AlphaCom 11.00 - 11.01 - Release Notes

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Next Release - AlphaCom 11.02 - Release Notes

This document provides the release notes for AlphaCom 11.1 with incremental bug fix releases. The release notes for AlphaCom 11.1 describe new features, improvements and issues fixed after AlphaCom 10.6.

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

AMC 11.1.3.6(2011-03-14)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 11.00.

System upgrade file:
alpha_sys-11.1.3.6.apkg
Running this software on an AMC 11 hardware require use of STIC

Errors Corrected

BZ 498 Transfer of Outgoing Billed Trunk Calls

Outgoing billed trunk calls being transferred before billing of the conversation had started would either stop billing or disconnect the conversation after 60 second. If transfer of a "billed" call was done before the acknowledge of started conversation sent to the billing system the call was not billed. To make a solution for this issue AlphaCom sends a "billing start" signal when the transfer is done. When preforming a transfer to an IP station or SIP station from an outgoing SIP trunk call the new conversation was set to duplex mode. This is now updated to select correct conversation mode dependent on type of the new A station.

BZ 503 Missing Audio During Message to Group for IP Stations

When using multipled unicast to IP stations there could be cases where some IP stations of the group did not receive audio.

BZ 517 LEC (Line Echo Cancellation) In Multi Module

The LEC system was not set up correctly in the slave module when using IP multi module audio. All type of "open duplex" conversations would be effected. Calls with handset to stations in master, AlphaNet or SIP could give echo.

BZ 525 ODC to Group, Limited Number of Members

The original AlphaCom code was restricted to 6 stations included from a group. This is now increased to 16 stations.

BZ 526 Alternative Routing of Audio

The system of Alternative routing using secondary routing was not working correctly.

New Features

Support for Line Check on ATLB12 cards

Line check/test is now supported with new software on the ATLB12 cards.(ATLB 5.51)

AMC 11.1.3.5(2011-01-10)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 11.00.

System upgrade file:
alpha_sys-11.1.3.5.apkg
Running this software on an AMC 11 hardware require use of STIC

Errors Corrected

BZ 452 Audio Disappear in One Direction During AlphaNet Call

If an IP station in conversation is called from a remote node the microphone of the station in conversation will be blocked when the new caller terminates the busy call.

BZ 480 IP Master LED continues to flash after all mails are deleted

A group of IP stations receives high priority mail and the mails are deleted from one of the stations in the group. The mails will be deleted but feature reminder LED will continue flash on all but the station who deleted the mail.

AMC 11.1.3.4(2011-01-04)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 11.00.

System upgrade file:
alpha_sys-11.1.3.4.apkg
Running this software on an AMC 11 hardware require use of STIC

Errors Corrected

BZ 365 IP Station Error Reported when Joining SX Conference

Sometimes when an IP desktop is joining a SX conference (820x), the station is reported as faulty (error message in display of Station 1).

BZ 440 Volume Override During $GM Playback

Volume override did not work when $GROUP_MSG was used for starting PA broadcast without gong tone.

BZ 442 Simultaneous Mute Program During Group Call and PRG

A combination of event handler macro "PRG" and activating group call with "mute program during group call" could generate reset.

BZ 452 Audio Disappear in One Direction During AlphaNet Call

In AlphaNet systems with transit nodes, old call information for a station could affect proceeding calls.

BZ 455 Event 13 During Start Up

The event 13 "station failure" was not reported OFF during restart for stations without failure prior to the restart. This gave some issues for the PrisCom systems. Now the event 13 will report OFF for all stations coming up.

BZ 462 Multi Module GroupCall with IP Stations

Multi Module Group Call initiated from slave module with only IP stations in master module reset the exchange.

BZ 468 Billing in Combination with Blind Transfer

When using Blind Transfer in a billed conversation the billing was stopped.

BZ 478 Scream Alarm and Event 23

Reset could occur in combination of IP station generating scream alarm and IP master station listen to the station in an SX conference.

BZ 481 Call Request Transfer in AlphaNet

When a call request was forwarded to the same node as the call request was generated, the call request mail message was blocked in the data routing.

BZ 486 No Mic Led in Conversation Between IP Station and slave stations

The MIC led on the IP station was not lit in conversation with stations in slave modules.

BZ 491 High Availability System Does not Work with Black AMC Boards

A test of hardware revisions compatible with the High Availability system was not upgraded with hardware revision 05 and 06 used for the Black AMC boards.

Multi Module in Combination with SIP and IP Stations

Due to missing initiation of an internal trans coding flag, calls from multi module could end up in silence due to unintentional internal trans coding setup. This depends on if previous call was trans coded toward for example SIP (G722 to G711).

Call Request Mail from RMD

Call request mail from Ring Master is now deleted when reply call established from AlphaCom.

AMC 11.1.3.3(2010-09-20)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 11.00.

System upgrade file:
alpha_sys-11.1.3.3.apkg
Running this software on an AMC 11 hardware require use of STIC

Errors Corrected

BZ 273 Conversation outgoing event in SX conference

When changing speaker in a SX conference the OFF event was not reported for the previous speaker.

BZ 274 M-key event in SX conference

The M-key event was reported on different sub event for analogue and IP stations during "talk" in SX-conf.

BZ 277 Event 13, faulty line

Physical numbers 605-668 used for IP AlphaNet/SIP etc was reported to event 13. Now removed from event report.

BZ 339 Event 9, Private Ringing during "private break in" function

The Private Ringing event was not reported ON again after "private break in".

BZ 346 Event 25, M-key sub event 10 when calling PNCI

The event 25 now report M-key as sub event 10 when calling PNCI.

BZ 362 Audio Tone Test report to OPC

The tone test state is now reported to the OPC server.

BZ 375 CRM Queue arrow disappears

The queue arrow in CRM queue was not visible on IP CRM and would disappear when queue operation preformed on analogue stations.

BZ 380 Call Request Mode when station down

If a disconnected station entered call request mode, then call request canceled, the station would start in call request mode when reconnected.

BZ 416 %1.mpri not working in event 10 and 12

Adjustments made to mail events so that %1.mpri and %1.tag always works in the ON event. OFF event will not currently be supported since the mail is deleted before the event is interpreted in the event handler thus no "mpri" information available.

BZ 425 Missing Audio in AlphaNet

Audio was cut if analog station calling IP station over AlphaNet lifts the handset. (Bug introduced 11.1.3.0)

IP stations registered with directory number

When using station configuring in AlphaWeb on IP stations configured for registration with directory number the current MAC address was saved thus locking the station to the current MAC address. Fixed.

Enhancements

Added PA subevents

Event handler triggers for recording and playback phase of group call with ReCall and message playback.

"R KEY" from SIP station calling to SIP trunk

When calling outgoing trunk AlphaCom stations can use DAK 4/8 to activate local AlphaCom features for inquiry/transfer etc. ATLB phones can use R-KEY for the same features.
For SIP stations the function is solved with a 2 digit pattern that must be dialed within a time limit.
Default the pattern for activating local features is "# #" within 500 ms.

ex_profile->glob_const.SIP_R_1   = 11    (Digit 0-9, 10 = *, 11 = #)
ex_profile->glob_const.SIP_R_int = 5     (Interval in 100ms)
ex_profile->glob_const.SIP_R_2   = 11    (Digit 0-9, 10 = *, 11 = #)

Stored Voice Message Playback During Conversation

DP message for stored voice message playback to one station of an conversation SVP_ST.

New Call Progress Event

SIP call progress event trigger for outgoing calls to SIP trunks.

Labeling of IP station function keys

It is now possible to label function keys in idle using the DAK string macro "T",

• IP station must be configured as station type: Master Station
• In AlphaPro label the function keys in the Users & Stations -> DAK -> Navig. keys. Example: "I84 TGrpC"

SysLog on TST

See TST#SysLog. Previously, some Syslog messages was printed to TST. Now no Syslog are printed by default, but by typing the "syslog" you can get all AMCD syslog messages above the specifed level.

AMC 11.1.3.1(2010-07-28)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 11.00.

System upgrade file:
alpha_sys-11.1.3.1.apkg
Running this software on an AMC 11 hardware require use of STIC

Errors Corrected

BZ 415 Audio Program missing on XE7

Audio Program failed on XE7 using ASLT as program input. This fault is only found on 11.1.3.0

AMC 11.1.3.0(2010-07-15)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 11.00.

System upgrade file:
alpha_sys-11.1.3.0.apkg
Running this software on an AMC 11 hardware require use of STIC

Errors Corrected

BZ 356 Global Messaging and Recall

Remote dialing into a local ReCall group call did not turn on microphone on initiator station (no sound).
Message to global group with the option set for return to standard group call after message end resulted in local feedback of group call audio in initiator station.

BZ 357 Feature Reminder IP station

The feature reminder blink sequence was not equal with analogue stations.

BZ 360 BZ 341 Event Handler IF statement

The event handler IF statement only evaluate numeric values thus IF statement on text evaluate faulty. A %strlen() operator added for evaluation if text string is empty.

BZ 395 Low volume when IP station initiate ReCall

Loudspeaking IP stations was using echo canceler during recording of ReCall message thus variable volume.

Transit call to SIP

Calling via transit node to SIP would reset private ringing after 6 second if not SIP OK received from SIP.

New Ring Master Daemon allows more than 6 simultaneous calls

The Ring Master integration Daemon could do an reset if more than 6 simultaneous calls was activated between AlphaCom and Ring Master.

Camp On busy when calling busy Ring Master station

When calling a busy Ring Master station from AlphaCom the AlphaCom will now give camp on busy. Works also for transit calls between two Ring Master system via AlphaCom.

BZ 405 and BZ 407

(These bugs was introduced on X version 11.0.9.4 thus never been released.)
SIP client call causes IP stations to loose connection. Missing audio on AlphaNet Call from IP Station to Analogue station

Enhancements

STIC Stentofon Identity Card

AMC running on AMC 11 hardware now supports the STIC.
Stored on STIC:

• Basic IP configuration
  • Eth0 and Eth1 IP addresses and mask
  • Default gateway
  • Filter setting for AlphaWeb and AlphaPro
• Node number
• License string
  

In the case moving STIC from one card to another InterCom configuration and other AlphaWeb configiraton must be restored with the use of the AlphaWeb Backup system.

SoftClient License

SoftClients now require use of a special SoftClient License. License string is increased from 24 to 36 bytes if SoftClient license is include. System still compatible with the 24 bytes license.

Billing PTR trunk

Trunk busy reported to Billing system, Billing can activate alternative trunk. Trunk "break out code" configurable in the AlphaCom database.

Volume override related $DP messages

Added new DP commands related to Volume Override.

$CAC Cancel all active calls
$MVO Mask volume override

AMC 11.0.3.1(2010-05-03)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 11.00.

System upgrade file:
alpha_sys-11.0.3.1.apkg

Errors Corrected

Priority of playback of voice messages

The priority of playback of defined message:

• 1. Recorded message from station
• 2. Uploaded message from AlphaWeb
• 3. ASVP message

Priority of IP station audio playback

When more than 1 audio source available:

• 1. Group Call
• 2. Program/conference distribution
• 3. Conversation

Missing audio in IP multi module

Conversation from IP stations/SIP stations to a slave module stations could have corrupted audio if the IP/SIP stations had previously been used in a trans coded IP conversation.

Voice Messaging "Meet Mee"

The meet me(99) system for messaging is changed to inhibit use of 99 for
ReCall and SVP messages from start. If control bit 2 (MeetMe) are enabled 99 is
activated from gong (or gong check if no gong). This will reject all 99 during
ReCall recording and for 99 during messaging function with MeetMe not enabled.

AMC 11.0.3.0(2010-04-22)

Release: Official, available on request
/opt/amc/bin/amcd
NVRAM version 11.00.

System upgrade file:
AMC card version 1-4 must be image upgraded with axeu_b2200_a_1100300.aimg
AMC card version 5 must have image axeu_b3000_a_1100300.aimg

Errors Corrected

BZ 333 Busy override not working from ATLB stations

Allow override features to be dialed from ATLB stations during private ringing or busy tone without use of R-key.

BZ 340 ODC to group failed

Setting up ODC to a group failed, only some of the participants where added.

BZ 275Display text from $SM RingingGroup to SIP Phones

Display text sent to SIP stations was only the sending station display text. Now custom text in the $SM command is used.

BZ 353 Incoming SIP INVITE with audio and video in SDP fails

SIPD fails to parse a Audio offer when the audio offer is followed by a video offer.

Enhancement

Integrated Support of Messaging features

Integrated ASVP support

The voice system from ASVP card can now be installed as a software package on system with the AMC XE hardware.
Require enhanced messaging license.

Recall, Delayed Group Playback

A system of delayed playback of group call is now available on both AMC new and old AMC IP hardware.
Require basic messaging license.

AutoAttendant

A voice menu system can be configured as a feature in AlphaCom. AMC XE hardware only.
Require basic messaging license.

Message Playback

Custom stored messages can be played back to local or global groups. Messages is uploaded from AlphaWeb. AMC XE hardware only. Require basic messaging license.

New Backup System

A more complete system backup is available from AlphaWeb.

New Event Handler features

New LOOP/ENDLOOP/BREAK function in event handler
New %ges Group Exclusion station macro in event handler.
New %dfmt : Convert decimal value to hex, octal or ASCII

Audio UDP port test

UDP ports used for IP audio is tested in idle to discover "spamming" from faulty equipment.

AlphaWeb Station Configuration

Basic setting for stations can now be configured from AlphaWeb.
Currently supported for stations in AlphaWeb:

• Directory number
• Display Name
• Volume
• Noice reduction level for IP stations.

Avoid attention tone when automatic seach on busy is activated

ex_profile.flags.no_att_tone_search = 0 

Change tone on call requester mode

ex_profile.glob_const.CRM_tone = 0 
0 = Private ringing
1 = Lo pri feature reminder bleep
2 = Hi pri feature reminder bleep

Adjustable delay before ID signalling is accepted from substation

ex_profile.timeouts.ID_C_delay = 3 (in 100ms interval)

Stentofon IP station easy configuration

The IP stations can now be configured with an open MAC address from AlphaPro (11). This means that the IP stations will do registration only with their directory number.
The IP station menu and WEB are updated with entries for adding wanted directory number. Only IP stations off line in AlphaCom will be allowed to register by directory number.
This will ease replacement of stations without involving AlphaPro.

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 01.16 (2011-03-22)
Minor change to decrease RX-buffer delays.

Version 01.15 (2011-03-08)

• Added linebuffering on command socket from AMCD. Hopefully it fixesBZ 503.

Version 01.14 (2010-11-04)

• Implemented rxr command, allowing different rtp-payloadtype in RX and TX direction (used for recording).

Version 01.13 (2010-02-18)

• BZ 324 Several improvements in order to handle reception of discontinuous transmission streams.

Version 01.12 (2010-02-11)

• Added source IP address of last received packet on "stats" reports on UPD channels

Version 01.11 (2010-01-07)

• Various bugfixes on "txtap" command (for recall).

Version 01.10 (2009-12-08)

• txtap command made child-program-based (start of recall support).

Version 01.09 (2009-11-06)

• BZ 164 rfc4733 (rfc2833) DTMF in RTP support.

Version 01.08: 2009-09-30

• General interface for plugin childprogram, support for ASVP functions.

Version 01.07: 2009-08-12

• Preliminary support for playout of wavfile, via plugin childprogram

Version 01.06: 2007-12-10
Description:

• Support of Line Echo Cancellation (by using DSP#2 for LEC, infrastructure added for controlling DSP#2)

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 01.37 (2011-07-01)

• Support for High Availability IP. Dont open the registrar database when suspended.

Version 01.36 (2011-04-13)

• Support audio level negotiation in AlphaNet ( AMCD 11.2.2.3 or newer)

Version 01.35 (2011-02-21)

• Redesign registrar. Better performance when a large number of SIP stations do registration.

Version 01.33 and 01.34: X versions, not released

Version 01.32 (2010-11-10)

• 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)

• BZ 463 Rejected Incomming INVITE do not release "call". All calls blocked after 100 rejects.

Version 01.30 (2010-10-21)

• Fixed showstopping bug for trunked SIP

Version 01.29 (2010-10-12)
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)

• 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)

• Support for recording function: Allowing recorder to register, dual RTP streams.

Version 01.26 (2010-04-14)

• BZ 353 Incomming SIP INVITEwith audio and video in SDP fails

Version 01.25 (2009-11-06)

• BZ 164 rfc4733 (rfc2833) DTMF in RTP support.

Version 01.24 (2009-10-05)

• BZ 221 Handle 301 and 302 redirections. Also route by request URI for incomming calls, not To header.

Version 01.23 (2009-09-03)

• BZ 242 Buffer overflow constructing To: in Invite
• BZ 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)

• Added support of configuration of SIP PORT (5060)

Version 01.21 (2009-08-04)

• BZ 223 , Segmentation fault when Display name over 20 byte received from SIP

Version 01.20 (2009-06-22)

• Various fixed in "sip-as-station" JSON interface

Version 01.19 (2009-06-04)

• SIPD: use Class of service 15, not 16, for incoming trunk calls
• BZ 195 Event trigger feature 52 or 85 from SIP phone doesn't cancel
• BZ 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)

• Minor corrections, avoiding some unwanted syslog messages

Version 01.17 (2009-02-09)

• Handling of AudioCodes_X-detect
• Fixed crash (in alpha_sdp_parse) when receive INVITE with no SDP body

Version 01.16 (2008-12-18)

• Introduced JSON based interface towards AMCD. Various cleansups.

Version 01.15 (2008-11-05)

• BZ 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)

• Send and receive INFOs for DTMF signals A - D
• Filter on '+' in directory number

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 02.11: (2011-03-21)
Number of internal DSP channels increased, DSP SW version 01.19, BZ 517.

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):

• 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):
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 01.19: (2011-03-21)
Number of internal DSP channels increased, Need dsp_drv version 01.19, BZ 517.

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

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

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

Version 01.15: 2009-02-17
Improved stability on inter-DSP serial link McBSP#1, used for LEC

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

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None as of now

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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 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).

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.