<br />
  
=== Issue 2907: AMC: Event - Related to UDP 8: ===
+
===Issue 2907: AMC: Event - Related to UDP 8:===
 
Events with Related to = UDP 8 was not triggered.<br />
 
Events with Related to = UDP 8 was not triggered.<br />
 
   
 
   
=== Issue 2995: Noise when VoIP call from DAK: ===
+
===Issue 2995: Noise when VoIP call from DAK:===
 
When making a VoIP call from a DAK programmed to do prefix + dirno (54 + 101), and DAK is kept pressed after the connection, loud noise was transmitted to the B-subscriber.<br />
 
When making a VoIP call from a DAK programmed to do prefix + dirno (54 + 101), and DAK is kept pressed after the connection, loud noise was transmitted to the B-subscriber.<br />
  
=== Issue 3009: Multi Module License use: ===
+
===Issue 3009: Multi Module License use:===
 
Multi Module automatically allocated all licenses not used for AlphaNet even in stand alone mode. This gave a confusing license overview in AlphaWeb<br />
 
Multi Module automatically allocated all licenses not used for AlphaNet even in stand alone mode. This gave a confusing license overview in AlphaWeb<br />
 
<br />
 
<br />
 
<br />
 
<br />
 
<br />
 
<br />
= AMC  10.00  (2006-06-01) =
+
=AMC  10.00  (2006-06-01)=
 
  Release: Official, available on request  
 
  Release: Official, available on request  
 
  /opt/amc/bin/amcd
 
  /opt/amc/bin/amcd
Line 822: Line 842:
 
patch_to_release_10_00.tbz2 must be installed on beta cards before alpha_sys_10_00.tbz2 is activated.<br /><br />
 
patch_to_release_10_00.tbz2 must be installed on beta cards before alpha_sys_10_00.tbz2 is activated.<br /><br />
  
== Errors Corrected ==
+
==Errors Corrected==
 
  See release note for beta software for error corrected during beta releases.
 
  See release note for beta software for error corrected during beta releases.
 
<br />
 
<br />
Line 832: Line 852:
 
<br />
 
<br />
 
==Known Issues==
 
==Known Issues==
=== Issue 3020: Missing “Exch. Missing” text: ===
+
===Issue 3020: Missing “Exch. Missing” text:===
 
When making a global group call and some exchanges do not receive the group call due to not available audio links, there should be a display message "Exchange Missing".<br />
 
When making a global group call and some exchanges do not receive the group call due to not available audio links, there should be a display message "Exchange Missing".<br />
=== Issue 3050: ODX from slaves via VoIP link: ===
+
===Issue 3050: ODX from slaves via VoIP link:===
 
ODX from stations in slave modules are not supported over VoIP link<br />
 
ODX from stations in slave modules are not supported over VoIP link<br />
=== Issue 3124: AlphaNet - NAT ===
+
===Issue 3124: AlphaNet - NAT===
 
AlphaNet does not work if NAT traverse is used.
 
AlphaNet does not work if NAT traverse is used.
  
Line 874: Line 894:
 
'''Version 01.06:''' 2007-12-10<br />
 
'''Version 01.06:''' 2007-12-10<br />
 
Description: <br>
 
Description: <br>
* Support of [[Line Echo Cancellation]] (by using DSP#2 for LEC, infrastructure added for controlling DSP#2)
+
 
 +
*Support of [[Line Echo Cancellation]] (by using DSP#2 for LEC, infrastructure added for controlling DSP#2)
  
 
'''Version 01.05:''' 2007-11-08<br />
 
'''Version 01.05:''' 2007-11-08<br />
 
Description: <br>
 
Description: <br>
* Optimised socket handling for unicast'ed groupcalls.
+
 
 +
*Optimised socket handling for unicast'ed groupcalls.
  
  
 
'''Version 01.04:''' 2007-10-17<br />
 
'''Version 01.04:''' 2007-10-17<br />
 
Description: <br>
 
Description: <br>
* G.729, first experimental support (no support for DTX, lost or reorderd packets).
+
 
 +
*G.729, first experimental support (no support for DTX, lost or reorderd packets).
  
 
'''Version 01.03:''' 2007-10-11<br />
 
'''Version 01.03:''' 2007-10-11<br />
 
Description: <br>
 
Description: <br>
* Issue 3269 Different UDP port on send/receive: Use sendto(), instead of connect()+write()
+
 
* txtap function to tap audio to an internal socket.  
+
*Issue 3269 Different UDP port on send/receive: Use sendto(), instead of connect()+write()
 +
*txtap function to tap audio to an internal socket.
  
 
'''Version 01.02:''' 2007-03-23<br />
 
'''Version 01.02:''' 2007-03-23<br />
 
Description: Released version.  <br />
 
Description: Released version.  <br />
 
Introduced in system upgrade file: alpha_sys_10_20.tbz2<br />
 
Introduced in system upgrade file: alpha_sys_10_20.tbz2<br />
* Jitterbuffer adjustments (Issue 3101 Xlite). Improve stabilty of delay adaptation, as well on adaptive delay target.
+
 
* Set IP TTL to 31 when connecting to UDP to multicast.
+
*Jitterbuffer adjustments (Issue 3101 Xlite). Improve stabilty of delay adaptation, as well on adaptive delay target.
 +
*Set IP TTL to 31 when connecting to UDP to multicast.
 +
 
 
'''Version 01.01:''' 2007-03-05<br />
 
'''Version 01.01:''' 2007-03-05<br />
 
Description:<br />
 
Description:<br />
* Fix issue 2935: Crash when ExpressTalk sends packet with zero payload at disconnect.
+
 
* Improved handling of termination signals with logging.  
+
*Fix issue 2935: Crash when ExpressTalk sends packet with zero payload at disconnect.
 +
*Improved handling of termination signals with logging.
 +
 
 
'''Version 01.00:''' 2006-05-31<br />
 
'''Version 01.00:''' 2006-05-31<br />
 
Description: Released version.  <br />
 
Description: Released version.  <br />
Line 908: Line 936:
  
 
'''Version 01.13 upcomming:''' <br>
 
'''Version 01.13 upcomming:''' <br>
* Send and receive INFOs for DTMF signals A - D
+
 
 +
*Send and receive INFOs for DTMF signals A - D
  
 
'''Version 01.12 (2008-01-03):''' <br>
 
'''Version 01.12 (2008-01-03):''' <br>
* Outgoing INVITE, early cancel: Cancel before "180 Ringing" caused lockup, because of incorrect check for "SIP-dialog".  
+
 
 +
*Outgoing INVITE, early cancel: Cancel before "180 Ringing" caused lockup, because of incorrect check for "SIP-dialog".
 
   
 
   
 
'''Version 01.11 (2007-12-10):''' <br>
 
'''Version 01.11 (2007-12-10):''' <br>
  
* Outgoing INVITE: Send AUDIO_PATH_STATE(TRYING) to AMCD immediately, allowing AMCD to handle cancel before first response from external SIP device.
+
*Outgoing INVITE: Send AUDIO_PATH_STATE(TRYING) to AMCD immediately, allowing AMCD to handle cancel before first response from external SIP device.
  
 
'''Version 01.10 (2007-10-12):''' <br>
 
'''Version 01.10 (2007-10-12):''' <br>
* CANCEL of outgoing INVITE: Send CANCEL, not BYE  
+
 
 +
*CANCEL of outgoing INVITE: Send CANCEL, not BYE
  
 
'''Version 01.09 (2007-09-18):''' <br>
 
'''Version 01.09 (2007-09-18):''' <br>
* Fixed bug in reINVITE handling as implemented in 01.08: Wrong RTP portnumber is used if port number in reINVITE is less than 4096 (0x1000).  
+
 
 +
*Fixed bug in reINVITE handling as implemented in 01.08: Wrong RTP portnumber is used if port number in reINVITE is less than 4096 (0x1000).
  
 
'''Version 01.08 (2007-08-30):''' <br>
 
'''Version 01.08 (2007-08-30):''' <br>
 
Only released in X-version 10.22 package
 
Only released in X-version 10.22 package
* Handle reINVITE which redirects RTP audio to different IP address and port.
+
 
* Fixed bug in parsing of AUDIO_LINK_OK from AMCD, which could be rejected erroneously
+
*Handle reINVITE which redirects RTP audio to different IP address and port.
 +
*Fixed bug in parsing of AUDIO_LINK_OK from AMCD, which could be rejected erroneously
  
 
'''Version 01.07 (2007-07-01):'''<br>
 
'''Version 01.07 (2007-07-01):'''<br>
 
Description: Released version, date 2007-06-01.<br>
 
Description: Released version, date 2007-06-01.<br>
 
Introduced in system upgrade file: alpha_sys_10_21x0604.tbz2<br>
 
Introduced in system upgrade file: alpha_sys_10_21x0604.tbz2<br>
* Removed memory leaks, increasing stability (issue 3176).
+
 
* Issues 3203, 3182.  
+
*Removed memory leaks, increasing stability (issue 3176).
* Debug error messages forwarded to syslog.  
+
*Issues 3203, 3182.
* Set scheduling priority.
+
*Debug error messages forwarded to syslog.
 +
*Set scheduling priority.
  
 
'''Version 01.05:'''<br>
 
'''Version 01.05:'''<br>
Line 948: Line 982:
  
 
'''Version 02.01''' (2007-12-10): <br>
 
'''Version 02.01''' (2007-12-10): <br>
 +
 
*Required for [[Line Echo Cancellation]] ("disable" command, used to free DSP power for LEC).
 
*Required for [[Line Echo Cancellation]] ("disable" command, used to free DSP power for LEC).
  
Line 955: Line 990:
  
 
'''Version 01.10''' (2008-01-16): <br>
 
'''Version 01.10''' (2008-01-16): <br>
* Same as 02.10, but for 2.4 linux kernel: Support for DSP_SW version 01.10
+
 
 +
*Same as 02.10, but for 2.4 linux kernel: Support for DSP_SW version 01.10
  
 
'''Version 01.01''' (2007-12-10): <br>
 
'''Version 01.01''' (2007-12-10): <br>
* Same as 02.01, but for 2.4 linux kernel
+
 
 +
*Same as 02.01, but for 2.4 linux kernel
  
 
'''Version 01.00:  Board support package 02.xx (Linux 2.4):'''<br />
 
'''Version 01.00:  Board support package 02.xx (Linux 2.4):'''<br />
Line 993: Line 1,030:
 
'''Version 01.04:''' 2007-10-17<br />
 
'''Version 01.04:''' 2007-10-17<br />
 
Description:  <br />
 
Description:  <br />
* 16 bit linear PCM at 8Hz support, which is required for the G.729 support in rtpdaemon 01.04.
+
 
 +
*16 bit linear PCM at 8Hz support, which is required for the G.729 support in rtpdaemon 01.04.
  
 
'''Version 01.03:''' 2007-03-05<br />
 
'''Version 01.03:''' 2007-03-05<br />
 
Description: Released version.  <br />
 
Description: Released version.  <br />
 
Introduced in system upgrade file: alpha_sys_10_21.tbz2<br />
 
Introduced in system upgrade file: alpha_sys_10_21.tbz2<br />
* Improved mixing units to support DTMF tones: 32 mixers, independent mixers(function=1 in "con" of input)
+
 
 +
*Improved mixing units to support DTMF tones: 32 mixers, independent mixers(function=1 in "con" of input)
  
 
'''Version 01.02:''' 2006-11-30<br />
 
'''Version 01.02:''' 2006-11-30<br />
 
Description:  <br />
 
Description:  <br />
* DC-reduction filter on signals from backplane (to IP).(First order high pass IIR filter: timeconstant T ca 4ms, cutoff frequency ca 40 Hz)
+
 
 +
*DC-reduction filter on signals from backplane (to IP).(First order high pass IIR filter: timeconstant T ca 4ms, cutoff frequency ca 40 Hz)
  
 
'''Version 01.01:''' 2006-06-14<br />
 
'''Version 01.01:''' 2006-06-14<br />
Line 1,046: Line 1,086:
 
==AMC hardware versions==
 
==AMC hardware versions==
  
=== Known problems AMC hardware 8000/4 ===  
+
===Known problems AMC hardware 8000/4===  
 
----
 
----
 
None as of now<br /><br />
 
None as of now<br /><br />
  
=== Known problems AMC hardware 8000/2 ===  
+
===Known problems AMC hardware 8000/2===  
 
----
 
----
=== Issue 2747: RCI not supported on ACE7: ===
+
===Issue 2747: RCI not supported on ACE7:===
 
RCI not supported on ACE7. The RCI signals on P1-c19 and P1-c22 are terminated in test points on AMC-IP. <br />
 
RCI not supported on ACE7. The RCI signals on P1-c19 and P1-c22 are terminated in test points on AMC-IP. <br />
  
=== Issue 2741: Redundancy control from APC: ===
+
===Issue 2741: Redundancy control from APC:===
 
The redundancy control system from APC is not working (software and hardware).<br />
 
The redundancy control system from APC is not working (software and hardware).<br />
  
=== Issue 2787: AMC serial port: No RX, no data on TX: ===
+
===Issue 2787: AMC serial port: No RX, no data on TX:===
 
It turns out that the RS232-drivers on the AMC-IP-board have an automatic shutdown when it detects missing received data (illegal voltage levels). This is fixed in hardware version 8000/4, but it can be fixed on older hardware versions by a [[No data on serial ports|minor modification]]. <br />
 
It turns out that the RS232-drivers on the AMC-IP-board have an automatic shutdown when it detects missing received data (illegal voltage levels). This is fixed in hardware version 8000/4, but it can be fixed on older hardware versions by a [[No data on serial ports|minor modification]]. <br />
  
=== Issue 2806: “Temperature Alarm" in AlphaCom E7: ===
+
===Issue 2806: “Temperature Alarm" in AlphaCom E7:===
 
This problem is probably related to the fact that the AMC-IP board never had a connection to the over-temp. signal from the ACE7 backplane. This is related to the problems with RCIs from the same backplane.
 
This problem is probably related to the fact that the AMC-IP board never had a connection to the over-temp. signal from the ACE7 backplane. This is related to the problems with RCIs from the same backplane.
  
 
==AMC Filter board==
 
==AMC Filter board==
  
=== Issue 2723: RS422/RS485 Signal Pinning: ===
+
===Issue 2723: RS422/RS485 Signal Pinning:===
 
The pin out for RS422 signals on the filter print for E20 and E26 differs from the pin out on the E7. The Rx+ is switched with the Rx- and the Tx+ is switched with the Tx-. The only consequence is that the same cable can't be used on E7 and the E20-series. From filter print version 3 (DB8001/3) the mapping is correct.<br />
 
The pin out for RS422 signals on the filter print for E20 and E26 differs from the pin out on the E7. The Rx+ is switched with the Rx- and the Tx+ is switched with the Tx-. The only consequence is that the same cable can't be used on E7 and the E20-series. From filter print version 3 (DB8001/3) the mapping is correct.<br />

Latest revision as of 05:42, 8 March 2022

Previous Release - AlphaCom 9.xx - Release Notes

Next Release - AlphaCom 10.4x - Release Notes

This document provides the release notes for AlphaCom 10 with incremental bug fix releases. The release notes for AlphaCom 10 describe new features, improvements and issues fixed after AlphaCom 9.

Software in production: AMC 10.20
Software released date: 2007-04-12
Note 1: We sometimes do bug fixes in older versions while working with a new version. You can’t read the list version by version, and always assume that a correction is included in the next. Be aware of high release numbers, e.g. 06.05 vs. 07.01. Check version dates, and also comments for each version.
Note 2: For each software version the NVRAM version is listed. If the NVRAM version is different, the AMC board must be cold started, and then you must do a SendAll from AlphaPro to restore the configuration.


Contents

AlphaCom 10.xx Release Notes


AMC 10.31X303 (2008-03-03)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 10.30. 

System upgrade file:
alpha_sys_10_31x0303.tbz2

Precautions:
alpha_sys_10_00.tbz2 must be installed.

AlphaPro 1028 or later must be used due to WACS crash caused by to large nvram size when using older versions.


Errors Corrected

Issue 3213 Reset when 7820 backup

Backup could crash due to overuse of system resources. The backup/restore function is optimized.

Issue 3274 and 3320 IP Program distribution and mixed Multi Module with E7/E20/E26

Combination of E7 / E20 / E26 program distribution will only work with prog.dist over IP.
Remember to open the RTP voip ports in AlphaWeb IP filters !

Issue 3278 Missing first digit in display when node number is 15

Low level function bug related to node numbers ending in 0x0F (HEX), 15,31,47, 63 etc.

Issue 3324 $ALRM to IP station

Assert when disconnecting $ALRM to IP stations.

Issue 3338 Time setting "785" and AlphaPro

Local time setting is now changing local time and not UTC time.

Issue 2758 Serial communication via Eth

Using 485 multidrop protocol can cause communication errors both on serial adapters and when connected directly to the AMC serial port.

Functional changes / Enhancement

Treble preemphasis adustment of all IP audio

IP audio connections in AlphaNet are not compatible with earlier versions of AMCD SW. This is because treble preemphasis of AlphaCom audio is now removed when converted to IP/RTP. This change is done in order to improve audio compatibility with IP stations and third party devices. When calling between AlphaCom with old and new SW, audio is connected, but the treble will be too low in one direction, and too high i the opposite direction. Beside not sounding right, the voice switched duplex failes to work properly, requiring extensive use of M-key.

External VoIP audio gain adjustment

IP audio Gain adjustment options. Gain towards SIP devices is now by default +18db, giving more appropriate audio level towards external exquipment.

Multicast reachability test

  • IP station multicast reachability tests (mping), with reporting to syslog.
  • IP station not receiving RTP audio after connect, error forwarded to syslog.

Fader support in IP only system

Use AMC-IP FPGA for Fader Resource. Allowing Groupcall and Simplex conference in "IP only" system.

Extended SIP-DTMP signaling

  • Outgoing call to SIP: DAK 2 - 5 sends DTMF signals A - D.

Pager support for 8 Bit Ascom ESPA protocol

Support Norwegian character to paging. Ascom Norwegian Characters

OPC server license support

License for connection of OPC server supported.

Global Conference Lockup Situations

Bug fix related to global conference, node reset, default speaker conflicts and resource lockup.

DECT station lockup

Robust handling of call termination to avoid lockup of ATLB stations.

Improved update of Watchdog

Reset due to false watchdog timeout fixed.

AMC 10.30 (2008-01-17)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 10.30. 

System upgrade file:
alpha_sys_10_30.tbz2

Precautions:
alpha_sys_10_00.tbz2 must be installed.

AlphaPro 1028 or later must be used due to WACS crash caused by to large nvram size when using older versions.


Errors Corrected

Issue 3052: $TPROG causes AMC reset:

$TPROG L%1.d U1 is used in the standard PrisCom database when the key returns to "0" position. Fixed(X10.23(1127))

Issue 3108: SIP trunk, short number not possible:

Short number usage
With PNCI you can use the "Phone" function to define 3- or 4 -digit short numbers for easy access to the most frequently used phone numbers. You could also do 71 transfer to short numbers. A short number should show up in the directory list of the display.
Short number system for SIP is now implemented.(X10.23(1127)) (AlphaPro is currently (1028) not supporting the configuration)

Issue 3201 SIP stations currently used

When configuring new SIP stations the license use is now updated without the need of a reset.

Issue 3206: Serial Communication Problems (1021)

Reset of AlphaCom in relations to Serial Port use. Serial port buffer made better.(X10.23(1127))

Issue 3235: Audio Event In AlphaNet

When calling a station in a different node, event type audio and always programmed in the called node is not triggered. This was due to the move of duplex from end node to start node. The default behavior of IP AlphaNet is now back to end node duplex.(X10.23(1127))

Issue 3242: SIP RE-INVITE

SIP implementation now does support RE-INVITE(X10.23(1127))

Issue 3269 SIP rtp audio on different UDP ports

Some SIP equipment sends RTP packets from different UDP port than the one they are receiving on. This is now supported by AlphaCom.

Issue 3270 SIP Use CANCEL, not BYE

After a ringing timeout the AlphaCom terminates the call with SIP command ‘BYE’ instead ‘CANCEL’, fixed.

Issue 3277 CR notification is missing/changed

Some Issues with Call request notification after ringing group implementation fixed.

Issue 3279 Event Handler discon group call

If group call is disconnected in event handler "8 Conversation Outgoing" related to the same group call in some situations the AlphaCom asserts. This is fixed.

Issue 3280 SIP registrar number starting with "0"

Fix of SIP reqistrar problem when numbers starting with Zero.

Issue 3281 ESPA444 stops

Protocol issues with the ESPA444 can get the pager protocol in a dead locked state. Fixed.

Issue 3283 Syslog report of "tst" warnings missing.

System Warning is now also reported to Syslog,

Issue 3284 Ringing Grp from handset lifted station

A call to a ringing group from a station with lifted handset will lockup exchange on acceptance. Exchange needs to be reset, fixed.

Issue 3285 Ringing group call back

The M-key needs to be pressed on the calling station if the calling station is in private. Fixed.

Issue 3301 Crash when Call Req to IP station

Call request to IP stations will generate a reset of the exchange, fixed.

Issue 3302 Event 10: %2.dir missing when SIP

Mail events related to ringing group fixed.

Issue 3316 Global grp call w IP station - reset

Reset problems related to IP station group calls fixed.

IP DECT hanging when canceling from AlphaCom

Problems of canceling early in audio setup from AlphaCom to Asacom IP DECT is solved

Search to Call Request

Fixes of preforming a Call Request in a search string when caller is located in a remote exchange.

Full E26 rack problems

Add workaround of faulty read of own board type when full E26 rack.

Pocket paging faults

Use of Alarm priority for pocket paging could reset the exchange.
Bleep priority 0 is now legal to use. RedBoot Fix

Issue 3268: Serial data in/out during boot

The serial data transmitted during reset is removed.

Functional changes / Enhancement

Hostname support

Hostname can be stored for SIP destinations instead of ip address.

Mutual exclusion groups increased

Mutual exclusion groups increased from 4 to 8

Number of groupcalls increased

The number of groupcalls is increased from 100 to 250 (No AlphaPro support 1028)

Local Echo Canceling

LEC
10 channels of local echo canceling are available for "open" handset conversation when using SIP, Multimodule IP and AlphaNetIP.
The channels are allocated when needed, when no more resourced switched duplex is used.

Faster synchronizing of database to flash

The speed of synchronizing the configuration database to flash after "SendAll", dak programming etc is increased.

Pocket paging "meet me" during conversation

Feature 25, Pocket paging meet me, alarm priority added to the autoload as dirno "48".

Outgoing conversation related to SIP

Private Ringing Outgoing
Event type 33, used for state of outgoing conversation to SIP.

Event handler %1.pag

New event handler operator %x.pag (x = 1 or 2) added for getting pager related to a station.

New DP command $VOL

VOL
New DP command $VOL, adjust station volume 0-9.

Restriction of syslog reporting

Syslog system and debug reports are restricted to 60 messages each hour to prevent overload because of faulty configuration or other active syslogging problems.
The limit can be adjusted from TST:

TST>>nvram - ex_profile.glob_const.syslog_block = 6 (6 = 60 messages pr hour)

Enhancement Issue 3266 Selective dialing

Short number usage
When calling in to the AlphaCom via SIP gateway, it is be possible to automatically connect to an AlphaCom feature which makes it possible to do selective dialing. If no dialing within a preset time, the call is forwarded to a predefined number. The short number configuration system is used (Currently not available in AlphaPro 1028)

Enhancement Issue 3272 SIP - DAK8, OnHold and Transfer

Transparent mode DAK configuration
Key used for activating transparent modus during SIP calls can now be change in NVRAM.

TST>>nvram - ex_profile.glob_const.trans_mode_dak = 8 (Default key is 8.)

Enhancement Issue 3275 Group call without M-key

Automatic M-key in group call
A group call can be configured to work as automatic "M-key" when initiated

TST>>nvram -  ex_profile.group[G].flags = F

G

  • group number 1 > 250

F

  • 0 = default (press M key)
  • 1 = automatic M-key from all stations
  • 2 = automatic M-key only from SIP calls.

Enhancement Issue 3304 Increase AlphaNet data links

The number of AlphaNet data links is increased from 20 to 50.

AMC 10.22 (2007-09-05)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 10.20. 

System upgrade file:
alpha_sys_10_22.tbz2

Precautions:
alpha_sys_10_00.tbz2 must be installed.

Errors Corrected

Issue 3149: Follow Me(72) and User/Phy number

There was a mix of user and physical number in the follow me function. Stations with different physical and user number would not have correct follow me behavior.

Issue 3157: Call Request Transfer to Phone

Call request forwarding is solved by the new feature Call request forwarding by search strings

Issue 3210: AlphaNet- CRM4 - display call request

The calling party’s number is not displayed when answering the call request (or MST mail) from a CRM IV type station in another node (AlphaNet). Only the name is shown. The error is only when the answering station has the Station Type set to "CRM 3&4" in software. Fixed.

Issue 3214: $CANM with source as global number

The $CANM can now use global number as source without adding the specific node number (use L2345 in stead of L(2)2345)

Issue 3225: IP substation: Call Req Mode - LED

If call requester mode is removed from an IP station the Audio to the IP station is now terminated.

Issue 3229: Formating of EXP not OK

Formating of %1.EXP is not working properly

Issue 3231: Statistic Log on IP station

Calls to IP stations in now logged in the SysLog statistics

Issue 3233: No Line error if Hot-line enabled

The flag ignore_st_down_in_conv allow stations to fail during conversation. For hot line station it means that if hot line is trigged during station down it will not be reported. Fixed by setting a short timeout on how long station can be down when active

Issue 3238: Hanging audio links to SIP/RMD

When initiating a call to SIP/RMD from AlphaCom and terminating during setup tone the Audio connection is not disconnected. Fixed.

Issue 3240:Ringing group - Loose CR mode when called

When in call requester mode, you can be called by anybody, which will take you out of call requester mode. It will be confusing if you expect your call to be answered and all of a sudden somebody else gets through. There should be a mechanism that when in call requester mode you can only be called by a station which has you in the queue. This is fixed by setting the caller in busy state and use automatic call override when answering from a ringing group.

Issue 3247/3252: AlphaCom uses duplex algorithm when conversation with Ring Master

The duplex algorithm is no longer used on AlphaCom side when calling/called by Ring Master.

SysLog

In some cases syslog reporting from AlphaCom was not correctly initiated.

Functional changes / Enhancement

DTMF tones during conversation

Feature for dialing DTMF tones during conversation DTMF feature (107) works for local conversation or towards SIP/RMD. Can Also be used as Data Message DTMF_CONN

Event trigger during conversation

New feature Event Trigger During Connection (108).

Parallel Ringing

Add stations in Parallel Ringing when station is in private ringing mode

Ringing Group

New behavior of call request. Private ringing tone on receiving station(s). Ringing Group (109)

  • SEND_MAIL added new flag for "auto delete" functionality and optional ringing tone in use with ringing group.
  • CANCEL_MAIL allowed LV in destination field

Global Conference

New fixes and adjustments

  • Regular interval Root of conference broadcast message was not sent in all situation. (A safety message for reconnecting if links have been down etc)
  • The interval of Root of conference broadcast message is now decreased from 1 hour to 1 minute. It means reconnecting of missing nodes and cleaning of unused links will be much faster after certain network/reset problems.
  • Useless single AlphaNet audio links to root nodes with no conference members are cleared.
  • When Root of conference is AMC 8/9 and there is an IP transit node with members of the conference, leaf IP nodes was not able to connect to the conference due to CODEC selection failure.
  • Better terminating algorithm when a conference only consist of default members (in several nodes).

HTTPS support added

Gives the possibility to access AlphaWeb over a secure connection. See AlphaWeb Technical

AlphaWeb Custom Scripting

AlphaWeb Custom Scripts lets the end user extend the AlphaWeb functionality. Typically this can be 'Click to Call' type of applications.

AMC 10.21 (2007-06-08)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 10.20. 

System upgrade file:
alpha_sys_10_21.tbz2

Precautions:
alpha_sys_10_00.tbz2 must be installed.

Error Corrected

Issue 3193: IP station -> SIP: codec problems

When an IP Substation is calling a SIP Gateway or SIP Phone, the codec of the IP station must be set to G711, else there is no audio. But when the IP substation codec is G711 there will be a "click" sound in the IP substation on regular intercom calls. Fixed.

New FPGA software v 01.68

FPGA software improvement of audio in full module exchanges.

AMC X10.21 (2007-06-04)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 10.20. 

System upgrade file:
alpha_sys_10_21x0604.tbz2
(Includes sipd version 01.07)

Precautions:
alpha_sys_10_00.tbz2 must be installed.


Errors Corrected

Issue 2899: $SM don't trigger RelTo = UDP

Intelligence added to the code receiving $SM (DP). If UDP group not specified, and mail sender is local, look up the UDP group from NVRAM. Can not be done if mail sender is in a different node. %1.udp macro added to the EventHandler, so a complete $SM can be sent in the first place.

Issue 2952: Dual Display MDF text corrections

Some display text cleanup regarding during mail sending.

Issue 3142: Line errors at startup

At startup there could be line errors reported at non existing physical numbers. Corrected

Issue 3161: Event handler %scutf, comma

AMCD Event handler: The %scutf macro now works properly when using comma as delimiter.

Issue 3163: Conversation incoming second user.

Conversation incoming [07] at B-sub (e.g.2001) which is a second user (not default) to phys. 10 (2000) does not work. The event handler is now updated so that incoming event %dir and %name reflect the actual user dialed.

Issue 3164: Call request from data protocol (from other than default user)

If you simulate by data a call request (sender e.g.2001) to CRM4 2500 the call request message is ok (sender 2001). You are able to call back (locally node2) with 70+8 or 7638 and you are connected to the default user (2000 FBSAR), but the call request message (initiated from 2001, not default) is not removed. If you call back by manual dialing 2001, the message is removed. Correct directory number (other than default user) is now used when canceling mail.

Issue 3176: SIP: Gateway blocked, reset required

Reports of reset of AlphaCom required due to blocking of audio lines to Mediatrix ISDN gateway. Fixed

Issue 3177: Eventhandler fault when related to > 4 digits

Event handler actions “Related to” more than 4 digits is now working.

Issue 3182: SIP ringing terminates after 30 sec.

The SIP ringing length is now following the AlphaPro configuration.

Issue 3188: Off Hook duplex when calling SIP

Handset operation /duplex flag is now tested during SIP setup, thus give DTMF signaling from ATLB stations.

Issue 3195: Intermodule IP audio lockup slave reset

Improvment to slave module audio availablility during slave reset.

Issue 3196: $ST Lxxx W0 don’t stop timer

Data command “$ST Lxxx W0” will now stop all timers running on station xxx.

Issue 3203: SIP: Handytone INFO digits ignored workaround

Workaround for problem with Grandstream Handytone 488, firmware 1.0.3.64. Ingoing call to Alphacom, then a lot of keying, for excample of * and # for simplex / groupcall keying. Then after some 10 - 20 incomming INFO messages, SIPD suddenly ignores a INFO message. This in turn locks up Grandstream, so that it does not send a BYE when the handset is replaced.

Issue 3205: Line test 7872 in multi module

The 7872 line test is now working again for Multi module AGA/AE1

Issue 3215: Busy/Private override in AlphaNet

Busy and private override is now working over AlphaNet using M-key.

Issue 3218: Mail related assert:

Improved clean-up system for AlphaCom mail elements.

Issue 3217: Call from SIP to IP station

When calling from SIP G711 codec to IP station G722 codec, transcoding is now correctly initiated.

AMC 10.20 (2007-04-12)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 10.20. 

System upgrade file:
alpha_sys_10_20.tbz2

Precautions:
alpha_sys_10_00.tbz2 must be installed.

Functional changes / Enhancement

Private ringing tone on Call Request function

Private ringing tone on mail at receiver station will be active when first mail in queue has the mail priority above 150 (default value of "globel_constant>priv_ring_mail_pri")

New event FEAT_M_KEY (31 )

ON = M key pres
OFF = M key release
Sub event 0 will give M key status when station is busy.
Sub event 1 will give M key status when station is idle.

New event FEAT_OFF_HOOK (32)

ON = OFF_HOOK
OFF = ON_HOOK
Sub event 0 will give HOOK state when station is busy.
Sub event 1 will give HOOK state when station is idle.
Only state changes within current state is reported on M_KEY and HOOK, it means if the station goes OFF_HOOK from idle the event OFF_HOOK in busy state will not be reported.

Errors Corrected

Issue 3114: Multi Module IP audio fails after slave reset:

When slave is reset, without the master being reset, the IP ICC audio links does now work

RTP jitter buffer improvements

Rtpdeamon version 01.02 improves the jitter buffer handling. Both handle larger difference in clock rate between sender and receiver, at the same more stable delay adaptation at the presence of jitter. Fixes issue 3101 with Xlite.



AMC X10.20 (2007-03-20)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 10.20. 

System upgrade file:
alpha_sys_10_20x0320.tbz2

Precautions:
alpha_sys_10_00.tbz2 must be installed.

Errors Corrected

Issue 3179: Calling from a slave to an E1 node:

It was not possible to dial global numbers from a slave module in multi module over IP to a different node connected with the master over AE1. (10.20x bug)



AMC X10.20 (2007-03-15)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 10.20. 

System upgrade file:
alpha_sys_10_20x0315.tbz2

Precautions:
alpha_sys_10_00.tbz2 must be installed.

Functional changes / Enhancement

Ring Master Daemon is included in the package.

See separate documentation.

Support of IP stations.

  • - Substation functionality
  • - Group Call is available
  • - Outputs on the stations can be related to RCO in AlphaPro. (RCO type station)
  • - Inputs can be related to DAK or directly to DAK_AS_RCI event in AlphaPro.

See separate documentation.

IP audio transit capacity is increased from 32 to 64 channels.

Transit audio is audio not going trough the backplane. (SIP to SIP, IP station to IP station, AlphaNet IP transit or a combination of IP audio)

AlphaWeb – AlphaNet information
Information page now contains node name and node software versions.

Errors Corrected

Issue 2885: Log enabling:

System log to Syslog is now working even when log port is not enabled in AlphaPro.

Issue 2935: SIP Softphone ExpressTalk causes AMC reset:

SIP Softphone ExpressTalk would reset the exchange when the conversation is cancelled from the phone (both incoming and outgoing calls).

Issue 3053: Conversation Outgoing event:

When making a call from intercom out on a SIP gateway (Mediatrix 2400) the event Conversation Outgoing is now triggered when the phone answers.

Issue 3154: $CALL in AlphaNet don't work:

The data protocol command $CALL is now working in AlphaNet.

Issue 3169: Serial communication block AMCD:

AMC would not start after power reset if serial communication was active during reset. (Other node sending data to resetting AMC) This will not happen on system running Linux 2.4.

Issue 3180: Program 7 has no display txt:

Audio program 7 (feat 5/7): The display text was not shown in the display when activating the audio program 7. For other programs it is ok. (I have seen this long time ago also, so it might be an old bug). Static License of 2 AlphaNet audio links does not work correctly. Only one audio link was working. Problems with SIP and Codec Selection
SIP Preferred codec selection configured in AlphaPro does now also filter on incoming audio stream. This cure some problems regarding codec mismatch between AlphaCom and SIP.



AMC 10.05 (2007-02-07)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 10.00. 

System upgrade file:
alpha_sys_10_05.tbz2

Precautions:
alpha_sys_10_00.tbz2 must be installed.

Functional changes / Enhancement

License keys:

SIP trunk and SIP stations is now separated in two different licenses.
SIP trunk and AlphaNet licenses is by default dynamic except for the 2 line AlphaNet license that still is static.
No audio routing programming is needed for SIP trunk or AlphaNet IP (except for 2 licenses).
AlphaNet and Multi Module IP use dynamic licenses from the same license pool.
If static routing is programmed in AlphaPro licenses for those audio lines is reserved from the license pool and can not be used for other dynamic audio links/multi module.
Priority of audio resources allocation is handled by the priority of the initiator station.

Errors Corrected

Issue 2715: AlphaNet Duplex in combination with AMC 8/9:

In combination AlphaNet with AlphaCom 8 and 9 and exchanges via IP there will be a problem with delay adjustment of the duplex algorithm when calls are made from IP to an AMC 8/9 exchange.
The duplex algorithm is now run in A node when going from an ASLT station to IP AlphaNet thus avoiding duplex in end AMC 8/9 nodes.
In transit systems with certain combinations of AMC- 8/9/IP, SIP, AGA, AE1 and Multi module IP some issues could still occur that needs special configuration. (See separate duplex document).

Issue 2887/3113: SIP automatic duplex switching:

When calling SIP there can be problems with the standard duplex algorithm due to DSP echo cancelling in the SIP station. A new duplex algorithm is available for duplex towards SIP stations that is speech controlled only from volume of the microphone signal from the SIP link.
(exchange flag “DSP_duplex = 1”)

Issue 2897: X-lite on-hold lockup:

Problems during on-hold feature in X-lite fixed.

Issue 2908: SIP-Ringing if call cancelled before answer:

Calling from AMC-station to SIP phone. If C-key is pressed on AlphaCom-station before the call is answered, the SIP phone keeps ringing if calling via a transit node.

Issue 2960/3119: SIP-Handy-Tone 488 making All Call.(* and # key):

SIP now handles * and # for both Grandstream and Mediatrix.

Issue 3011: Duplex switching in mixed environment:

When the audio path go transit from IP to analog/E1 delay information to the automatic duplex routine is lost. Duplex delay is now forwarder/backwarded from transit AGA to IP links to the duplex node.

Issue 3012: Echo in SIP handset:

When talking with handset conversation between Intergard station and SIP station the SIP station will get echo in the handset due to overhearing in AlphaCom handset (and no echo cancelling). Handset to handset communication will now be forced in duplex. (Default delay setting for SIP = 30ms. Full duplex can be obtained if parameter “max_off_hook_delay” is adjusted to 40 ms or more)

Issue 3015: AlphaNet: No Camp On Busy:

There is no "camp on busy" when all AlphaNet lines are in use. Instead one gets a rejection tone. Same behaviour with AGA line, AE1 and VoIP. Feature implemented.

Issue 3017: Name list:

After AMC auto load dirno's 9542 - 9545 are in the name list (614).

Issue 3034: Time management:

When changing time in AlphaWeb, the new time is now also written to the hardware clock.

Issue 3051: IP address with leading 0's:

Interpretation of leading 0 in AlphaWeb is fixed.

Issue 3067: AlphaWeb: Same subnet on Eth0 and 1:

Configuration of both Ethernet phys on the same sub net is now tested.

Issue 3070: SIP: Long phone no on DAK/Substation:

Up to 16 digit phone number now allowed in DAK "784" and Call-forward "71".

Issue 3116: AMC: Speech channel locks up:

During global conference and failed SIP calls speech channels could be locked up.

Issue 3143: AMC: 99 answer in global group call:

Not possible to answer global group calls from other that initiating node. This feature is fixed.

Issue 3060: SIP & AlphaNet licenses:

It is now possible to install 30 AlphaNet and 20 SIP trunk licenses.

Issue 3069: SIP: Phone number in display:

For outgoing/incoming calls the number of shown digits is 16 (including event handler).

Issue 3087: SIP: Mediatrix 1204 - Dial Out:

Transmit digit by digit as dialled now supports:

  • - Programming of phone number on DAK from AlphaPro
  • - DAK, "784" from station
  • - Call forwarding from station "71"

Both for Mediatrix and AudioCodes

Issue 3105: SIP Trunk: DAK call fails (AudioCodes):

The first or the two first DTMF digits of the phone number is not transmitted. 400 ms delay before sending digits. Delay can be extended with exchange timeout "sip_dial_dly".

Issue 3107: SIP Trunk: Call Forward (71):

Manual transfer to phone using 71 don't work (PNCI you could 71 + 0 + phone + M, or 71 + <shortnumber>). Fixed

Issue 3136: AlphaNet: Global SX Conference:

AlphaNet: Global SX Conference. Problems after node reset and issues of reconnect are fixed.

Issue 3141: Call to unregistered SIP stations:

Call from AlphaCom to a SIP phone, SIP phone is configured at registrar node, but the phone is not registered: Then SIP sends the INVITE to its own IP address, which is processed, and forwarded in a loop until all RTP resources are used up. IP address check implemented.

Issue 3156: AlphaWeb show no licence:

AMC now generate correct infor in the license info file.



AMC X10.05 (2007-01-18)

Release: Beta, available on request
/opt/amc/bin/amcd
NVRAM version 10.00. 

System upgrade file:
alpha_sys_10_05x0118.tbz2

Precautions:
alpha_sys_10_00.tbz2 must be installed.

Errors Corrected

Issue 3072: $CPYM removes two char in name:

When a mail is copied to another station using $CPYM L%1.dir W%2.tag L<dirno>, the two first characters in the name of the sender is removed. E.g. if the sender is "Donald Duck" it will appear as "nald Duck".

Issue 3074: 626 cancel call request:

The problem is related to 626. This code blocks the station for a few seconds when you hang up.

Issue 3088: Transfer of outgoing calls:

When doing an outgoing phone call from AlphaCom via SIP Gateway (tested with Mediatrix 1204 (analogue) and Mediatrix ISDN) you cannot transfer the call to another intercom station.

Issue 3115: AlphaNet Global SX Conference:

Cancelling and reinitiate a global conference do not distribute audio. Also problems when resetting member nodes of a conference.



AMC X10.05 (2006-12-21)

Release: Beta, available on request
/opt/amc/bin/amcd
NVRAM version 10.00. 

System upgrade file:
alpha_sys_10_05x1221.tbz2

Precautions:
alpha_sys_10_00.tbz2 must be installed.

Errors Corrected

Issue 3111: Multi Module Group Call Block:

MultiModule and group call could block. Fixed one reproduced case.

Issue 3112: Reset after cancel of call setup by the $CALL command:

When resetting A-station in a call established by the $CALL the exchange did a reset. Fixed

Issue 3121: In hotline call off-hook action do not work:

Fixed.



AMC 10.04 (2006-12-12)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 10.00. 

System upgrade file:
alpha_sys_10_04.tbz2

Precautions:
alpha_sys_10_00.tbz2 must be installed.

Functional changes / Enhancement

AlphaPro Password:

The AlphaPro password is now the same as the administrator password for AlphaWeb (default user: admin, password alphaadmin)

Issue 3031: Ignor station down as default auto load:

The flag is enalbled as default from AMC 10.04

== Issue 3068: SIP: Inter-digit timeout: ===
TST->Nvram “ex_profile.timeouts.dig_col_timeout” configures the inter digit timeout in 100 ms steps during collection of digits on feature 81 and 83. Default is 3 seconds.

Errors Corrected

Issue 3024/3039: AlphaNet & MultiModule IP - 7872 causes reset:

MultiModule over IP: Dialling 7872 on station 101 in master causes reset.

Issue 3038: EDIO - %sscan:

The macro %sscan(string, search-string) returned wrong result when there was no match with the search-string.

Issue 3076/3085: Special characters in AlphaCom display text:

AlphaCom display text could not include /-,. æøå etc. when making an SIP call.

Issue 3086: TST Error messages during SIP call:

Error messages introduced in 10.03 during call SIP reset is now fixed.

Issue 3089: Events when not "Default User”:

The eventhandler is now working for Conversation event “related to” user that are not default users.

Issue 3091: Mediatrix 1204 - No M-key */#:

Key signaling of * and # from mediatrix is now implemented. Signaling of * and # out of SIPD is now also using * and # towards SIP equipment

Issue 3093: RCO PULS bug:

It was possible to activate the RCO puls system by accident from the event handler scripts due to error in the script parser. (The RCO system then got inverted behavior)

Issue 3095: Context params in built in cmds, RCO:

The “%chg(on,off)” command did not work correctly for rco.

Issue 3096: Mediatrix 1204, direct dial out:

SIPD is now hanling collected digits from AlphaCom.

Issue 3098: Multi Module Group call on AGA lines:

Group Call with AMC IP master and AGA/AE1 audio could reset the exchange.

Issue 3103: Eventhandler; left adjustment of %:

Left adjustment of %macroes did not work. Example: "%1.exp(3<)"



AMC 10.03 (2006-20-11)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 10.00. 

System upgrade file:
alpha_sys_10_03.tbz2 This package has support of both board support package 02.xx (Linux 2.4) and 03.xx(Linux 2.6)

Precautions:
alpha_sys_10_00.tbz2 must be installed.

Functional changes

Pulse RCO:

Support for generating a pulse of specified length on a “logical” RCO.
New parameter “Duration” is added to the event handler-built-in command “RCO” and the data protocol message $SLRC. Examples, pulse RCO # 13 on for 1 second: “rco 13 on 10” or “$SLRC W13 1 w10”.
Duration is in tenths of second. If Duration parameter is present, and non-zero, the RCO state specified in previous parameter lasts for the specified time. After the time has expired, the RCO is toggled to the opposite state. If a new $SLRC/RCO on the same RCO arrives while the pulse timer is running, the timer is cancelled, and the new message determines the new RCO state. Duration = 0 means infinite duration.

DAK RCI EVENT:

New event Dak key as RCI (30). Sub event = DAK key number. ON = Dak key press, OFF = dak key release.

Private ringing for incomming SIP calls:

New tst nvram flag available ”ex_profile.flags.private_ringing_SIP”. If ”1” then calls from SIP will use private ringing mode. If “0” then calls will connect as an intercom call with normal priority.

Station Bit "no display"

Added support for turning off the automatic display detection of a station. Useful for sub stations etc.

Errors Corrected

Multi module in combination with AMC 8 or 9 reset:

Incompatible tone handling fixed.

Issue 2948: Technical Alarm Inputs (RCI) on AC7:

Amc software RCI supports for AMC hardware 8000/4.

Issue 3027: MultiModule and SIP:

Combination of Multi Module and SIP is working .

Issue 3032: Search string on AlphaNet IP:

When doing a call from node A to Node B and the station in node B has a search string active the search will now work.

Issue 3055: Slow call setup when dialling from Master to Slave:

Software “bottle neck” fixed.

Issue 3067: Volume (783) and COS (7873) are not stored in flash:

Volume and COS setting is now stored in nvram for survival after reset.

SIPD 01.02 upgrades and corrections

Event “Conversation Outgoing” :

Outgoing calls to SIP: Report Event “Conversation Outgoing” (8) ON when receiving SIP 200 OK, reports OFF at disconnect.

CANCEL_MAIL:

Outgoing calls to SIP: Send CANCEL_MAIL for outgoing call, which cancels call request from that SIP phone.

Events for digits from called SIP:

Outgoing calls to SIP: If called phone presses digits which are sent to AlphaCom as SIP INFO, "Event Trigger Feature" (15) is reported. The digit is sub event (0-9, * = 10, # = 11). Calling AlphaCom station is Event Owner, and called SIP phone number and node number is related to. This could be used for e.g. Dooropening.

Dialing after initial connect sent as SIP INFO:

Outgoing calls to SIP: Digits dialed after initial connect are forwarded as SIP INFO digit messages to the remote SIP device. This can be used for two step dialing from AlphaCom to an external system, or to the PSTN.

Register from port different than 5060 allowed:

Call to SIP: Use port-number stored in registry for contact. Allows SIP device to register from any (random) port, not just 5060. Fixes issue with XLITE 3.0.

Registrar, remove all bindings:

Store only one binding for each user. Remove binding if expires==0. Fixes failure if SIP device registered again from different address/port

SDP media description:

Call from SIP: handle that SDP a= missing, use m=... Fixes issue with XLITE 3.0.

Issue 3059: SIP response handling, call to SIP:

Call to SIP: Improved handling of non-successful SIP responses 300->699: Forward response to AMCD followed before proper disconnect of call towards AMCD. Finally SIP ACK is now sent.
This allows AMCD to handle busy SIP phone properly, and avoids retransmissions from SIP phones due to missing the ACK. Forward 1xx invite responses to AMCD, instead of sending a faked 180 ringing directly on AudioPathSetup from AMCD. Fixes issue with short ring tone when two-step dialing to a SIP gateway.

Call on Hold from SIP:

When receiving a re-INVITE, respond with 488 not acceptable here. Quick fix of problem that X-lite was not able to disconnect a call if the call was put on hold by pressing other line button.

Display name:

Outgoing calls to SIP: don’t send STATION_INFO to AMCD if called name not received from SIP. Fixes issue with blank upper display line for outgoing calls.

Issue 3058: Default route was required:

Fixed failure in IP address lookup if default route not defined for the system, even if no routing was required.

Crash fixes:

Fixed memory crash if SIP 1xx response received out of context. Fixed crash if receiving REGISTER with more than 9 digits in user name

SIP Debug/Trace:

SIP trace/debug console on UNIX socket /tmp/sipd_trace. Connect using "tst -s /tmp/sipd_trace" from local shell. Possible to change trace level during runtime, press digits 0-7. Default level is 4, which prints SIP and AlphaNet messages. Press 0 to stop tracing. Formatting of message trace improved.

SIPd crash debug:

If SIPD crashes due to segmentation violation, a short debug message is printed to “/var/log/sipd_crashes”. Only the last crash is recorded, to avoid filling up the file system.

Issue 3007: SIP - digit during connection:

It is now possible to send digits from AlphaCom to SIP during a SIP connection.

SIPD 01.02 known limitations


SIPD do not remove expired REGISTERs. Can cause problems if changing numbers and node numbers of SIP phones. Workaround: Log in, and delete "/var/opt/amc/nvram/sip_registrar", reset exchange.

Digits during conversation can not be sent or received using RTP (RFC 2833 ). Only SIP INFO supported



AMC 10.02 (2006-08-22)

Release: Official, available on request 
/opt/amc/bin/amcd
NVRAM version 10.00. 

System upgrade file:
alpha_sys_10_02.tbz2

Precautions:
alpha_sys_10_00.tbz2 must be installed.

Only FPGA for the backplane issue is functionally changed. (FPGA 1.66) AMCD is functionally equal to 10.01.

Errors Corrected

Issue 2742: FPGA problems with backplanes:

In AlphaCom E20 and E26 with green backplanes there could be an audio issue with approx. more than 10 cards in the rack. (22 cards with FPGA 1.60),



AMC 10.01 (2006-08-09)

Release: Official, available on request 
/opt/amc/bin/amcd
NVRAM version 10.00. 

System upgrade file:
alpha_sys_10_01.tbz2

Precautions:
alpha_sys_10_00.tbz2 must be installed.

Errors Corrected

Issue 3019: Global group call:

AlphaNet VoIP channel lock up when receiving GGC from AMC 09.xx

Issue 3021: Global group call, multi module/IP:

Build global group call (7879) was needed after reset

Issue 3022: Group call:

Multi module over IP. M-press during Gong was ignored

Issue 3023: Global group call:

Multi module/IP; Reset when receiving GGC.

Claim 3219: Report EVH_CONV_OUTGO OFF:

EVH_CONV_OUTGO_OFF is now reported if disconnect during connection tone.

Issue 2881: AlphaWeb- AMC config backup page:

Restore backup: AMCD respond with "OK" to AlphaWeb _before_ reset, so AlphaWeb returns to backup page.

Issue 3025: Multi Module serial port:

When the serial port is set for multi module the data message “SET MODE = 66" was not transmitted, thus the slave would never become a slave.

Issue 3027: Multi Module and SIP:

SIP was not working in combination with IP multi module.

Issue 3030: RS485:

RS485 was not working because mismatch in configuration handling.

Issue 3016 AlphaNet Call Priority:

Station with Call Setup Priority = Alarm did not get through when all AlphaNet lines are busy due to conversation, global group call or SX conference.

Issue 2907: AMC: Event - Related to UDP 8:

Events with Related to = UDP 8 was not triggered.

Issue 2995: Noise when VoIP call from DAK:

When making a VoIP call from a DAK programmed to do prefix + dirno (54 + 101), and DAK is kept pressed after the connection, loud noise was transmitted to the B-subscriber.

Issue 3009: Multi Module License use:

Multi Module automatically allocated all licenses not used for AlphaNet even in stand alone mode. This gave a confusing license overview in AlphaWeb



AMC 10.00 (2006-06-01)

Release: Official, available on request 
/opt/amc/bin/amcd
NVRAM version 10.00. 

System upgrade file:
alpha_sys_10_00.tbz2

Precautions:
patch_to_release_10_00.tbz2 must be installed on beta cards before alpha_sys_10_00.tbz2 is activated.

Errors Corrected

See release note for beta software for error corrected during beta releases.





Known Issues (Latest release)


Known Issues

Issue 3020: Missing “Exch. Missing” text:

When making a global group call and some exchanges do not receive the group call due to not available audio links, there should be a display message "Exchange Missing".

Issue 3050: ODX from slaves via VoIP link:

ODX from stations in slave modules are not supported over VoIP link

Issue 3124: AlphaNet - NAT

AlphaNet does not work if NAT traverse is used.

Log System (Syslog)


The logging is handled by the syslog-ng log daemon. This can be configured to route the logs to different media. The log can be stored on on-board flash, or sent over the network with different protocols.

Known Issues

Large log files on on-board FLASH

The log is stored on a limited sized Flash partition by a file system. Routines to handle larger log amounts, like automatic clean up of older files have been implemented, but we still experience problems if the log rates get to large. These problems occur before a theoretical log rate versus free space analysis, mainly because of file system issues with a small number of available flash sectors. This file system factor makes it difficult to make an accurate estimate of handled log rates so the following is based on experience:

Limits:

Log rates lesser than 1440 events a day (an average of one each minute) should be handled with no problem.
Log rates of 17280 events a day (one each 5th second) is experienced to give problems.

Guideline:

If you are above 1440 events a day you should evaluate to use the Remote Syslog option, and turn off the Local Filesystem Log.
If you are closer to the 17280 events log rate we strongly advise to use only the Remote Syslog option. 
If your system are logging above the 1440 events a day to the filesystem you should regularly monitor the logs.

Note:

From AlphaCom 10.23 a log limiter on the System and Debug log is implemented, allowing for a maximum of 60 events a hour. 
If this limit  is reached, it will be notified by a log message. Because of this limit the above guidline will only apply 
to the Statistics log.

Hardware Drivers


Rtpdaemon

/opt/amc/bin/rtpdaemon

Rtpdaemon is a user mode service which handle the RTP audio streams. It receives control commands from the AMCD main application on a control socket (/tmp/rtpd). Rtpdaemon transfers and receives RTP packets via standard Linux network sockets. Rtpdaemon packs and unpacks RTP packets. Received packets are buffered before play out. It transfers and receives serialized audio data to the DSP via a DSP driver.

Version 01.06: 2007-12-10
Description:

Version 01.05: 2007-11-08
Description:

  • Optimised socket handling for unicast'ed groupcalls.


Version 01.04: 2007-10-17
Description:

  • G.729, first experimental support (no support for DTX, lost or reorderd packets).

Version 01.03: 2007-10-11
Description:

  • Issue 3269 Different UDP port on send/receive: Use sendto(), instead of connect()+write()
  • txtap function to tap audio to an internal socket.

Version 01.02: 2007-03-23
Description: Released version.
Introduced in system upgrade file: alpha_sys_10_20.tbz2

  • Jitterbuffer adjustments (Issue 3101 Xlite). Improve stabilty of delay adaptation, as well on adaptive delay target.
  • Set IP TTL to 31 when connecting to UDP to multicast.

Version 01.01: 2007-03-05
Description:

  • Fix issue 2935: Crash when ExpressTalk sends packet with zero payload at disconnect.
  • Improved handling of termination signals with logging.

Version 01.00: 2006-05-31
Description: Released version.
Introduced in initial release

SIPdaemon

/opt/amc/bin/sipd

SIPdaemon is a user mode service which handle data communications with SIP devices. It receives AlphaNet control commands from the AMCD main application on a TCP socket (port 40000).

Version 01.13 upcomming:

  • Send and receive INFOs for DTMF signals A - D

Version 01.12 (2008-01-03):

  • Outgoing INVITE, early cancel: Cancel before "180 Ringing" caused lockup, because of incorrect check for "SIP-dialog".

Version 01.11 (2007-12-10):

  • Outgoing INVITE: Send AUDIO_PATH_STATE(TRYING) to AMCD immediately, allowing AMCD to handle cancel before first response from external SIP device.

Version 01.10 (2007-10-12):

  • CANCEL of outgoing INVITE: Send CANCEL, not BYE

Version 01.09 (2007-09-18):

  • Fixed bug in reINVITE handling as implemented in 01.08: Wrong RTP portnumber is used if port number in reINVITE is less than 4096 (0x1000).

Version 01.08 (2007-08-30):
Only released in X-version 10.22 package

  • Handle reINVITE which redirects RTP audio to different IP address and port.
  • Fixed bug in parsing of AUDIO_LINK_OK from AMCD, which could be rejected erroneously

Version 01.07 (2007-07-01):
Description: Released version, date 2007-06-01.
Introduced in system upgrade file: alpha_sys_10_21x0604.tbz2

  • Removed memory leaks, increasing stability (issue 3176).
  • Issues 3203, 3182.
  • Debug error messages forwarded to syslog.
  • Set scheduling priority.

Version 01.05:
Description: Released version.
Introduced in system upgrade file: alpha_sys_10_20.tbz2

DSP driver

/opt/amc/modules/dsp_drv

DSP driver is a kernel mode driver which provides a device file interface (/dev/dsp/) to RTP daemon for communicating control commands and audio to/from the DSPs.

Version 02.10: (2008-01-16)
Support for DSP_SW version 01.10: Setup of codec remapping for optimized DSP SW. (DSP SW 01.10 require driver version 02.10, but driver version 02.10 can support older versions of DSP SW)

Version 02.01 (2007-12-10):

Version 02.00: Board support package 03.xx (Linux 2.6):
Description: Released version.
Introduced in system upgrade file: alpha_sys_10_03.tbz2

Version 01.10 (2008-01-16):

  • Same as 02.10, but for 2.4 linux kernel: Support for DSP_SW version 01.10

Version 01.01 (2007-12-10):

  • Same as 02.01, but for 2.4 linux kernel

Version 01.00: Board support package 02.xx (Linux 2.4):
Description: Released version.
Introduced in system upgrade file: alpha_sys_10_03.tbz2

DSP SW

/opt/amc/images/amc_dsp.hex

SW for the two DSPs. Currently the two DSPs runs identical SW. DSP does the G.711/G.722 transcoding. It also generates tones which are used in the system. Audio is transferred to the FPGA in 16 bit PCM format.

Version 01.14: 2008-04-21
Conference mixing resources in DSP.

Version 01.13: 2008-04-07
More AGC adjusting: Reset at connection start, and wait 5 sec before start adjusting.

Version 01.12: 2008-03-03
AGC adjusted, stability timer wait 1 sec before reducing gain.

Version 01.11: 2008-02-15
Treble preemphasis towards backplane (DeEmp from backplane done in FPGA). Level adjustment and limiter function on 20xx/21xx codecs, ref AMCD 10.31.

Version 01.10: 2008-01-16
Optimized codec processing. Each codec type has now variable number of channels, sum is always 32. Controlled by structures set up from dsp_drv. Also optimised DTMF generators, units 0400-040f. Bottom line is that 32 channels is now working, with headroom for further development.
Requires dsp_drv version 02.10 / 01.10

Version 01.06: 2007-12-10
Full version with Line Echo Cancellation (LEC). 10 LEC-instances. (LEC processing moved to DSP#2, allowing 32 codec instances in DSP#1).

Version 01.05: 2007-11-20
Customer specific variant with Line Echo Canceling (LEC). Based on OSLEC. 6 channels of LEC, number of codec channels (G711/G722) reduced to 6.

Version 01.04: 2007-10-17
Description:

  • 16 bit linear PCM at 8Hz support, which is required for the G.729 support in rtpdaemon 01.04.

Version 01.03: 2007-03-05
Description: Released version.
Introduced in system upgrade file: alpha_sys_10_21.tbz2

  • Improved mixing units to support DTMF tones: 32 mixers, independent mixers(function=1 in "con" of input)

Version 01.02: 2006-11-30
Description:

  • DC-reduction filter on signals from backplane (to IP).(First order high pass IIR filter: timeconstant T ca 4ms, cutoff frequency ca 40 Hz)

Version 01.01: 2006-06-14
Description: Released version.
Introduced in system upgrade file: alpha_sys_10_00.tbz2

FPGA FW

/opt/amc/images/amc_ip_fpga.bit

Firmware for the FPGA. FPGA converts audio between PCM and the AlphaCom SigmaDelta format. FPGA also interfaces the time slotted audio buses on the AlphaCom backplane, and thus replaces the SBI ASIC used on earlier AlphaCom boards.

Version 01.67:
Description: Released version.
Introduced in system upgrade file: alpha_sys_10_03.tbz2

MBI driver

/opt/amc/modules/mbi_irq

MBI irq driver is a kernel mode driver which provides a signal to the AMCD main application when an interrupt is generated from the master backplane interface (MBI)

Version 01.00: Board support package 02.xx (Linux 2.4):
Description: Released version.
Introduced in system upgrade file: alpha_sys_10_00.tbz2

Version 02.00: Board support package 03.xx (Linux 2.6):
Description: Released version.
Introduced in system upgrade file: alpha_sys_10_03.tbz2

LED / Watchdog driver

/opt/amc/modules/dev_amc_wdog

The watchdog driver is a kernel mode driver which is used for updating the hardware watchdog. This driver is also used for accessing the AMC-card LEDs

Version 01.00: Board support package 02.xx (Linux 2.4):
Description: Released version.
Introduced in system upgrade file: alpha_sys_10_00.tbz2

Version 02.00: Board support package 03.xx (Linux 2.6):
Description: Released version.
Introduced in system upgrade file: alpha_sys_10_03.tbz2



Hardware Versions


AMC hardware versions

Known problems AMC hardware 8000/4


None as of now

Known problems AMC hardware 8000/2


Issue 2747: RCI not supported on ACE7:

RCI not supported on ACE7. The RCI signals on P1-c19 and P1-c22 are terminated in test points on AMC-IP.

Issue 2741: Redundancy control from APC:

The redundancy control system from APC is not working (software and hardware).

Issue 2787: AMC serial port: No RX, no data on TX:

It turns out that the RS232-drivers on the AMC-IP-board have an automatic shutdown when it detects missing received data (illegal voltage levels). This is fixed in hardware version 8000/4, but it can be fixed on older hardware versions by a minor modification.

Issue 2806: “Temperature Alarm" in AlphaCom E7:

This problem is probably related to the fact that the AMC-IP board never had a connection to the over-temp. signal from the ACE7 backplane. This is related to the problems with RCIs from the same backplane.

AMC Filter board

Issue 2723: RS422/RS485 Signal Pinning:

The pin out for RS422 signals on the filter print for E20 and E26 differs from the pin out on the E7. The Rx+ is switched with the Rx- and the Tx+ is switched with the Tx-. The only consequence is that the same cable can't be used on E7 and the E20-series. From filter print version 3 (DB8001/3) the mapping is correct.