H323 IP Trunk avec Elastix et Avaya (Codec ???)

Discussion in 'Elastix 2.x' started by daudet, Oct 8, 2009.

  1. daudet

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    Bonjour
    je viens d installer Elastix pour en faire un pont de conference sur notre PBX Avaya.

    presentement j ai un Trunk H323 2 canaux entre le Elastix et Avaya. Du Avaya Ext 2270 je compose un poste ou le trunk access code et le Elastix me repond Goodbye et deconnection.
    Du Elastix avec un Softphone je compose le 2270 ca sonne mais le rien du cote Avaya.

    J'ai un deuxieme PC avec Trixbox meme config. Meme chose
    Si je fais un trace ou status du poste 2270 sur l'Avaya il me montre la connection au Elastix/Trixbox avec un codec 711a

    J'y suis presque mais il manque un petit quelque chose dans la syncronisation et codec.

    Donc j'attend vos reponses.

    Merci

    Daniel
     
  2. danardf

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    Salut.

    Il faut peut-être regarder ce qu'il y a dans la config h323.
    Je sais qu'il y a eu des problèmes h323<->avaya, mais ont-ils été réglé?!

    Si tu peux faire une trace debug de ton appel h323 sur Elastix en mode CLI et nous attacher le résultat dans ton prochain post, ce serait cool.

    Peut-être une histoire de context:
    Si tu veux que tes extensions fassent partie intégrante de ton réseau VoIP (Avaya + Elastix) choisir peut-être le context from-internal.

    Tiens nous au courant. ;)
     
  3. danardf

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    Ha... Daniel.. j'oubliais.

    Que donne côté Elastix la commande show channeltypes?
     
  4. daudet

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    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

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    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:
    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.comThis e-mail address is being protected from spam bots, you need JavaScript enabled to view it >
    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.comThis e-mail address is being protected from spam bots, you need JavaScript enabled to view it >
    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?outnumkipoutnum" 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@172.16.58.24This e-mail address is being protected from spam bots, you need JavaScript enabled to view it :1720|300|tr" in new stack
    -- Called 2270@172.16.58.24This e-mail address is being protected from spam bots, you need JavaScript enabled to view it :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. danardf

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    [definity]
    type=friend
    context=internal <-- from-internal à moins que tu ais cloné le context en internal?
    host=172.16.58.24
    port=1720 <-- déjà mis dans [general], donc pas besoin.
    disallow=all
    allow=ulaw <-- Tu dis qu'avec tribox c'est du G711a, pourquoi G711µ ?
    canreinvite=no
    dtmfmode=internal <-- Ce n'est pas RFC2833 ou inband ou info ?

    Fais attention à la cohérence entre [general] et [definity] et le codec qu'utilise Avaya.
     

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