View source for AlphaCom 11.02 - Release Notes

[[Category: Release notes]]
'''Software in production:'''	AMC 11.00<br\>
'''Software released date:'''	2010-04-22<br\>
'''Note:'''  For each software version the NVRAM version is listed.  
If the NVRAM version is different, the AlphaCom InterCom configuration will get default configuration, and then you must do a SendAll from AlphaPro to restore the configuration.<br\>
All AlphaWeb configuration will be kept.<br\><br\>

=AlphaCom 11.2.3.x Release Notes=
<br\>

= AMC 11.2.3.11 (2015-01-29) =
 Release: Official, available on request
 NVRAM version 11.02 (Size:1054540).
 
'''The nvram size is changes thus all configuration will be lost after upgrade from 11.2.3.x.''' <br>
'''Turbine Output and DAK key mapping is changed'''

''Running this software on AMC 11 hardware require use of STIC'' <br>
 
'''System upgrade file:'''<br\>
alpha_sys-11.2.3.11.zip <br>

Due to an issue with the AlphaWeb upgrade dependency test, the "alphaweb_upgrade_fix.apkg" must be installed prior to alpha-sys-11.2.3.11.<br>
(Not required if the system is already running version 11.2.3.10).<br>

alphaweb_upgrade_fix.apkg<br>
alpha_sys-11.2.3.11.apkg<br\>

===M100 to AlphaCom IP Connections now will Use LEC===
Calling from M100 to an IP connection (AlphaNet, SIP or IP station) could generate echo during the conversation. Now the internal Line Echo Canceler algorithms of the AlphaCom is used for M100 connections.

===Remappig of Turbine and Turbine Extended DAK keys and I/O (MTN-531)===
The mapping of I/O and DAK keys are changes to avoid overlapping situations.
*DAK keys and DAK LED when configured in AlphaPro as "DAK_as_RCI" or "IND" or "DAK", index from 1.
*INPUT/OUTPUT when configured in AlphaPro as "IND" or "DAK key" or "DAK_as RCI", index from 11.
*Station flag for moving INPUT/OUTPUT to index 101. (Ask support for further information)

===Recordig Issues (MTN-551)===
Recording of outgoing analogue calls to AlphaNet did not send audio to the recorder.

===Added Directory number "7632" to Autoload (MTN-571)===
Directory number "7632" was missing in the autoload of AMC

===SIP Station Registration Memory Issue===
Leaking memory due to repeated SIP station registration fixed.

=== Stability Issues ===
Fix stability issues related to Trunk setup in combination with SIP stations.

=== SysLog Issues ===
Issues related to generating text for statistics syslog

==Enhancements==
===Parameter for Volume Override Event Handler: AMS and AMG (MTN-549)===
Volume override can now be turned off for AMS and AMG use in event handler.

 AMS <phys.no> <alarm.no> <hvo>
 <hvo> handset volume override, optional parameter, 1 = on, 0 = off. Default = 1

 AMS 23 2 0 – Alarm message 2 (8192) to station 23, without volume override

===$DTS Support in 11.2 (MTN-555)===
The $DTS function is now supported in 11.2.

===Turn off "Off Hook Timeout"===
Setting the Off Hook Timeout to 0 now disable the off hook detection timer.

===Use of Wavfiles for generating Private Ringing Tone in Stations (MTN-563)===
From AlphaWeb Messages upload "Announcement Message(30)" with indexes from 80-89. (Ringing tone playback index 1-10)
The selection of message to private ringing playback for each station can be configured in latest AlphaPro version. <br>
(Ringing Groups can also be configured with related wav playback. Currently no AlphaPro support) <br>

= AMC 11.2.3.10 (2014-06-12) =
 Release: Official, available on request
 NVRAM version 11.02.

''Running this software on AMC 11 hardware require use of STIC'' <br>
 
'''System upgrade file:'''<br\>
alpha_sys-11.2.3.10.zip <br>
The ZIP file includes an AlphaWeb patch file to be uploaded prior to the alpha_sys file <br\> 
alphaweb_upgrade_fix.apkg<br>
alpha_sys-11.2.3.10.apkg<br\>

== Error Corrected ==
===Stability Issues Related to High Traffic and Recording. (MTN-445, MTN-443)===
*Fixed issue when station in a slave module lifts handset during recording.
*Fixed possible conflict if two recording sessions ended at the same time.
*Fixed possible issue when going fast from conversation mode "duplex" -> "simplex" -> "close".
*Better handling of disconnect and "SIP OK" signaling from the Recorder.

===Issues with Proxy Station Names===
Added cleanup of the temporary directory number table if no free table entries available.(Used for incoming and outgoing calls on trunks: AlphaNet and SIP)

===Global Ringing Group and SIP. (MTN-446)===
Issue with SIP stations in Global ringing group. Dependent on timing calls could be missed out on SIP stations.

===Issues with Hardware Watchdog Update During Linux Boot===
The hardware watchdog could on some AMC cards time out during Linux boot resulting in endless reset loop. The boot timing is now fixed.
It a card does not start due to watchdog timeout the following can be done:
    Disable Hardware Watchdog, start the AMC card.
    From AlphaWeb upload and install the package “alphaweb_upgrade_fix.apkg”. (No reset required)
    From AlphaWeb upload and install the package “alpha-sys-11.2.3.10.apkg”
    When Installation is finished and the system up and running: Enable Hardware Watchdog and verify that the system will boot as normal.

===ODX and Digit During Conversation===
Fixed possible issue during ODX and digit during conversation.

===M100 and Handset Use===
Fixed issue with handset and m-key use when calling from AlphaCom to M-100.

==Enhancements==
===Turbine DAK as RCI===
To avoid overlap with DAK modules the RCI inputs of Turbine and Turbine Extended are signaled as DAK 101 and upwards.
===Turbine RCO ===
Control of the Turbine RCOs must be done with [[$SET_IPRCO]] or with logical RCO configuration from AlphaPro.

===Support of Turbine Extended===
Possible to control DAK IND on Turbine extended stations.

===SX conference log===
SysLog statistics now contains SX conference information.

===Collected Digits on SIP trunk===
Support of a maximum of 32 digit collect on SIP trunk. <br>
Previous maximum collected digits was 16.

===Correct "missed call" Information on SIP stations===
SIP stations registered on AlphaCom and receiving calls that is not answered will now be able to indicate "missed calls" (if supported in the SIP phone).
Correct status of "missed call" also supported when the SIP phone is member of a Ringing Group or Parallel ringing feature.

= AMC 11.2.3.9(2013-09-03) =
 Release: Official, available on request
 /opt/amc/bin/amcd
 NVRAM version 11.02.
'''System upgrade file:'''<br\>
alpha_sys-11.2.3.9.apkg<br\>
''Running this software on AMC 11 hardware require use of STIC'' <br>
''11.2.3.x in Master module with older software in IP slave modules will give audio problems''
== Error Corrected ==
=== The System Can Block After Appox. 120 Conversations ===
Due to system resources allocation issues the call setup can block when all 256 time-slots in the back-plane have been used once. The error was introduced in version 11.2.3.8.

= AMC 11.2.3.8(2013-06-27) =
 Release: Official, available on request
 /opt/amc/bin/amcd
 NVRAM version 11.02.
'''System upgrade file:'''<br\>
alpha_sys-11.2.3.8.apkg<br\>
''Running this software on AMC 11 hardware require use of STIC'' <br>
''11.2.3.x in Master module with older software in IP slave modules will give audio problems''
== Error Corrected ==
=== EDO as TCP server disconnects after sending a string ({{MTN|283}})===
An EDO port in TCP server mode disconnects after sending a string depending on the setting if the port was in TCP client mode first.
=== Support for ATLB12 version 5.54 ===
The ATLB12 version 5.54 with new ringing pattern needs special attention to time slot use during duplex operation. Using older software can result it blocked duplex switching operation (Phone to ASLT station). <br>
A system with only handset use will not be affected.

== Enhancement ==
===Data Command for Sending Digit Signaling to SIP station ({{MTN|148}})===
New [[D2S|$D2S]] command for digit sending to AlphaCom registered SIP-station.
===Priority Direct Paging with Feature 93 ({{MTN|160}})===
 Parameter 2 = 1 uses priority page template for message sending. Other values uses normal paging template.
===Backward Compatibility Flag for $EXT_MAIL ({{MTN|193}})===
The "net_ref_obj" parameters of the $EXT_MAIL will for AMC 11 generate various types of net_obj_ref classes. This is according to system specification but some 3rd part systems have not implemented all the net_obj_ref classes. To force AMC to generate equal net_obj_ref classes as for AMC 9-10 a new flag has been introduced.
 .ex_profile.flags.AMC10_bc


= AMC 11.2.3.7(2012-12-20) =
 Release: Official, available on request
 /opt/amc/bin/amcd
 NVRAM version 11.02.
'''System upgrade file:'''<br\>
alpha_sys-11.2.3.7.apkg<br\>
''Running this software on AMC 11 hardware require use of STIC'' <br>
''11.2.3.x in Master module with older software in IP slave modules will give audio problems''
== Error Corrected ==
===During boot the AMC Card Send ARP for the Default IP address===
The AMC cards did gratuitous ARP broadcast of IP address 169.254.1.5 early in the boot sequence. <br>
This is now removed to avoid IP-HA issues if default address is used as the operational address.

== Enhancement ==
=== OPC System added IP-ARIO State Information. ({{MTN|101}}), ({{MTN|102}})===
Following OPC information are now available in the $ST_STATE message:
*State information of the 8 input and 8 outputs of the IP-ARIO.
*The PA-fault state bitmap.
*IP Station hardware type.
===IP Station Signaling During Tone Setup===
Improved setup signaling for audio quality (Turbine).

= AMC 11.2.3.6(2012-12-17) =
 Release: Official, available on request
 /opt/amc/bin/amcd
 NVRAM version 11.02.
'''System upgrade file:'''<br\>
alpha_sys-11.2.3.6.apkg<br\>
''Running this software on AMC 11 hardware require use of STIC'' <br>
''11.2.3.x in Master module with older software in IP slave modules will give audio problems''

== Error Corrected ==
===Unintended IP HA Switchover due to High CPU Load ({{MTN|88}})===
The timeout for monitoring of 100% CPU load is increased from 10 to 30 seconds.  <br>
This to avoid unintended switchover due to for example large number of simultaneous IP station registrations. <br>
(MTN-93 will also cure the same problem)<br>

===Stability Issues of the High Availability System ({{MTN|50}})===
Adjustments to avoid conflicting situations resulting in unnecessary switchover. <br>
The arbitration algorithm extended to evaluate 
*Ethernet connectivity. (Avoid going operational when no link on the Ethernet Connection.)
*Number of current registered IP stations. (The node with the highest number of registered IP stations will win a conflicting situation)
Previous IP-HA version decreased its own priority 60 second after start up. This could result in monitoring link failure and unintentional HA switchover in situation when system load is high.  <br>
===IP-HA Software Version Conflict ({{MTN|74}})===
Previous IP HA version could report software conflict also after both HA nodes was upgraded. An additional reset to remove the conflicting situation was required. <br>
This is now solved.
===Station Lockup During ASVP Playback and no IP resources ({{MTN|51}})===
In situation with all IP resources in use and triggering features using voice messaging the system could partly become unresponsive.
===Issues in Header of AlphaNet broadcast messages ({{MTN|48}})===
Destination node number mismatch of broadcast messages.

== Enhancement ==
===Avoid Repeated Error Messages to SysLog ===
Continuous conflicting IP-HA situations would generate repeated error messages. (some every 3 seconds).<br>
Now these situations are reported only once:<br>
*Conflicting operational mode.<br>
*Conflicting operational address.<br>
*Conflicting software version.<br>
===Restrict the IP Station Registration Process ({{MTN|93}})===
The previous version would run with 100% CPU when hundreds of IP stations did simultaneous registration. <br>
The registration process is now default restricted to allow at max 20 new stations every second. <br>
Adjustable from tst:
  ex_profile.glob_const.IPS_reg_pr_sec = 20 
===Avoid Intensive Software Processing when all IP Audio Resources are Busy===
Previous version went through intensive software processes when activating features requiring IP audio in situations when all IP audio resources already busy. <br>
Now the test of free resources is done early in the process. This to avoid saturation if a number of simultaneous events occurs, thus running out of resources.
===Node reset syslog information from IP-HA standby exchanges ({{MTN|75}})===
The previous IP HA version suppressed reset information to syslog from the standby exchanges. <br>
Now syslog receives the reset information message also from standby exchanges.
===Hostname of IP-HA Configuration Slave is Changed ({{MTN|83}})===
Added “_S” to the hostname of the HA configuration slave. This to distinguish the IP-HA exchange source hostname when external SysLog server is used.
===Additional information added to the “789”-info feature===
The “789” feature added:
*Inform about HA-IP configuration slave is currently operational. <br>
(If configuration master is operational no information is added.)
*Inform about local IP address of the IP station activating the "789" system info command.
===Avoid logging "history" of commands used on SSH console ({{MTN|96}})===
Use of the SSH console result in logging to a history buffer of the 1000 last commands from SSH. <br>
Unnecessary write operations to the flash file system should be avoided. The default “.bash_history” file update is now deactivated. <br>
===Flag Forcing Selected Codec to be used for Incoming Trunk Calls===
AlphaCom will by default accept other codecs for incoming call from remote trunk as long as the codec is on supported by AlphCom (G722/G711u/G711a). <br>
Setting the force_CODEC flag will only accept the AlphaPro selected codec.
 .ex_profile.flags.force_CODEC = 1




= AMC 11.2.3.5(2012-09-20) =
 Release: Official, available on request
 /opt/amc/bin/amcd
 NVRAM version 11.02.
'''System upgrade file:'''<br\>
alpha_sys-11.2.3.5.apkg<br\>
''Running this software on AMC 11 hardware require use of STIC'' <br>
''11.2.3.x in Master module with older software in IP slave modules will give audio problems''

== Error Corrected ==
=== ({{MTN|17}}) and ({{MTN|18}})) Problems with IP-High-Availibility===
The IP-HA system could leak memory thus slowing down the AMC card.<br>
The IP-HA system could in some situations not restart correctly.<br>

=== ({{MTN|20}}) Door Opening from IP Station DAK Keys ===
The signaling of digit "6" for door opening from an IP station DAK key could in some cases fail due to internal signal timing issues.
 
=== ({{MTN|27}}) IP-ARIO Does not Have SX Audio After Exchange Reset ===
Issues with codec selection after reset could inhibit IP-ARIO being default speaker after restore of conference.

=== ({{MTN|34}}) Digit Key is DAK Does not Work ===
The flag "Digit key as DAK" did not work for IP stations.

=== ({{MTN|27}}) M-press on a Turbine station in simplex conference leads to error messages ===
Using DAK key for M-key signaling programmed as "DAK as RCI" could generate SysLog warning messages for some features. This applied to all types of IP-stations not only Turbine.
 
=== AlphaWeb and Turbine Stations ===
The AlphaWeb station configuration and IP station software upgrade menu now support Turbine stations.

== Enhancement ==
===({{MTN|16}}) Delayed Automatic Dimming of the Display Back light of IP Stations ===
The dimming of the back light in IP stations is now delayed with 5 second. <br>
This allow the DualDisplay DAK panel to have back light after selecting a new DAK page.

=== Sub event for IP-HA Configuration Master ===
Sub event 232 of [[System_Status_(Event_Type)|system status event]] now report if the AMC card is configuration master in a HA-pair.

=== MPC Protocol with Global Numbers ===
MPC protocol commands now support global numbers.

= AMC 11.2.3.4(2012-07-26) =
 Release: Official, available on request
 /opt/amc/bin/amcd
 NVRAM version 11.02.
'''System upgrade file:'''<br\>
alpha_sys-11.2.3.4.apkg<br\>
''Running this software on AMC 11 hardware require use of STIC'' <br>
''11.2.3.x in Master module with older software in IP slave modules will give audio problems''

== Error Corrected ==
===({{MTN|5}}) New STIC driver ===
Problems with internal driver reading the "STIC" could result in no response from some menus in AlphaWeb.

===({{MTN|11}}) SoftClient and Group Call===
The IP-SoftClient does not support multicast audio, when using multicast audio the SoftClient would not receive group call. <br>
Now the SoftClient will receive uni-cast audio also when other IP stations receives multicast.

===({{MTN|13}}) AMC Local Fader Resources===
Only 11 of the 12 internal AMC board faders was available for use thus resetting the system if the 12th was used. 

===M100 Format of Display Text ===
M100 integration issue with display text format fixed.

===MSC Talkback in "off hook" Mode===
Fixed issues with (IP) MSC station using talk-back when "off hook".

==={{Bugzilla|751}} ATLB12 Station Corrupted Ringing Pattern===
The ATLB12 stations ringing patterns could be altered due to changed ringing tone activation system in previous AlphaSys version.

=== Billing Blocking Trunk Calls ===
The Billing system could block succeeding users from calling trunk lines if the previous user was blocked.

==={{Bugzilla|734}} IP Dummy Station Failure===
In AlphaSys 11.2.3.3 the IP dummy station did not work correctly. Dummy stations could be reported as "not registered".

==={{Bugzilla|670}} SX-Conference and Default Member ===
When activating and deactivating a SX-conference several times in a short time period the default member did not always operate correctly.

=== SX Conference and Handset ===
When using handset and entering SX-conference the off hook tone is still active.

==={{Bugzilla|718}} Noise Reduction Setting Reset after Exchange Reset ===
The changes of noise reduction setting for IP stations was not stored correctly to flash in AlphaCom. 

===({{MTN|2}}) Cancel Calls with $CAC ===
The parameter for priority of the call to be canceled was not correctly checked thus canceling call with high priority than the parameters to $CAC allowed could occur.

==={{Bugzilla|746}} Recording Level Adjustment ===
The level of recording was not correctly adjusted to "telecom" levels for single stream "mixed" IP station recording.

==={{Bugzilla|746}} Recording Audio Missing for Mixed IP-IP Conversations ===
Conversations of two IP stations in mixed audio stream only receive only audio from one of the IP stations.

===Recording and Global Group Call Initiated from Slave Module===
Sometimes a group call initiated from a slave module in combination with recording could fail and block internal resources. 

===Issues with Removing the Recorder Configuration===
AlphaCom did not always detected that the recorder was removed from the configuration.

===SIP and Ringing Group ===
*Incoming call from SIP with no directory number failed when initiating a ringing group call. To avoid the problem the following is added:
**Default node + trunk physical proxy station is inserted as default source number.
**To configure desired source number different from default: 
 Configure a "user phone extension" in AlphaPro with wanted phone number. 
 (Remove physical number to allow phone number configuration)
  In NVRAM configure ".ex_profile.ip_config.user_def_trunk_drno = X" with X as the user index number used as source phone number. 
 (This is the "user index" 1-600, not the "directory number".)

===Ringing Group Issues in Combination with ASVP Messages===
There could be issues when using a custom "your call is registered" ASVP message for ringing group from SIP trunk.
 
===Ringing Group and M100=== 
Initiating ringing group call from M100 in combination with a multi-node AlphaNet gave issues if a remote AlphaCom node answer the ringing group call.

===({{MTN|4}}) Avoid Following Transfer of Initiator Station of Ringing Group Call===
If a station currently in transfer mode initiated a ringing group call the reply of the ringing group call would go to the transferred station.

===SIP stations and ODX conferences===
Issues when including SIP stations to an ODX conference ("56").

===AlphaNet Transit SIP Calls in Switched Duplex Mode===
If all duplex resources was busy when a new transit AlphaNet to SIP call requiring duplex is initiating,
the handling of the resources was corrupted thus making the duplex resource busy until next reset.

===Control of Back Light on IP Display Stations===
Previously the IP station display always was in "busy" mode after reset. (Backlight ON).<br>
The IP stations with display now gets the back light setting for current operation state after reset (idle, mail or busy). 

==Enhancement==
==={{Bugzilla|702}} Tone Test with Increased Volume ===
The tone test will now increase the volume of the test tone for each repeat of tone test when failed. Both for analogue and IP stations.

===New IP Turbine Station Support===
This software have added support for the new Turbine stations. (New call led control.)

=== Added System Syslog Report Faulty Station ===
Report to syslog for previously faulty station when they are reported OK is added. This to be able to check if the faulty station was down for a short or long period.

===New Behavior for AsaCom Integration===
Update of display text when doing outgoing AlphaNet call in local node changed.

==New Event Handler Features==
=== New Macro [[%nip]](), Node IP Address ===
 %nip(n): Return IP address of remote node "n". This function can not return the IP address of the local node.
=== New Macro [[%syse]](), System Event ===
 %syse(s): Return the current state of "27-system event" for sub-event "s".
=== New Macro [[%rn]](), Read Node Number ===
 %rn(dirno): Return the node number related to the directory number (global or local number).

===({{MTN|10}}) New IP-HA Sub Events added for [[System_Status_(Event_Type) | 27 - System Status]] ===

* Sub event 230. 
** ON     Local Exchange in IP standby mode.
** OFF    Local Exchange in normal operation. (Master mode)
* Sub event 231.
** ON     IP HA in connection with related HA node.
** OFF    IP HA, no connection with the related HA node.

=== New Event Report type [[Feature_State_Info_(Event_Type) | 39 - Feature State Info]] ===
* Sub-event 10:  User denied access to feature. %1 = user, %2 = denied feature.
* Sub-event 20:  User initiating call to busy station. %1 = user, %2 = busy user.
* Sub-event 21:  Re-open parked connection. %1 = A user (Doing for example inquiry), %2 = B user (parked).
* Sub-event 67:  Free station found in group hunt. %1 = user, %2 = Group where station found.
* Sub-event 106: Billing: Low balance, abort around the corner. %1 = user
Only "ON" change available.

= AMC 11.2.3.3(2012-03-07) =
 Release: Official, available on request
 /opt/amc/bin/amcd
 NVRAM version 11.02.
'''System upgrade file:'''<br\>
alpha_sys-11.2.3.3.apkg<br\>
''Running this software on AMC 11 hardware require use of STIC'' <br>
''11.2.3.x in Master module with older software in IP slave modules will give audio problems''


==Error Corrected==
==={{Bugzilla|647}} {{Bugzilla|688}} HAIP not working on 11.2.3.3 x versions===
High Availability IP
*Database replication failure.
*Systems with no SIP configuration would generate lots of error logging.
*Space in the HA descriptive name gave file name problems.
*Sometimes reboot was needed to clean up

==={{Bugzilla|709}} Using RFC2833 for DTMF signaling did fail in a conversation after RE-INVITE ===
AlphaCom did not remember the digit signaling type used for SIP conversation in situation after a RE-INVITE thus digit signaling from SIP was not working correctly.
==={{Bugzilla|667}} SIP (DECT) and ringing group===
AlphaCom supported maximum 9 digits for the "from" number when sending ringing group SIP-INVITE to SIP-stations (Incoming SIP trunk call to tinging groups with IP DECTs). <br>
When more than 9 digits number space was inserted resulting in faulty call setup towards SIP stations. Now 16 digits are supported.

=== SIP-station, too many digits ===
Problems with more than 15 digits in "to" number from SIP station. <br> 
Feature 81/83 with "digit collect" and collecting all digits in SIP up front now supports 8 digits for trunk number and 16 digits for B number.
Allowing maximum 24 digits in "to"-number received from SIP-station. Without "digit collect" in feature 81/83 at most 30 digits in "to"-number from SIP-station is supported.
* NOTE: The AlphaCom directory table is still limited to 8 digit numbers, but this issues are related to outgoing trunk calls with long numbers.

=== Reset of SIP Call to MP114 AudioCode gateway ===
Calls in "Early media" state would only last 60 second. 

===AlphaNet Routing in mixed AlphaNet. AMC9->AMCIP->AMCIP->AMC9===
Faulty AlphaNet setup using Area code feature in the AlphaNet routing: AMC9->AMCIP->AMCIP->AMC9.
===DIP messages to AMC 8/9===
Avoid tunneling of DIP messages to AMC 8/9 in mixed AlphaNet. This would generate error messages at AMC9.

===Old type ODX required duplex resources ===
Not needed duplex resources were allocated when configuring ODX with forced handset (original ODX).

==={{Bugzilla|717}} EDIO overlapping port numbers===
EDIO 5 and 6 did not work correctly due to overlapping internal port numbering in AlphaCom

==={{Bugzilla|727}} OPC station input state was reported in the output bitmap===
The state of IP station input was not correctly updated in OPC.
===Improved Command Server task monitoring===
Internal monitoring of AlphaCom tasks could fail leading to software reset.

===Missing ringing signal on ATLB12 phones===
An internal issue in the ATLB12 software resulted in termination of ringing signal of all ringing phones on the same ATLB12 card if one of the ringing phones lifts handset.
Workaround fix; AMCD resend the "start ringing signal" message to all ringing ATLB12 lines with same timing as the ringing signal tone for ASLT stations. 

===Reset when parameter errors $DSPL===
The $DSPL command used with illegal station parameter (remote node number in netref) could result in AMC reset.

===Duplex Switching Quality===
When calling a remote IP station over IP AlphaNet from an analogue station the duplex algorithm will run in the remote node. 
It was found that the calculation of RTP link delay will report the delay approx 10ms to long. 
This will effect the duplex switching decision when the IP station is speaker and analogue station is listener result in "chopped" audio.
Added a configurable default (negative) offset of 10ms:
 ex_profile.timeouts.duplex_delay_offset = 1
 
==={{Bugzilla|695}} Set master module to slave mode  ===
It was not possible to set a master module to slave mode if the master module was configured with slaves. <br>
If a slave with copy of master configuration was chilly reset so that this slave then became a master resulted in no more slave mode for this module.

==={{Bugzilla|694}} One way audio in multi module ===
Calling from AlphaNet to a station in Multi Module slave. If both station lift handset the result was microphone closing in slave station handset.

===Freeing of resources during faulty global SX conference setup ===
Missing audio links for a global SX conference would result in blocked UDPS resources.

==={{Bugzilla|576}} Tone Test on XE1 ===
Tone test can now be used for IP stations on XE1, this require new software on IP stations.

==={{Bugzilla|668}}{{Bugzilla|99}} ODX Conference problem when missing RTP audio===
ODX would be blocked in case of missing RTP audio from one of the participants.
Fixed problems with multi module and IP station in ODX conference.

==={{Bugzilla|656}} Et0/Eth1 IP address conflict reported when no conflict===
AlphaWeb range check of IP addresses fixed.

===$STM and mail events does not work if stations have 8 digits directory number ===
Internal mail parameter storing was not large enough for 8 digits.
Increase the parameter  size. This change also changes the state data structure -> chilly restart. 
The maximum number of mails in the mail pool is reduced from 2500 to 1500 because the physical NVRAM size limitations.

==={{Bugzilla|648}} $DP C-key and toggling of private mode ===
Avoid toggling of private/open mode for IP stations when using C / $C to terminate call request mode.
===M100D AlphaNet Node Type Report ===
M100D now answer requests for node info from remote nodes in AlphaNet. Node information is used to determine some parameters for AlphaNet call setup.

==Enhancements==
==={{Bugzilla|653}} Call forwarding to ringing group===
Call forwarding (71) to ringing group is allowed.

===Configure stations to use DTMF as cancel tone===
New station flag added to change the cancel tone of a station to a selectable DTMF tone.
(station flags3 - bit number 3)
[[Release_Notes_AlphaCom_10.6x#Added_DTMF_tones_as_handset_tone_after_call_reset_from_peer_station|Select DTMF tone]]

===RTP audio check for AlphaNet/SIP trunk calls default disabled===
The check for received RTP audio in AlphaNet links is now default disabled due to issues on SIP links where mute operation at remote end will stop sending RTP audio thus AlphaCom will reset the connection. 
The RTP check can be enabled with use of node flag bit 6.
===AGA/AE1 AlphaNet line test===
Allowing wider range of adjustment of the AlphaNet audio test speed. Previously from 10 sec ->, now from 0,6 sec ->, required for quicker alternative routing. 
Default test speed still 15 seconds.
===Billing export of CVS files===
===SysLog===
*$DP routing fault message added information to distinguish between "no routing" and "link down"
*Restart log message added information of standby state, IP and APC
===Event Macro [[%tin]], return current number of trunk channels used towards a specific trunk===

===20 day free license when STIC is not working===
In case of STIC fault the free license time is increased to 20 days.
==={{Bugzilla|649}} Allow setting priority threshold for cancel call request mail ===
 37 &000281 .ex_profile.glob_const.CR_CM_threshold = 255 
===Low level debugging for RIO===
New flag to enable more status output from the RIO operation.
*Logging of protocol fault, resending, failed ISO polling, crc error etc.
*Logging of resending of application layer device $PING.
 42 &000216 .ex_profile.flags.RIO_logging = 0 (0x00000000)
===IP Station Event Blocking ===
Allow blocking of events from IP station with $DP message. <br>
Station Event Block:
 $SEB L102 U1 U1 (Block M key from IP station)
 Parameters:
 1. Station
 2. Block event 
 3. Block state 0/1
 Block event number:
 1 Block M-key from IP station
 2 Block Handset state from IP station 
 3 Block Digit keys from IP station 
 4 Block DAK keys from IP station 
 5 Block C-key from IP station
 Block state will always be cleared after reset.

===Event Reporting intended for SIP node routed to own node instead===
When calling from SIP-trunk to AlphaCom the "Conversation Outgoing" event was sent to the SIP node (Start node). (And SIP have no event handler) <br>
*Problem: SIP calling a group call, how to trigger an event. GroupCall will only generate ConvOutgoing sent to start node.
*Solution: Changed routing of SIP Event Reports to local node, allowing event trigged with owner SIP UDP group at local node. 
Physical number will be the Trunk line, %1.nam, %1.dirno will be the SIP-trunk device info.

= AMC 11.2.3.2(2011-09-13) =
 Release: Official, available on request
 /opt/amc/bin/amcd
 NVRAM version 11.02.
'''System upgrade file:'''<br\>
alpha_sys-11.2.3.2.apkg<br\>
''Running this software on AMC 11 hardware require use of STIC'' <br>
''11.2.3.x in Master module with older software in IP slave modules will give audio problems''

==Error Corrected==
==={{Bugzilla|627}} IP-High Availability and IP GroupCall ===
HA Multi cast group call was in some situations not working correctly.
===IP High Availability and Free License===
HA now works also when using Free License. The free license will not be transferred from config master to standby thus both exchanges should have free license or one free license and one with all licensees required.
==={{Bugzilla|621}} IP High Availability and upgrade of IP stations ===
IP stations can now be upgraded from AlphaWeb also when running a HA system
=== {{Bugzilla|616}} Time zone setting in combination with IP High Availability corrupt the XML config file===
Configuring time zone after configured IP High Availability would corrupt the configuration file.
==={{Bugzilla|637}} APC-High Availability,  No gratuitous ARP on switchover ===
The gratuitous ARP broadcast sent after reset of an AMC card was not working thus some Ethernet switches would not route messages correctly when switching AMC card in the APC based system. For IP High Availability the gratuitous ARP worked correctly.
==={{Bugzilla|635}} Pending Flash Sync and AlphaWeb Reboot ===
If pending writing to the configuration flash file and the AlphaWeb reboot system maintenance function was used the database could be corrupted. 
==={{Bugzilla|629}} AlphaNet with alt.routing and AMC 9 ===
Problems when connecting a mix of AMC9 and AMC IP nodes. Some messages was not routed to the AMC9 system.
==={{Bugzilla|630}} Multi Drop Master Protocol ===
Multi Drop master protocol used with 2-wire RS485 had a number of collisions and did auto detect 4-wire behavior when pressed to the limit. Changed to forced 2-wire operation and cleaning of receive buffer before sending a message. 4-wire will work with 2-wire operation. The old auto detect can be turned on with the parameter:
 41 &000212 .ex_profile.flags.SMD_autodetect = 0 
==={{Bugzilla|632}} Billing Activated even if not configured ===
The billing system was started when billing license was activated. The system is now changed to also check if billing is configured.
===IP station Multi Cast Status ===
When reconnecting an IP station with "failed" multi cast group audio status the failed status was cleared and the failure state not detected before next AlphaCom reset.
=== {{Bugzilla|618}} In Event 10, %mpri is not working ===
Event 10-New mail %mpri/%tag failed if owner was a station. (OK if UDP).
=== {{Bugzilla|622}} Event 23, New current mail event not triggered ===
Event type 23 - New current mail is now triggered when navigating the queue using 70 + 7/9.
=== {{Bugzilla|574}} Busy override against two SIP phones in conversation===
Using busy override towards a SIP-SIP conversation failed.
===Configuring serial ICC did start unintended ICC trace in tst.===
Due to internal use of numbering a trace in TST would start if ICC on serial port was configured.
===Free Seating and high physical numbers===
Free Seating in WACS could move IP stations to physical numbers above 604 thus configuring AlphaNet proxy stations as IP stations inhibiting AlphaNet to work correctly.

==Enhancement ==
===Option for removing license===
Factory default and License can now be removed from the "System Recovery" menu in AlphaWeb.
=== SIP-trunk related physical station now also work for incoming call===
This allow restrictions and event handler to be programmed for selected numbers on incoming SIP trunk calls.

===Individual ringing volume for incoming conversation===
Option for individual volume setting on the ringing tone for IP station.
 .module_profile.st_profile[1].ring_volume = 0 (0 = follow station volume)
===$CALL flag for Forced Fixed Open Duplex===
The DP command [[CALL_SETUP|$CALL]] now can be forced to open duplex.

= AMC 11.2.3.1(2011-07-05) =
 Release: Official, available on request
 /opt/amc/bin/amcd
 NVRAM version 11.02.
'''System upgrade file:'''<br\>
alpha_sys-11.2.3.1.apkg<br\>
''Running this software on AMC 11 hardware require use of STIC'' <br>
''11.2.3.x in Master module with older software in IP slave modules will give audio problems''

== Errors Corrected ==
=== {{Bugzilla|532}} {{Bugzilla|441}} Configuration of IP only node - XE1 in node routing ===
When connecting XE1 in AlphaNet with nodes having analogue stations the XE7/20/26 routing tables should be configured with the "IP only" flag in AlphaPro to force duplex control in local node when talking to XE1.

=== $[[COPY_MAIL]] with Lower Priority ===
When lowering priority with $CPYM the station with the mail in queue was not informed thus when lowering from urgent feature reminder to no feature reminder, the urgent feature reminder continued on the station.
=== {{Bugzilla|534}} {{Bugzilla|597}} IP Station as MCS Station ===
Problems with some dedicate MCS ([[MCS_station]]) functions when using an IP station.
=== IP Station Volume Level Adjusted from IP Station Volume Control now Stores Same Values as used by AlphaPro ===
Previously "in between" values compared to AlphaPro volume values could be stored when changing volume from the IP station, thus no volume was shown in AlphaPro after get all. 
=== {{Bugzilla|581}} Try to Avoid Trans Coding in Calls from SIP Station to SIP Station ===
Previously the initial codec used for the call initiating SIP station was not stored thus the setup to the destination SIP station would not use the initial codec as preferred.
=== {{Bugzilla|569}} The OPC Server did not Receive the Correct Mail Information from Remote Nodes ===
The $[[EXT_MAIL]] from a node with sending station of the mail from another nodes did not contain a correct source "net obj ref"
=== {{Bugzilla|533}} Reset when Dialing 7984 when "Program On" ===
Fixed reset when 7984 activated from DAK key during program on.
=== {{Bugzilla|544}} $CALL setup with call option flag 1 terminate after 15seconds in AlphaNet ===
Using call option 1 for [[CALL_SETUP | $CALL]] setup in AlphaNet terminated call from start node within 15 seconds.
=== {{Bugzilla|476}} Volume Of IP stations after Restore of Backup ===
The volume of IP stations got a invalid value after restore of backup.
=== {{Bugzilla|499}} Feature Reminder Speed of IP station ===
The feature reminder led blink speed is now the same as for analogue stations.
=== {{Bugzilla|502}} IND Message with station 0 ===
The programming of "IND 0 0 0 0 OFF" from the event handler would reset the exchange.

=== {{Bugzilla|443}}  Call Timer for Transit Calls to RMD ===
To avoid endless transit calls between two Ring Master exchanges, the standard call timer is started in the last transit exchange.
=== {{Bugzilla|329}} Avoid crossing of AUDIO_PATH_DISCONNECT ===
Crossing AUDIO_PATH_DISCONNECT messages between SIPD and AlphaCom generated SysLog noise.
=== {{Bugzilla|432}} Parsing of DP messages ===
In Simple Link Layer (and from EVH) the string "$1" would reset the exchange.
=== {{Bugzilla|427}} AMC reset when alarm message is sent to active speaker in SX ===
=== {{Bugzilla|482}} Node Number not Updated after Restore of Backup ===
The node number is stored on the STIC, this was not updated after restore of a backup.
=== {{Bugzilla|472}} Echo Canceler Active During Private Ringing ===
The sound of the private ringing tone could be muted by the Echo Canceler in the IP station.

==Enhancement==
=== High Availability ===
*Information to be updated
=== IP-ARIO support ===
Support of [[IP-ARIO]].
* DAK as RCI
* Logical RCI
* Logical RCO
* Remote RS232

New and changed event handler macros
 [[wudd]]        (Added label and string variables)
 [[%udd]]        (Added label and string variables)
 [[IND_-_Indicator_control_for_CRMIV_LEDs,_buzzer_etc. |gind]]        (NEW  group version of ''ind'')
 [[%paf]]        (NEW, PA fault in UDP group check)
 [[%pai]]        (NEW, PA fault indicator in DAK block)
 [[%pap]]        (NEW, PA fault mail priority)
 [[%prci]]       (NEW, Current Physical RCI state)
 [[%prco]]       (NEW, Current Physical RCO state)

$DP messages:
 $[[PA_CMD]]     (NEW, PA command, ACK/CLEAR)
 $[[SET_IPRCO]]  (NEW, Set physical RCO on IP station/IP-ARIO)
 $[[CANCEL_MAIL]](Added option for delete mail with specific priority, Mail-id = 253)

=== Support of IP Dual Display ===
DAK display control of the IP Dual Display.

=== Recording ===
Support for [[STENTOFON_Recording]].

=== New Event Handler Macro ===
*[[%rdir]] Find directory number of a physical number.
*[[%rphy]] Find physical number of a directory number.

=== {{Bugzilla|479}} Feature From Idle Event Trigger ===
A new event trigger [[Feature_From_Idle_(Event_Type)| 37 Feature From Idle]] trigger "ON" event related to "feature directory number" of the feature code dialed from the owner station or UDP group.
=== Added Timeout to the AlphaNet Broadcast Forward Table ===
When connecting AMC9 and AMC_IP the broadcast numbering could lead to problems with broadcast messages from AMC9.

=== $[[VOL]] Without Volume Parameter Refresh Current Volume Setting on the Station ===
Handy when using %gnv - $snv from event handler to manipulate volume of stations, (nvram volume value does not correspond to the $VOL parameter)

=== Added Connect to Group Message ===
$[[CONN_2_GROUP]] This DP message allow the initiator of a group call to also receive the audio from the group call. Used for Special stations with separate mic and speaker and for production test of IP-ARIO.

=== Default use 20ms RTP packages in Multi Module audio ===
The package size used for multi module is based on the setting of RTP package size for the routing entry to the local node.
This value is default to 20ms but after a SendAll from AlphaPro the value is set to 0. This gave 10ms package size, this is now changed to be interpreted as 20ms.

=== {{Bugzilla|459}} RTP Stream Station ===
Implementation of a dummy IP station configured from AlphaPro. Can be used to stream audio to and from the exchange.
Use [[VOIP_AUDIO_CONFIG|$VAC]] to configure IP address and port to be used.

=== {{Bugzilla|434}} [[CALL_SETUP |$CALL]] with "Listen" Option ===
Possible to start a silence call to an IP station. No LED or loudspeaker turned on the B party.

=== Statistics SysLog ===
*When Calling a Trunk the private ringing and busy state was missing in statistic log.
*Last received reply code from SIP during outgoing trunk calls is stored in the Call Statistic log: "S180" 
=== Added 10 Minutes Timeout to OPC Server Registration in AlphaCom ===
After a OPC server was brief connected to AlphaCom the OPC and node address was reserved until next reset. (Not possible to connect OPC through other nodes) 
Now the OPC address is free again if the OPC has been disconnected for 10 minutes.

=== {{Bugzilla|109}} Faster {{AlphaWiki|Scheduler (Event Type)}} catchup ===
Catchup one minute each 100ms (was each 1 second).

==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=
<br\>
==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.<br\>

'''Version 01.16  (2011-03-22)''' <br>
Minor change to decrease RX-buffer delays.

'''Version 01.15  (2011-03-08)''' <br>
*Added linebuffering on command socket from AMCD. Hopefully it fixes{{Bugzilla|503}}.

'''Version 01.14  (2010-11-04)''' <br>
*Implemented rxr command, allowing different rtp-payloadtype in RX and TX direction (used for recording).

'''Version 01.13  (2010-02-18)''' <br>
* {{Bugzilla|324}} Several improvements in order to handle reception of discontinuous transmission streams. 

'''Version 01.12  (2010-02-11)''' <br>
*  Added source IP address of last received packet on "stats" reports on UPD channels

'''Version 01.11  (2010-01-07)''' <br>
* Various bugfixes on "txtap" command (for recall).

'''Version 01.10  (2009-12-08)''' <br>
* txtap command made child-program-based (start of recall support).

'''Version 01.09  (2009-11-06)''' <br>
* {{Bugzilla|164}} rfc4733 (rfc2833) DTMF in RTP support.

'''Version 01.08:''' 2009-09-30 <br>
* General interface for plugin childprogram, support for ASVP functions.

'''Version 01.07:''' 2009-08-12<br>
* Preliminary support for playout of wavfile, via plugin childprogram

'''Version 01.06:''' 2007-12-10<br\>
Description: <br>
* Support of [[Line Echo Cancellation]] (by using DSP#2 for LEC, infrastructure added for controlling DSP#2)

'''Version 01.05:''' 2007-11-08<br\>
Description: <br>
* Optimised socket handling for unicast'ed groupcalls.

'''Version 01.04:''' 2007-10-17<br\>
Description: <br>
* G.729, first experimental support (no support for DTX, lost or reorderd packets).

'''Version 01.03:''' 2007-10-11<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. 

'''Version 01.02:''' 2007-03-23<br\>
Description: Released version.  <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.
'''Version 01.01:''' 2007-03-05<br\>
Description:<br\>
* 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\>
Description: Released version.  <br\>
Introduced in initial release<br\><br\>

==SIPdaemon==
 /opt/amc/bin/sipd
[[AlphaCom_SIP_interface|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).<br\>

'''Version 01.42  (2012-08-29)''' <br/> 
*  Skip updating the register DB on REGISTER refresh (no change in contact or expires. Reset register timeout when reading DB at startup. Reduces writing to flash.

'''Version 01.41  (2011-11-30)''' <br/> 
* Fixes related to IP high availability and no SIP stations configured.

'''Version 01.40  (2011-11-22)''' <br/> 
* Avoid disconnect of setup from AlphaCom during "Early Media" situations
* Better handshaking between AMCd and SIPd during termination of SIP station call.

'''Version 01.38  (2011-07-26)''' <br/> 
* Allow incoming call from SIP trunk related AlphaCom configured SIP station. (AlphaCom SIP station related to trunk)

'''Version 01.37  (2011-07-01)''' <br/> 
* Support for High Availability IP. Dont open the registrar database when suspended.

'''Version 01.36  (2011-04-13)''' <br/> 
* Support audio level negotiation in AlphaNet ( AMCD 11.2.2.3 or newer)

'''Version 01.35  (2011-02-21)''' <br/>
* Redesign registrar. Better performance when a large number of SIP stations do registration.

'''Version 01.33 and 01.34: X versions, not released''' <br/>

'''Version 01.32  (2010-11-10)''' <br/>
* Send BYE on active sessions if the main application terminates.
* Reject SIP requests with 481 if To header with tag not matching a dialog 
* Re-INVITE: don't process SDP message if the o= version number has not changed 

'''Version 01.31  (2010-10-29)''' <br/>
* {{Bugzilla|463}} Rejected Incomming INVITE do not release "call". All calls blocked after 100 rejects. 

'''Version 01.30  (2010-10-21)''' <br/>
* Fixed showstopping bug for trunked SIP

'''Version 01.29  (2010-10-12)''' <br/>
''Don't use! Outgoing call to SIP trunk with fixed IP not working''
* Incomming INVITE matching to registrar trunk: look up using to Via, not From.

'''Version 01.28  (2010-09-30)''' <br/>
* Support for specifying domain in From header of outgoing INVITE, for Trunks
* Incomming INVITE matching to trunk: look up using to Via, not From.

'''Version 01.27  (2010-09-07)''' <br>
* Support for recording function: Allowing recorder to register, dual RTP streams. 

'''Version 01.26  (2010-04-14)''' <br>
* {{Bugzilla|353}} Incomming SIP INVITEwith audio and video in SDP fails

'''Version 01.25  (2009-11-06)''' <br>
* {{Bugzilla|164}} rfc4733 (rfc2833) DTMF in RTP support.

'''Version 01.24  (2009-10-05)''' <br>
* {{Bugzilla|221}} Handle 301 and 302 redirections. Also route by request URI for incomming calls, not To header. 

'''Version 01.23  (2009-09-03)''' <br>
* {{Bugzilla|242}} Buffer overflow constructing To: in Invite 
* {{Bugzilla|243}} Infinite loop in json-c library
* If no sip registrar node, only trunk nodes: set  default_node to first trunk node. This makes sip-trunk nodes appear in AlphaWeb (broken since SIPD 01.19).

'''Version 01.22  (2009-08-18)''' <br>
* Added support of configuration of SIP PORT (5060)

'''Version 01.21  (2009-08-04)''' <br>
* {{Bugzilla|223}} , Segmentation fault when Display name over 20 byte received from SIP

'''Version 01.20  (2009-06-22)''' <br>
* Various fixed in  "sip-as-station" JSON interface

'''Version 01.19  (2009-06-04)''' <br>
* [[SIPD]]: '''use [[Class of service]] 15, not 16, for incoming trunk calls '''
* {{Bugzilla|195}} Event trigger feature 52 or 85 from SIP phone doesn't cancel
* {{Bugzilla|205}} Added "Voice help wanted" and "Allowed to page absent user" bits to ingoing AUDIO_PATH_SETUP payload,
* JSON based interface for incoming INVITE and REGISTER, for "sip-as-station", first alpha version

'''Version 01.18  (2009-03-30)''' <br>
* Minor corrections, avoiding some unwanted syslog messages 

'''Version 01.17  (2009-02-09)''' <br>
* Handling of [[ABSD_Billing#AudioCodes_X-detect|AudioCodes_X-detect]]
* Fixed crash (in alpha_sdp_parse) when receive INVITE with no SDP body 

'''Version 01.16  (2008-12-18)''' <br>
* Introduced JSON based interface towards AMCD. Various cleansups.

'''Version 01.15 (2008-11-05)''' <br>
* {{Bugzilla|85}} Sipd reset when not resolving hostname fails 
* sipd do not require a node definition in AMCD  to start (  /tmp/sipd_config )

'''Version 01.13 (2008-06-29)''' <br>
* Send and receive INFOs for DTMF signals A - D
* Filter on '+' in directory number

'''Version 01.12 (2008-01-03):''' <br>
* Outgoing INVITE, early cancel: Cancel before "180 Ringing" caused lockup, because of incorrect check for "SIP-dialog". 
 
'''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.

'''Version 01.10 (2007-10-12):''' <br>
* CANCEL of outgoing INVITE: Send CANCEL, not BYE 

'''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). 

'''Version 01.08 (2007-08-30):''' <br>
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):'''<br>
Description: Released version, date 2007-06-01.<br>
Introduced in system upgrade file: alpha_sys_10_21x0604.tbz2<br>
* Removed memory leaks, increasing stability (issue 3176).
* Issues 3203, 3182. 
* Debug error messages forwarded to syslog. 
* Set scheduling priority.

'''Version 01.05:'''<br>
Description: Released version.  <br>
Introduced in system upgrade file: alpha_sys_10_20.tbz2<br\><br\>

==HA IP daemon==
 /opt/amc/bin/haipd

Haipd is the process which handles [[High Availability IP (Redundancy)]]. 

'''Version 01.05c:''' 2012-07-26<br\>
* Added reporting of current state for connection to related HA node. (27 - System Event reporting)
* Changed priority of the haipd after amcd start-up.
* Better check when HA config is deleted from AlphaWeb.

'''Version 01.05:''' 2011-11-30<br\>
* {{Bugzilla|688}}: Database replication broken in previous version. Two other problems also fixed.

'''Version 01.04:''' 2011-09-22<br\>
* {{Bugzilla|647}}: Cleanup of takeover-IP at startup after a haipd-crash did not work, was broken in version 1.01. Haipd aborted without cleanup if socket send failed. 
* {{Bugzilla|642}}: Reduced log chatiness when HA not configured.

'''Version 01.03:''' 2011-09-06<br\>
Don't remove /tmp/haipd/suspend_ip before operational address is actually installed.

'''Version 01.02:''' 2011-09-02<br\>
If not HA configured, do gratuitous ARP on interfaces at startup. 

'''Version 01.01:''' 2011-08-19<br\>
Description: {{Bugzilla|621}}, {{Bugzilla|627}}: Ensure that outgoing connections and packets get the operational address as source address.  <br\>

'''Version 01.00:''' 2011-07-06<br\>
Description: Released version.  <br\>

==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. <br\>

'''Version 02.11:''' (2011-03-21)<br>
Number of internal DSP channels increased, DSP SW version 01.19, {{Bugzilla|517}}. 

'''Version 02.10:''' (2008-01-16)<br>
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): <br>
*Required for [[Line Echo Cancellation]] ("disable" command, used to free DSP power for LEC).

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

'''Version 01.01''' (2007-12-10): <br>
* Same as 02.01, but for 2.4 linux kernel

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

==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.<br>

'''Version 01.20:''' (2013-04-04)<br>
AGC: If limit kicked in, level was reduced by 17dB. Now only reduce by 1.6dB as a minimum.

'''Version 01.19:''' (2011-03-21)<br>
Number of internal DSP channels increased, Need dsp_drv version 01.19, {{Bugzilla|517}}. 

'''Version 01.18:''' (2010-04-22)<br>
03xx mixer resources improved. Use for Recording.

'''Version 01.17:''' (2010-02-24)<br>
Reset AGC when codec is disconnected.

'''Version 01.16:''' (2009-11-24)<br>
Generate G.711 0 dBm0 1000 Hz reference sequence, for testing.

'''Version 01.15:''' 2009-02-17<br>
Improved stability on inter-DSP serial link McBSP#1, used for [[Line Echo Cancellation|LEC]]

'''Version 01.14:''' 2008-04-21<br>
Conference mixing resources in DSP.

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

'''Version 01.12:''' 2008-03-03<br>
AGC adjusted, stability timer wait 1 sec before reducing gain.
 
'''Version 01.11:''' 2008-02-15<br>
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<br>
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. <br>
Requires dsp_drv version 02.10 / 01.10

'''Version 01.06:''' 2007-12-10<br>
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<br>
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<br\>
Description:  <br\>
* 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\>
Description: Released version.  <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)

'''Version 01.02:''' 2006-11-30<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)

'''Version 01.01:''' 2006-06-14<br\>
Description: Released version.  <br\>
Introduced in system upgrade file: alpha_sys_10_00.tbz2<br\><br\>

==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. <br\>

'''Version 01.67:'''<br\>
Description: Released version.  <br\>
Introduced in system upgrade file: alpha_sys_10_03.tbz2<br\><br\>

==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) <br\>

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

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

==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 <br\>

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

'''Version 02.00:  Board support package 03.xx (Linux 2.6):'''<br\>
Description: Released version.  <br\>
Introduced in system upgrade file: alpha_sys_10_03.tbz2<br\>
<br\>
<br\>
<br\>
=Hardware Versions=
<br\>
==AMC hardware versions==

=== Known problems AMC hardware 8000/4 === 
----
None as of now<br\><br\>

=== 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. <br\>

=== Issue 2741: Redundancy control from APC: ===
The redundancy control system from APC is not working (software and hardware).<br\>

=== 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 will be fixed in future hardware version, but it can be fixed on current hardware by a minor modification. Remove R698 and R695. The pins must be connected to Vcc (3,3V). <br\>

=== Issue 2806: “Temperature Alarm" in ACE.7: ===
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.<br\>