Elastix 1.3 no funciona con troncales IP SIP

Discussion in 'Elastix 2.x' started by teletax, Nov 14, 2008.

  1. teletax

    Joined:
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    En la version 1.3.2 de Elastix tengo un problema serio de troncales IP SIP.

    Esta configuracion funcionaba bien en la version anterior

    Trunk Name SellVOIP1
    allow=ulaw
    context=from-pstn
    disallow=all
    host=IP del suplidor
    insecure=very
    notransfer=yes
    qualify=yes
    type=peer

    User Contex from-pstn
    allow=ulaw
    context=from-pstn
    disallow=all
    host=IP del suplidor
    insecure=very
    notransfer=yes
    qualify=yes
    type=user

    Ahora en la nueva version 1.3.2 ya no funciona. Se registra bien, pero cuando intento hacer una llamada me sale el error
    que muestro. Me da todods los canales entan ocupados.

    Alguien me puede ayudar por favor.

    Gracias,

    Rodrigo Cuadra
    rcuadra@aplitel.com

    -- Executing [13054283910@from-internal:1] Macro("SIP/1002-b7b0d3b0", "user-callerid|SKIPTTL|") in new stack
    -- Executing [s@macro-user-callerid:1] NoOp("SIP/1002-b7b0d3b0", "user-callerid: device 1002") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/1002-b7b0d3b0", "AMPUSER=1002") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/1002-b7b0d3b0", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("SIP/1002-b7b0d3b0", "1|Set|REALCALLERIDNUM=1002") in new stack
    -- Executing [s@macro-user-callerid:5] NoOp("SIP/1002-b7b0d3b0", "REALCALLERIDNUM is 1002") in new stack
    -- Executing [s@macro-user-callerid:6] Set("SIP/1002-b7b0d3b0", "AMPUSER=1002") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/1002-b7b0d3b0", "AMPUSERCIDNAME=GrandStream GX2000") in new stack
    -- Executing [s@macro-user-callerid:8] GotoIf("SIP/1002-b7b0d3b0", "0?report") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/1002-b7b0d3b0", "AMPUSERCID=1002") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/1002-b7b0d3b0", "CALLERID(all)="GrandStream GX2000" <1002>") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/1002-b7b0d3b0", "REALCALLERIDNUM=1002") in new stack
    -- Executing [s@macro-user-callerid:12] ExecIf("SIP/1002-b7b0d3b0", "0|Set|CHANNEL(language)=") in new stack
    -- Executing [s@macro-user-callerid:13] NoOp("SIP/1002-b7b0d3b0", "TTL: ARG1: SKIPTTL") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("SIP/1002-b7b0d3b0", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,23)
    -- Executing [s@macro-user-callerid:23] NoOp("SIP/1002-b7b0d3b0", "Using CallerID "GrandStream GX2000" <1002>") in new stack
    -- Executing [13054283910@from-internal:2] Set("SIP/1002-b7b0d3b0", "_NODEST=") in new stack
    -- Executing [13054283910@from-internal:3] Macro("SIP/1002-b7b0d3b0", "record-enable|1002|OUT|") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/1002-b7b0d3b0", "0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/1002-b7b0d3b0", "recordingcheck|20081113-075223|1226584343.132") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
    recordingcheck|20081113-075223|1226584343.132: Outbound recording enabled.
    recordingcheck|20081113-075223|1226584343.132: CALLFILENAME=OUT-1002-1226584343.132
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:999] MixMonitor("SIP/1002-b7b0d3b0", "OUT-1002-1226584343.132.wav") in new stack
    -- Executing [13054283910@from-internal:4] Macro("SIP/1002-b7b0d3b0", "dialout-trunk|1|13054283910||") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/1002-b7b0d3b0", "DIAL_TRUNK=1") in new stack
    -- Executing [s@macro-dialout-trunk:2] ExecIf("SIP/1002-b7b0d3b0", "0|Authenticate|") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1002-b7b0d3b0", "0?disabletrunk|1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/1002-b7b0d3b0", "DIAL_NUMBER=13054283910") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/1002-b7b0d3b0", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/1002-b7b0d3b0", "GROUP()=OUT_1") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1002-b7b0d3b0", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1002-b7b0d3b0", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/1002-b7b0d3b0", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1002-b7b0d3b0", "outbound-callerid|1") in new stack
    -- Executing [s@macro-outbound-callerid:1] GotoIf("SIP/1002-b7b0d3b0", "1?start") in new stack
    -- Goto (macro-outbound-callerid,s,3)
    -- Executing [s@macro-outbound-callerid:3] NoOp("SIP/1002-b7b0d3b0", "REALCALLERIDNUM is 1002") in new stack
    -- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/1002-b7b0d3b0", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,9)
    -- Executing [s@macro-outbound-callerid:9] Set("SIP/1002-b7b0d3b0", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:10] Set("SIP/1002-b7b0d3b0", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:11] Set("SIP/1002-b7b0d3b0", "TRUNKOUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:12] GotoIf("SIP/1002-b7b0d3b0", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,16)
    -- Executing [s@macro-outbound-callerid:16] GotoIf("SIP/1002-b7b0d3b0", "1?usercid") in new stack
    -- Goto (macro-outbound-callerid,s,18)
    -- Executing [s@macro-outbound-callerid:18] GotoIf("SIP/1002-b7b0d3b0", "1?report") in new stack
    -- Goto (macro-outbound-callerid,s,22)
    -- Executing [s@macro-outbound-callerid:22] NoOp("SIP/1002-b7b0d3b0", "CallerID set to "GrandStream GX2000" <1002>") in new stack
    -- Executing [s@macro-dialout-trunk:12] AGI("SIP/1002-b7b0d3b0", "fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
    == Begin MixMonitor Recording SIP/1002-b7b0d3b0
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/1002-b7b0d3b0", "OUTNUM=13054283910") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/1002-b7b0d3b0", "custom=SIP/SellVOIP1") in new stack
    -- Executing [s@macro-dialout-trunk:15] GotoIf("SIP/1002-b7b0d3b0", "1?gocall") in new stack
    -- Goto (macro-dialout-trunk,s,17)
    -- Executing [s@macro-dialout-trunk:17] Macro("SIP/1002-b7b0d3b0", "dialout-trunk-predial-hook|") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1002-b7b0d3b0", "0?bypass|1") in new stack
    -- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/1002-b7b0d3b0", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:20] Dial("SIP/1002-b7b0d3b0", "SIP/SellVOIP1/13054283910|300|") in new stack
    -- Couldn't call SellVOIP1/13054283910
    == Everyone is busy/congested at this time (0:0/0/0)
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/1002-b7b0d3b0", "s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/1002-b7b0d3b0", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/1002-b7b0d3b0", "TRUNK Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack
    -- Executing [13054283910@from-internal:5] Macro("SIP/1002-b7b0d3b0", "outisbusy|") in new stack
    -- Executing [s@macro-outisbusy:1] Playback("SIP/1002-b7b0d3b0", "all-circuits-busy-now|noanswer") in new stack
    -- <SIP/1002-b7b0d3b0> Playing 'all-circuits-busy-now' (language 'en')
    -- Executing [s@macro-outisbusy:2] Playback("SIP/1002-b7b0d3b0", "pls-try-call-later|noanswer") in new stack
    -- <SIP/1002-b7b0d3b0> Playing 'pls-try-call-later' (language 'en')
    -- Executing [s@macro-outisbusy:3] Macro("SIP/1002-b7b0d3b0", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1002-b7b0d3b0", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("SIP/1002-b7b0d3b0", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/1002-b7b0d3b0", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("SIP/1002-b7b0d3b0", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("SIP/1002-b7b0d3b0", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("SIP/1002-b7b0d3b0", "") in new stack
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1002-b7b0d3b0' in macro 'hangupcall'
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1002-b7b0d3b0' in macro 'outisbusy'
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1002-b7b0d3b0'
    == End MixMonitor Recording SIP/1002-b7b0d3b0
     
  2. JoseVelasco

    Joined:
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    Imagino que será una tontería pero has probado a poner primer disallow=all y luego allow=ulaw ?

    Yo uso 1.3-2 y tengo una troncal SIP con Sarevoz y no tengo problemas en llamar, otra cosa ya es recibir llamadas, pero para llamar no tengo problemas.

    Ciao
     
  3. netillo123X

    Joined:
    Feb 4, 2009
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    hola que tal el problema esta en que si la troncal sip esta hecha entre dos PBX con diferentes versiones de frepbx no funciona ya me di cuenta que para que las troncales sip funciones deberas tener la mismoa version de elastix en ambos equipos para que las versiones de freepbx funcionen

    lo que sucede es que aun que los protocolos son los mismos hay versiones de elastix nuevas endonde tambien asterisk se renueva y freepbx tambien entonces freepbx genera un monton de macros nuevas para que asterisk funcione en modo grafico de aqui el problema cuando ya tienes dominado asterisk en una version de pronto sale una nueva pero resulta que esa version tiene un detalle fuera del alcance de free pbx por lo que freepbx se actualiza ala par de asterisk entonces si de un lado tienes un elastix de vieja version y de otro lado igual no hay problema el detalle viene cuando en uno de los dos lados actualizas una version ya no es nada es compatible


    suerte

    recomiendo que utilices mismas versiones
     
  4. jlara

    Joined:
    Nov 1, 2007
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    Saludos,

    ya pudiste resolver el problema que tenias?

    pues tengo el mismo problema que tu pero con broadvoice y la troncal se registra pero no puedo sacar ni recibir llamadas.

    un coridal saludo.
     

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