AlphaCom 10.xx - Release Notes
From Zenitel Wiki
Software in production: AMC 10.20<br\> Software released date: 2007-04-12<br\> Note 1: We sometimes do bug fixes in older versions while working with a new version. You can’t read the list version by version, and always assume that a correction is included in the next. Be aware of high release numbers, e.g. 06.05 vs. 07.01. Check version dates, and also comments for each version. <br\> Note 2: For each software version the NVRAM version is listed. If the NVRAM version is different, the AMC board must be cold started, and then you must do a SendAll from AlphaPro to restore the configuration.<br\><br\><br\>
Contents
AlphaCom 10.xx Release Notes
<br\>
AMC 10.20 (2007-04-12)
Release: Official, available on request /opt/amc/bin/amcd NVRAM version 10.20.
System upgrade file:<br\> alpha_sys_10_20.tbz2<br\>
Precautions:<br\> alpha_sys_10_00.tbz2 must be installed.<br\><br\>
Functional changes / Enhancement
Private ringing tone on Call Request function<br\> Private ringing tone on mail at receiver station will be active when first mail in queue has the mail priority above 150 (default value of "globel_constant>priv_ring_mail_pri")<br\> New event FEAT_M_KEY (31 )<br\> ON = M key pres<br\> OFF = M key release<br\> Sub event 0 will give M key status when station is busy.<br\> Sub event 1 will give M key status when station is idle.<br\> New event FEAT_OFF_HOOK (32)<br\> ON = OFF_HOOK<br\> OFF = ON_HOOK<br\> Sub event 0 will give HOOK state when station is busy. <br\> Sub event 1 will give HOOK state when station is idle.<br\> Only state changes within current state is reported on M_KEY and HOOK, it means if the station goes OFF_HOOK from idle the event OFF_HOOK in busy state will not be reported.<br\><br\>
Errors Corrected
Issue 3114: Multi Module IP audio fails after slave reset:<br\> When slave is reset, without the master being reset, the IP ICC audio links does now work RTP jitter buffer improvements. <br\> Rtpdeamon version 01.02 improves the jitter buffer handling. Both handle larger difference in clock rate between sender and receiver, at the same more stable delay adaptation at the presence of jitter. Fixes issue 3101 with Xlite.<br\> <br\> <br\> <br\>
AMC X10.20 (2007-03-20)
Release: Official, available on request /opt/amc/bin/amcd NVRAM version 10.20.
System upgrade file:<br\> alpha_sys_10_20x0320.tbz2<br\>
Precautions:<br\> alpha_sys_10_00.tbz2 must be installed.<br\><br\>
Errors Corrected
Issue 3179: Calling from a slave to an E1 node:<br\> It was not possible to dial global numbers from a slave module in multi module over IP to a different node connected with the master over AE1. (10.20x bug)<br\> <br\> <br\> <br\>
AMC X10.20 (2007-03-15)
Release: Official, available on request /opt/amc/bin/amcd NVRAM version 10.20.
System upgrade file:<br\> alpha_sys_10_20x0315.tbz2<br\>
Precautions:<br\> alpha_sys_10_00.tbz2 must be installed.<br\><br\>
Functional changes / Enhancement
Ring Master Daemon is included in the package. See separate documentation.<br\>
Support of IP stations.<br\>
- - Substation functionality
- - Group Call is available
- - Outputs on the stations can be related to RCO in AlphaPro. (RCO type station)
- - Inputs can be related to DAK or directly to DAK_AS_RCI event in AlphaPro.
See separate documentation.<br\>
IP audio transit capacity is increased from 32 to 64 channels. Transit audio is audio not going trough the backplane. (SIP to SIP, IP station to IP station, AlphaNet IP transit or a combination of IP audio)<br\>
AlphaWeb – AlphaNet information<br\> Information page now contains node name and node software versions.<br\><br\>
Errors Corrected
Issue 2885: Log enabling:<br\> System log to Syslog is now working even when log port is not enabled in AlphaPro.<br\>
Issue 2935: SIP Softphone ExpressTalk causes AMC reset:<br\> SIP Softphone ExpressTalk would reset the exchange when the conversation is cancelled from the phone (both incoming and outgoing calls). <br\>
Issue 3053: Conversation Outgoing event:<br\> When making a call from intercom out on a SIP gateway (Mediatrix 2400) the event Conversation Outgoing is now triggered when the phone answers.<br\>
Issue 3154: $CALL in AlphaNet don't work:<br\> The data protocol command $CALL is now working in AlphaNet.<br\>
Issue 3169: Serial communication block AMCD:<br\> AMC would not start after power reset if serial communication was active during reset. (Other node sending data to resetting AMC) This will not happen on system running Linux 2.4. <br\>
Issue 3180: Program 7 has no display txt:<br\> Audio program 7 (feat 5/7): The display text was not shown in the display when activating the audio program 7. For other programs it is ok. (I have seen this long time ago also, so it might be an old bug). Static License of 2 AlphaNet audio links does not work correctly. Only one audio link was working. Problems with SIP and Codec Selection <br\> SIP Preferred codec selection configured in AlphaPro does now also filter on incoming audio stream. This cure some problems regarding codec mismatch between AlphaCom and SIP.<br\> <br\> <br\> <br\>
AMC 10.05 (2007-02-07)
Release: Official, available on request /opt/amc/bin/amcd NVRAM version 10.00.
System upgrade file:<br\> alpha_sys_10_05.tbz2<br\>
Precautions:<br\> alpha_sys_10_00.tbz2 must be installed.<br\><br\>
Functional changes / Enhancement
License keys:<br\> SIP trunk and SIP stations is now separated in two different licenses.<br\> SIP trunk and AlphaNet licenses is by default dynamic except for the 2 line AlphaNet license that still is static.<br\> No audio routing programming is needed for SIP trunk or AlphaNet IP (except for 2 licenses).<br\> AlphaNet and Multi Module IP use dynamic licenses from the same license pool.<br\> If static routing is programmed in AlphaPro licenses for those audio lines is reserved from the license pool and can not be used for other dynamic audio links/multi module.<br\> Priority of audio resources allocation is handled by the priority of the initiator station.<br\><br\>
Errors Corrected
Issue 2715: AlphaNet Duplex in combination with AMC 8/9:<br\> In combination AlphaNet with AlphaCom 8 and 9 and exchanges via IP there will be a problem with delay adjustment of the duplex algorithm when calls are made from IP to an AMC 8/9 exchange.<br\> The duplex algorithm is now run in A node when going from an ASLT station to IP AlphaNet thus avoiding duplex in end AMC 8/9 nodes. <br\> In transit systems with certain combinations of AMC- 8/9/IP, SIP, AGA, AE1 and Multi module IP some issues could still occur that needs special configuration. (See separate duplex document). <br\>
Issue 2887/3113: SIP automatic duplex switching:<br\> When calling SIP there can be problems with the standard duplex algorithm due to DSP echo cancelling in the SIP station. A new duplex algorithm is available for duplex towards SIP stations that is speech controlled only from volume of the microphone signal from the SIP link.<br\> (exchange flag “DSP_duplex = 1”) <br\>
Issue 2897: X-lite on-hold lockup:<br\> Problems during on-hold feature in X-lite fixed.<br\>
Issue 2908: SIP-Ringing if call cancelled before answer:<br\> Calling from AMC-station to SIP phone. If C-key is pressed on AlphaCom-station before the call is answered, the SIP phone keeps ringing if calling via a transit node. <br\>
Issue 2960/3119: SIP-Handy-Tone 488 making All Call.(* and # key):<br\> SIP now handles * and # for both Grandstream and Mediatrix.<br\>
Issue 3011: Duplex switching in mixed environment:<br\> When the audio path go transit from IP to analog/E1 delay information to the automatic duplex routine is lost. Duplex delay is now forwarder/backwarded from transit AGA to IP links to the duplex node.<br\>
Issue 3012: Echo in SIP handset:<br\> When talking with handset conversation between Intergard station and SIP station the SIP station will get echo in the handset due to overhearing in AlphaCom handset (and no echo cancelling). Handset to handset communication will now be forced in duplex. (Default delay setting for SIP = 30ms. Full duplex can be obtained if parameter “max_off_hook_delay” is adjusted to 40 ms or more)<br\>
Issue 3015: AlphaNet: No Camp On Busy:<br\> There is no "camp on busy" when all AlphaNet lines are in use. Instead one gets a rejection tone. Same behaviour with AGA line, AE1 and VoIP. Feature implemented.<br\>
Issue 3017: Name list:<br\> After AMC auto load dirno's 9542 - 9545 are in the name list (614).<br\>
Issue 3034: Time management:<br\> When changing time in AlphaWeb, the new time is now also written to the hardware clock.<br\>
Issue 3051: IP address with leading 0's:<br\> Interpretation of leading 0 in AlphaWeb is fixed. <br\>
Issue 3067: AlphaWeb: Same subnet on Eth0 and 1:<br\> Configuration of both Ethernet phys on the same sub net is now tested.<br\>
Issue 3070: SIP: Long phone no on DAK/Substation:<br\> Up to 16 digit phone number now allowed in DAK "784" and Call-forward "71".<br\>
Issue 3116: AMC: Speech channel locks up:<br\> During global conference and failed SIP calls speech channels could be locked up.<br\>
Issue 3143: AMC: 99 answer in global group call:<br\> Not possible to answer global group calls from other that initiating node. This feature is fixed.<br\>
Issue 3060: SIP & AlphaNet licenses:<br\> It is now possible to install 30 AlphaNet and 20 SIP trunk licenses. <br\>
Issue 3069: SIP: Phone number in display:<br\> For outgoing/incoming calls the number of shown digits is 16 (including event handler).<br\>
Issue 3087: SIP: Mediatrix 1204 - Dial Out:<br\> Transmit digit by digit as dialled now supports:
- - Programming of phone number on DAK from AlphaPro
- - DAK, "784" from station
- - Call forwarding from station "71"
Both for Mediatrix and AudioCodes<br\>
Issue 3105: SIP Trunk: DAK call fails (AudioCodes):<br\> The first or the two first DTMF digits of the phone number is not transmitted. 400 ms delay before sending digits. Delay can be extended with exchange timeout "sip_dial_dly".<br\>
Issue 3107: SIP Trunk: Call Forward (71):<br\> Manual transfer to phone using 71 don't work (PNCI you could 71 + 0 + phone + M, or 71 + <shortnumber>). Fixed<br\>
Issue 3136: AlphaNet: Global SX Conference:<br\> AlphaNet: Global SX Conference. Problems after node reset and issues of reconnect are fixed.<br\>
Issue 3141: Call to unregistered SIP stations:<br\> Call from AlphaCom to a SIP phone, SIP phone is configured at registrar node, but the phone is not registered: Then SIP sends the INVITE to its own IP address, which is processed, and forwarded in a loop until all RTP resources are used up. IP address check implemented.<br\>
Issue 3156: AlphaWeb show no licence:<br\> AMC now generate correct infor in the license info file.<br\> <br\> <br\> <br\>
AMC X10.05 (2007-01-18)
Release: Beta, available on request /opt/amc/bin/amcd NVRAM version 10.00.
System upgrade file:<br\> alpha_sys_10_05x0118.tbz2<br\>
Precautions:<br\> alpha_sys_10_00.tbz2 must be installed.<br\><br\>
Errors Corrected
Issue 3072: $CPYM removes two char in name:<br\> When a mail is copied to another station using $CPYM L%1.dir W%2.tag L<dirno>, the two first characters in the name of the sender is removed. E.g. if the sender is "Donald Duck" it will appear as "nald Duck".<br\>
Issue 3074: 626 cancel call request:<br\> The problem is related to 626. This code blocks the station for a few seconds when you hang up.<br\>
Issue 3088: Transfer of outgoing calls:<br\> When doing an outgoing phone call from AlphaCom via SIP Gateway (tested with Mediatrix 1204 (analogue) and Mediatrix ISDN) you cannot transfer the call to another intercom station.<br\>
Issue 3115: AlphaNet Global SX Conference:<br\> Cancelling and reinitiate a global conference do not distribute audio. Also problems when resetting member nodes of a conference.<br\> <br\> <br\> <br\>
AMC X10.05 (2006-12-21)
Release: Beta, available on request /opt/amc/bin/amcd NVRAM version 10.00.
System upgrade file:<br\> alpha_sys_10_05x1221.tbz2<br\>
Precautions:<br\> alpha_sys_10_00.tbz2 must be installed.<br\><br\>
Errors Corrected
Issue 3111: Multi Module Group Call Block:<br\> MultiModule and group call could block. Fixed one reproduced case.<br\>
Issue 3112: Reset after cancel of call setup by the $CALL command:<br\> When resetting A-station in a call established by the $CALL the exchange did a reset. Fixed<br\>
Issue 3121: In hotline call off-hook action do not work:<br\> Fixed.<br\> <br\> <br\> <br\>
AMC 10.04 (2006-12-12)
Release: Official, available on request /opt/amc/bin/amcd NVRAM version 10.00.
System upgrade file:<br\> alpha_sys_10_04.tbz2<br\>
Precautions:<br\> alpha_sys_10_00.tbz2 must be installed.<br\><br\>
Functional changes / Enhancement
AlphaPro Password:<br\> The AlphaPro password is now the same as the administrator password for AlphaWeb (default user: admin, password alphaadmin)<br\>
Issue 3031: Ignor station down as default auto load:<br\> The flag is enalbled as default from AMC 10.04<br\>
Issue 3068: SIP: Inter-digit timeout:<br\> TST->Nvram “ex_profile.timeouts.dig_col_timeout” configures the inter digit timeout in 100 ms steps during collection of digits on feature 81 and 83. Default is 3 seconds.<br\><br\>
Errors Corrected
Issue 3024/3039: AlphaNet & MultiModule IP - 7872 causes reset:<br\> MultiModule over IP: Dialling 7872 on station 101 in master causes reset.<br\>
Issue 3038: EDIO - %sscan:<br\> The macro %sscan(string, search-string) returned wrong result when there was no match with the search-string.<br\>
Issue 3076/3085: Special characters in AlphaCom display text:<br\> AlphaCom display text could not include /-,. æøå etc. when making an SIP call.<br\>
Issue 3086: TST Error messages during SIP call:<br\> Error messages introduced in 10.03 during call SIP reset is now fixed.<br\>
Issue 3089: Events when not "Default User”:<br\> The eventhandler is now working for Conversation event “related to” user that are not default users.<br\>
Issue 3091: Mediatrix 1204 - No M-key */#:<br\> Key signaling of * and # from mediatrix is now implemented. Signaling of * and # out of SIPD is now also using * and # towards SIP equipment<br\>
Issue 3093: RCO PULS bug:<br\> It was possible to activate the RCO puls system by accident from the event handler scripts due to error in the script parser. (The RCO system then got inverted behavior)<br\>
Issue 3095: Context params in built in cmds, RCO:<br\> The “%chg(on,off)” command did not work correctly for rco.<br\>
Issue 3096: Mediatrix 1204, direct dial out:<br\> SIPD is now hanling collected digits from AlphaCom.<br\>
Issue 3098: Multi Module Group call on AGA lines:<br\> Group Call with AMC IP master and AGA/AE1 audio could reset the exchange.<br\>
Issue 3103: Eventhandler; left adjustment of %:<br\> Left adjustment of %macroes did not work. Example: "%1.exp(3<)"<br\> <br\> <br\> <br\>
AMC 10.03 (2006-20-11)
Release: Official, available on request /opt/amc/bin/amcd NVRAM version 10.00.
System upgrade file:<br\> alpha_sys_10_03.tbz2 This package has support of both board support package 02.xx (Linux 2.4) and 03.xx(Linux 2.6)<br\>
Precautions:<br\> alpha_sys_10_00.tbz2 must be installed.<br\><br\>
Functional changes
Pulse RCO:<br\> Support for generating a pulse of specified length on a “logical” RCO. <br\> New parameter “Duration” is added to the event handler-built-in command “RCO” and the data protocol message $SLRC. Examples, pulse RCO # 13 on for 1 second: “rco 13 on 10” or “$SLRC W13 1 w10”. <br\> Duration is in tenths of second. If Duration parameter is present, and non-zero, the RCO state specified in previous parameter lasts for the specified time. After the time has expired, the RCO is toggled to the opposite state. If a new $SLRC/RCO on the same RCO arrives while the pulse timer is running, the timer is cancelled, and the new message determines the new RCO state. Duration = 0 means infinite duration.<br\>
DAK RCI EVENT:<br\> New event Dak key as RCI (30). Sub event = DAK key number. ON = Dak key press, OFF = dak key release.<br\>
Private ringing for incomming SIP calls:<br\> New tst nvram flag available ”ex_profile.flags.private_ringing_SIP”. If ”1” then calls from SIP will use private ringing mode. If “0” then calls will connect as an intercom call with normal priority.<br\><br\>
Errors Corrected
Multi module in combination with AMC 8 or 9 reset:<br\> Incompatible tone handling fixed.<br\>
Issue 2948: Technical Alarm Inputs (RCI) on AC7:<br\> Amc software RCI supports for AMC hardware 8000/4.<br\>
Issue 3027: MultiModule and SIP:<br\> Combination of Multi Module and SIP is working .<br\>
Issue 3032: Search string on AlphaNet IP:<br\> When doing a call from node A to Node B and the station in node B has a search string active the search will now work.<br\>
Issue 3055: Slow call setup when dialling from Master to Slave:<br\> Software “bottle neck” fixed.<br\>
Issue 3067: Volume (783) and COS (7873) are not stored in flash:<br\> Volume and COS setting is now stored in nvram for survival after reset.<br\><br\>
SIPD 01.02 upgrades and corrections
Event “Conversation Outgoing” :<br\> Outgoing calls to SIP: Report Event “Conversation Outgoing” (8) ON when receiving SIP 200 OK, reports OFF at disconnect. <br\>
CANCEL_MAIL:<br\> Outgoing calls to SIP: Send CANCEL_MAIL for outgoing call, which cancels call request from that SIP phone.<br\>
Events for digits from called SIP:<br\> Outgoing calls to SIP: If called phone presses digits which are sent to AlphaCom as SIP INFO, "Event Trigger Feature" (15) is reported. The digit is sub event (0-9, * = 10, # = 11). Calling AlphaCom station is Event Owner, and called SIP phone number and node number is related to. This could be used for e.g. Dooropening.<br\>
Dialing after initial connect sent as SIP INFO:<br\> Outgoing calls to SIP: Digits dialed after initial connect are forwarded as SIP INFO digit messages to the remote SIP device. This can be used for two step dialing from AlphaCom to an external system, or to the PSTN.<br\>
Register from port different than 5060 allowed:<br\> Call to SIP: Use port-number stored in registry for contact. Allows SIP device to register from any (random) port, not just 5060. Fixes issue with XLITE 3.0.<br\>
Registrar, remove all bindings:<br\> Store only one binding for each user. Remove binding if expires==0. Fixes failure if SIP device registered again from different address/port<br\>
SDP media description:<br\> Call from SIP: handle that SDP a= missing, use m=... Fixes issue with XLITE 3.0.<br\>
Issue 3059: SIP response handling, call to SIP:<br\> Call to SIP: Improved handling of non-successful SIP responses 300->699: Forward response to AMCD followed before proper disconnect of call towards AMCD. Finally SIP ACK is now sent.<br\> This allows AMCD to handle busy SIP phone properly, and avoids retransmissions from SIP phones due to missing the ACK. Forward 1xx invite responses to AMCD, instead of sending a faked 180 ringing directly on AudioPathSetup from AMCD. Fixes issue with short ring tone when two-step dialing to a SIP gateway.<br\>
Call on Hold from SIP:<br\> When receiving a re-INVITE, respond with 488 not acceptable here. Quick fix of problem that X-lite was not able to disconnect a call if the call was put on hold by pressing other line button.<br\>
Display name:<br\> Outgoing calls to SIP: don’t send STATION_INFO to AMCD if called name not received from SIP. Fixes issue with blank upper display line for outgoing calls.<br\>
Issue 3058: Default route was required:<br\> Fixed failure in IP address lookup if default route not defined for the system, even if no routing was required.<br\>
Crash fixes:<br\> Fixed memory crash if SIP 1xx response received out of context. Fixed crash if receiving REGISTER with more than 9 digits in user name<br\>
SIP Debug/Trace:<br\> SIP trace/debug console on UNIX socket /tmp/sipd_trace. Connect using "tst -s /tmp/sipd_trace" from local shell. Possible to change trace level during runtime, press digits 0-7. Default level is 4, which prints SIP and AlphaNet messages. Press 0 to stop tracing. Formatting of message trace improved.<br\>
SIPd crash debug:<br\> If SIPD crashes due to segmentation violation, a short debug message is printed to “/var/log/sipd_crashes”. Only the last crash is recorded, to avoid filling up the file system.<br\>
Issue 3007: SIP - digit during connection:<br\> It is now possible to send digits from AlphaCom to SIP during a SIP connection.<br\>
SIPD 01.02 known limitations
SIPD do not remove expired REGISTERs. Can cause problems if changing numbers and node numbers of SIP phones. Workaround: Log in, and delete "/var/opt/amc/nvram/sip_registrar", reset exchange.<br\>
Digits during conversation can not be sent or received using RTP (RFC 2833 ). Only SIP INFO supported<br\> <br\> <br\> <br\>