H323 IP Trunk with Elastix and Avaya PB

Discussion in 'General' started by daudet, Oct 8, 2009.

  1. daudet

    Joined:
    Oct 7, 2009
    Messages:
    7
    Likes Received:
    0
    Hello all
    i am trying to get an IP trunk H.323 working between Avaya PBX and Elastix

    so far i can call from Avaya x 2270 to exten 4420 or trunk access code 730 toward Elastix and it ring once and i get Goodbye and a disconnect.

    from the Elastix with sip softphone i was able once to call 2270 and the Avaya phone rang and after answering i have no talk path.

    Now when i call 2270 i have ringback but the Avaya 2270 does not ring anymore.

    i don't know what i am missing.
    i have try numerous config found on the web but no luck so far.

    I have a second Trixbox and i have the same result. That is why i built a Elastix today thinking that it would work right off the box, but this is not the case.

    Any help is welcome. about the link and codec.

    Daniel
     
  2. hinzinho

    Joined:
    Sep 18, 2009
    Messages:
    461
    Likes Received:
    0
    I have Asterisk and Avaya talking between each other. I do know that only certain version of the Avaya works. Let me know what model of the Avaya you have and I'll find the instructions tomorrow.
     
  3. daudet

    Joined:
    Oct 7, 2009
    Messages:
    7
    Likes Received:
    0
    I have a Prologix
    System: G3csiV12 Software Version: R012i.00.1.224.0

    i also have a S8300/G700
    Software Version: R014x.00.4.739.0
     
  4. daudet

    Joined:
    Oct 7, 2009
    Messages:
    7
    Likes Received:
    0
    Voici mon 00h323.conf

    [general]
    port=1720
    bindaddr=172.16.58.111
    progress_setup=8
    progress_alert=8
    faststart=yes
    h245tunneling=yes
    gatekeeper=DISABLE
    disallow=all
    allow=ulaw
    dtmfmode=inband
    context=internal

    [definity]
    type=friend
    context=internal
    host=172.16.58.24
    port=1720
    disallow=all
    allow=ulaw
    canreinvite=no
    dtmfmode=internal

    Show channeltypes

    Type Description Devicestate Indications Transfer
    ---------- ----------- ----------- ----------- --------
    Agent Call Agent Proxy Channel yes yes no
    Phone Standard Linux Telephony API Driver no yes no
    MGCP Media Gateway Control Protocol (MGCP) yes yes no
    OOH323 Objective Systems H323 Channel Driver no yes no
    IAX2 Inter Asterisk eXchange Driver (Ver 2) yes yes yes
    SIP Session Initiation Protocol (SIP) yes yes yes
    Local Local Proxy Channel Driver yes yes no
    DAHDI DAHDI Telephony Driver w/PRI w/OPENR2 no yes no
    ----------
    8 channel drivers registered.
    The 'show channeltypes' command is deprecated and will be removed in a future release. Please use 'core show channeltypes' instead.
     
  5. daudet

    Joined:
    Oct 7, 2009
    Messages:
    7
    Likes Received:
    0
    Call from 2270 Avaya to 2525 Elastix

    Welcome to Elastix
    ----------------------------------------------------

    To access your Elastix System, using a separate workstation (PC/MAC/Linux)
    Open the Internet Browser using the following URL:
    http://172.16.58.111

    [root@elastix ~]# asterisk -rvvvv
    Asterisk 1.4.25, Copyright (C) 1999 - 2008 Digium, Inc. and others.
    Created by Mark Spencer <markster@digium.com>
    Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
    This is free software, with components licensed under the GNU General Public
    License version 2 and other licenses; you are welcome to redistribute it under
    certain conditions. Type 'core show license' for details.
    =========================================================================
    == Parsing '/etc/asterisk/asterisk.conf': Found
    == Parsing '/etc/asterisk/extconfig.conf': Found
    Connected to Asterisk 1.4.25 currently running on elastix (pid = 2616)
    Verbosity is at least 4
    == Starting OOH323/(null)-0f36 at internal,2525,1 failed so falling back to exten 's'
    == Starting OOH323/(null)-0f36 at internal,s,1 still failed so falling back to context 'default'
    -- Executing [s@default:1] Playback("OOH323/(null)-0f36", "vm-goodbye") in new stack
    -- <OOH323/(null)-0f36> Playing 'vm-goodbye' (language 'en')
    -- Executing [s@default:2] Macro("OOH323/(null)-0f36", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("OOH323/(null)-0f36", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("OOH323/(null)-0f36", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("OOH323/(null)-0f36", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("OOH323/(null)-0f36", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("OOH323/(null)-0f36", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("OOH323/(null)-0f36", "") in new stack
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'OOH323/(null)-0f36' in macro 'hangupcall'
    == Spawn extension (default, s, 2) exited non-zero on 'OOH323/(null)-0f36'
    elastix*CLI>



    Call from 2001 Elastic to 2270 Avaya

    [root@elastix ~]# asterisk -rvvv
    Asterisk 1.4.25, Copyright (C) 1999 - 2008 Digium, Inc. and others.
    Created by Mark Spencer <markster@digium.com>
    Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
    This is free software, with components licensed under the GNU General Public
    License version 2 and other licenses; you are welcome to redistribute it under
    certain conditions. Type 'core show license' for details.
    =========================================================================
    == Parsing '/etc/asterisk/asterisk.conf': Found
    == Parsing '/etc/asterisk/extconfig.conf': Found
    Connected to Asterisk 1.4.25 currently running on elastix (pid = 2616)
    Verbosity is at least 4
    -- Executing [2270@from-internal:1] Set("SIP/2001-09be9d10", "INTRACOMPANYROUTE=YES") in new stack
    -- Executing [2270@from-internal:2] Macro("SIP/2001-09be9d10", "user-callerid|SKIPTTL|") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/2001-09be9d10", "AMPUSER=2001") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/2001-09be9d10", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/2001-09be9d10", "1|Set|REALCALLERIDNUM=2001") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/2001-09be9d10", "AMPUSER=2001") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/2001-09be9d10", "AMPUSERCIDNAME=Daniel 2001") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/2001-09be9d10", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/2001-09be9d10", "AMPUSERCID=2001") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/2001-09be9d10", "CALLERID(all)="Daniel 2001" <2001>") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/2001-09be9d10", "REALCALLERIDNUM=2001") in new stack
    -- Executing [s@macro-user-callerid:10] ExecIf("SIP/2001-09be9d10", "0|Set|CHANNEL(language)=") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/2001-09be9d10", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,20)
    -- Executing [s@macro-user-callerid:20] NoOp("SIP/2001-09be9d10", "Using CallerID "Daniel 2001" <2001>") in new stack
    -- Executing [2270@from-internal:3] Set("SIP/2001-09be9d10", "_NODEST=") in new stack
    -- Executing [2270@from-internal:4] Macro("SIP/2001-09be9d10", "record-enable|2001|OUT|") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/2001-09be9d10", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/2001-09be9d10", "recordingcheck|20091008-065940|1254999580.48") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
    recordingcheck|20091008-065940|1254999580.48: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/2001-09be9d10", "") in new stack
    -- Executing [2270@from-internal:5] Macro("SIP/2001-09be9d10", "dialout-trunk|2|2270||") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/2001-09be9d10", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/2001-09be9d10", "0?sub-pincheck|s|1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/2001-09be9d10", "0?disabletrunk|1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/2001-09be9d10", "DIAL_NUMBER=2270") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/2001-09be9d10", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/2001-09be9d10", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/2001-09be9d10", "0?nomax") in new stack
    -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/2001-09be9d10", "0?chanfull") in new stack
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/2001-09be9d10", "1?skipoutcid") in new stack
    -- Goto (macro-dialout-trunk,s,12)
    -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/2001-09be9d10", "0|AGI|fixlocalprefix") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/2001-09be9d10", "OUTNUM=2270") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/2001-09be9d10", "custom=AMP") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/2001-09be9d10", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)tr") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/2001-09be9d10", "dialout-trunk-predial-hook|") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/2001-09be9d10", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/2001-09be9d10", "0?bypass|1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/2001-09be9d10", "1?customtrunk") in new stack
    -- Goto (macro-dialout-trunk,s,21)
    -- Executing [s@macro-dialout-trunk:21] Set("SIP/2001-09be9d10", "pre_num=AMP:OOH323/") in new stack
    -- Executing [s@macro-dialout-trunk:22] Set("SIP/2001-09be9d10", "the_num=OUTNUM") in new stack
    -- Executing [s@macro-dialout-trunk:23] Set("SIP/2001-09be9d10", "post_num=@172.16.58.24:1720") in new stack
    -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/2001-09be9d10", "1?outnum:skipoutnum") in new stack
    -- Goto (macro-dialout-trunk,s,25)
    -- Executing [s@macro-dialout-trunk:25] Set("SIP/2001-09be9d10", "the_num=2270") in new stack
    -- Executing [s@macro-dialout-trunk:26] Dial("SIP/2001-09be9d10", "OOH323/2270@172.16.58.24:1720|300|tr") in new stack
    -- Called 2270@172.16.58.24:1720
    == Spawn extension (macro-dialout-trunk, s, 26) exited non-zero on 'SIP/2001-09be9d10' in macro 'dialout-trunk'
    == Spawn extension (from-internal, 2270, 5) exited non-zero on 'SIP/2001-09be9d10'
    -- Executing [h@macro-dialout-trunk:1] Macro("SIP/2001-09be9d10", "hangupcall|") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/2001-09be9d10", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("SIP/2001-09be9d10", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/2001-09be9d10", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("SIP/2001-09be9d10", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("SIP/2001-09be9d10", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("SIP/2001-09be9d10", "") in new stack
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2001-09be9d10' in macro 'hangupcall'
    == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/2001-09be9d10'
    elastix*CLI>
     
  6. hinzinho

    Joined:
    Sep 18, 2009
    Messages:
    461
    Likes Received:
    0
    The Avaya systems we have are the IP Office. This is what I have in our H323 file:

    Code:
    [general]
    port=1720
    bindaddr=<ip address of Asterisk>
    gateway=no
    faststart=yes
    h245tunneling=yes
    h323id=ObjSysAsterisk
    e164=100
    callerid=asterisk
    gatekeeper = DISABLE
    context=default
    disallow=all     ;Note order of disallow/allow is important.
    allow=ulaw
    dtmfmode=rfc2833
    progress_setup = 8
    progress_alert = 8
    
    [AvayaPBX]
    type=friend
    port=1720
    ip=<ip of Avaya PBX>
    context=from-internal
    disallow=all
    allow=ulaw
    rtptimeout=60
    
     

Share This Page