Linea conecta pero no está disponible

Discussion in 'Elastix 2.x' started by Alferez, Jul 9, 2009.

  1. Alferez

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    Primero presentarme y agradeceros el producto y el soporte, ya que aunte había trasteado con Asterisk y su Asterisk@home, este es múcho más completo y más amigable.

    Llevo enredando desde hace 4 años con voip y la verdad es que nunca he tenido la oportunidad de desarrollarla como ahora, ya que tengo los servicios de internet contratados con Ya.com (operadora española que da los servicios de telefonía por voip) y por fin tengo un lína con la que puedo trastear tranquilamente.

    Tras mucho probar logré loguearme con la cuenta de yacom, pero el problema es que cuando se llama desde fuera la centralita descuelga pero en vez de saltar el ivr o directamente a la línea como he configurado me dice que el nº no está disponible (o algo parecido en inglés) y si intento hacer una llamada desde dentro me dice que todas las lineas están ocupadas.

    Alguna ayuda??

    Gracias y perdon por el tostón.
     
  2. aparicio_juan

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    Podrias poner un poco mas de detalles, como que tipo de configuracion usas, si ves desde la consola, la troncal esta registra y los internos tambien, en fin que tenes puesto en la central, y las llamadas que dices entrantes que son por ip tambien. que version estas usando.

    Aparicio Juan Jose
    www.voipip.com.ar
    www.onlytechnology.com.ar
     
  3. jcastellanos

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    esta en elk mismo contexto?
     
  4. Alferez

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    Uso Asterisk 1.5.2
    Tan sólo tengo configurado una troncal SIP que registra con mi operador, ya que cuando llamas a mi número la centralita si descuelta, pero da el siguiente log:
    Code:
      -- Executing [s@from-sip-external:1] GotoIf("SIP/voipd.ya.com-084cd950", "0?from-trunk||1") in new stack
        -- Executing [s@from-sip-external:2] Set("SIP/voipd.ya.com-084cd950", "TIMEOUT(absolute)=15") in new stack
        -- Channel will hangup at 2009-07-10 17:37:07 UTC.
        -- Executing [s@from-sip-external:3] Answer("SIP/voipd.ya.com-084cd950", "") in new stack
        -- Executing [s@from-sip-external:4] Wait("SIP/voipd.ya.com-084cd950", "2") in new stack
        -- Executing [s@from-sip-external:5] Playback("SIP/voipd.ya.com-084cd950", "ss-noservice") in new stack
        -- <SIP/voipd.ya.com-084cd950> Playing 'ss-noservice' (language 'en')
        -- Executing [s@from-sip-external:6] PlayTones("SIP/voipd.ya.com-084cd950", "congestion") in new stack
        -- Executing [s@from-sip-external:7] Congestion("SIP/voipd.ya.com-084cd950", "5") in new stack
      == Spawn extension (from-sip-external, s, 7) exited non-zero on 'SIP/voipd.ya.com-084cd950'
        -- Executing [h@from-sip-external:1] NoOp("SIP/voipd.ya.com-084cd950", "Hangup") in new stack
        -- Executing [h@from-sip-external:2] Set("SIP/voipd.ya.com-084cd950", "DID=s") in new stack
        -- Executing [h@from-sip-external:3] Goto("SIP/voipd.ya.com-084cd950", "s|1") in new stack
        -- Goto (from-sip-external,s,1)
        -- Executing [s@from-sip-external:1] GotoIf("SIP/voipd.ya.com-084cd950", "0?from-trunk|s|1") in new stack
        -- Executing [s@from-sip-external:2] Set("SIP/voipd.ya.com-084cd950", "TIMEOUT(absolute)=15") in new stack
        -- Channel will hangup at 2009-07-10 17:37:19 UTC.
        -- Executing [s@from-sip-external:3] Answer("SIP/voipd.ya.com-084cd950", "") in new stack
      == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/voipd.ya.com-084cd950'
    
    Las llamadas no son por ip, sino desde lineas externas o móviles con el mismo resultado siempre.
     
  5. Alferez

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    He conseguido que las llamadas entrantes lleguen bien, salta el ivr y despues correctamente a las extensiones.
    He tenido que activar la opcion Allow Anonymous Inbound SIP Calls?.
    Pero las llamadas desde dentro hacia fuera no tengo manera.

    Gracias.
     
  6. aparicio_juan

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    Yo tambien tuve que activar el anonymouns es esa version de elastix. desde donde haces las llamadas salientes, las llamadas salientes las quieres hacer por el mismo proveedor que las entrantes, podrias pegar la configuracion que tienes, o primero el debug desde que marcas al numero asta que te tira el mensaje de que no pueden conectarse.

    Aparicio Juan Jose
    www.voipip.com.ar
    www.onlytechnology.com.ar
     
  7. Alferez

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    Las llamadas salientes tb deben salir por el mismo operador (el operador permite hasta 3 simultaneas).
    Intento hacer las llamadas a móviles o a fijos y el resultado es el mismo. Que todas las lineas están ocupadas.

    Este es el log:
    Code:
        -- Executing [0695XXXXXX@from-internal:1] Macro("SIP/601-084e68f8", "user-callerid|SKIPTTL|") in new stack
        -- Executing [s@macro-user-callerid:1] Set("SIP/601-084e68f8", "AMPUSER=601") in new stack
        -- Executing [s@macro-user-callerid:2] GotoIf("SIP/601-084e68f8", "0?report") in new stack
        -- Executing [s@macro-user-callerid:3] ExecIf("SIP/601-084e68f8", "1|Set|REALCALLERIDNUM=601") in new stack
        -- Executing [s@macro-user-callerid:4] Set("SIP/601-084e68f8", "AMPUSER=601") in new stack
        -- Executing [s@macro-user-callerid:5] Set("SIP/601-084e68f8", "AMPUSERCIDNAME=Jose") in new stack
        -- Executing [s@macro-user-callerid:6] GotoIf("SIP/601-084e68f8", "0?report") in new stack
        -- Executing [s@macro-user-callerid:7] Set("SIP/601-084e68f8", "AMPUSERCID=601") in new stack
        -- Executing [s@macro-user-callerid:8] Set("SIP/601-084e68f8", "CALLERID(all)="Jose" <601>") in new stack
        -- Executing [s@macro-user-callerid:9] Set("SIP/601-084e68f8", "REALCALLERIDNUM=601") in new stack
        -- Executing [s@macro-user-callerid:10] ExecIf("SIP/601-084e68f8", "0|Set|CHANNEL(language)=") in new stack
        -- Executing [s@macro-user-callerid:11] GotoIf("SIP/601-084e68f8", "1?continue") in new stack
        -- Goto (macro-user-callerid,s,20)
        -- Executing [s@macro-user-callerid:20] NoOp("SIP/601-084e68f8", "Using CallerID "Jose" <601>") in new stack
        -- Executing [0695XXXXXX@from-internal:2] Set("SIP/601-084e68f8", "_NODEST=") in new stack
        -- Executing [0695XXXXXX@from-internal:3] Macro("SIP/601-084e68f8", "record-enable|601|OUT|") in new stack
        -- Executing [s@macro-record-enable:1] GotoIf("SIP/601-084e68f8", "1?check") in new stack
        -- Goto (macro-record-enable,s,4)
        -- Executing [s@macro-record-enable:4] AGI("SIP/601-084e68f8", "recordingcheck|20090710-203410|1247250849.21") in new stack
        -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
      recordingcheck|20090710-203410|1247250849.21: Outbound recording not enabled
        -- AGI Script recordingcheck completed, returning 0
        -- Executing [s@macro-record-enable:5] MacroExit("SIP/601-084e68f8", "") in new stack
        -- Executing [0695XXXXXX@from-internal:4] Macro("SIP/601-084e68f8", "dialout-trunk|2|695XXXXXX||") in new stack
        -- Executing [s@macro-dialout-trunk:1] Set("SIP/601-084e68f8", "DIAL_TRUNK=2") in new stack
        -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/601-084e68f8", "0?sub-pincheck|s|1") in new stack
        -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/601-084e68f8", "0?disabletrunk|1") in new stack
        -- Executing [s@macro-dialout-trunk:4] Set("SIP/601-084e68f8", "DIAL_NUMBER=695XXXXXX") in new stack
        -- Executing [s@macro-dialout-trunk:5] Set("SIP/601-084e68f8", "DIAL_TRUNK_OPTIONS=tr") in new stack
        -- Executing [s@macro-dialout-trunk:6] Set("SIP/601-084e68f8", "OUTBOUND_GROUP=OUT_2") in new stack
        -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/601-084e68f8", "0?nomax") in new stack
        -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/601-084e68f8", "0?chanfull") in new stack
        -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/601-084e68f8", "0?skipoutcid") in new stack
        -- Executing [s@macro-dialout-trunk:10] Set("SIP/601-084e68f8", "DIAL_TRUNK_OPTIONS=") in new stack
        -- Executing [s@macro-dialout-trunk:11] Macro("SIP/601-084e68f8", "outbound-callerid|2") in new stack
        -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/601-084e68f8", "0|SetCallerPres|") in new stack
        -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/601-084e68f8", "0|Set|REALCALLERIDNUM=601") in new stack
        -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/601-084e68f8", "1?normcid") in new stack
        -- Goto (macro-outbound-callerid,s,6)
        -- Executing [s@macro-outbound-callerid:6] Set("SIP/601-084e68f8", "USEROUTCID=601") in new stack
        -- Executing [s@macro-outbound-callerid:7] Set("SIP/601-084e68f8", "EMERGENCYCID=") in new stack
        -- Executing [s@macro-outbound-callerid:8] Set("SIP/601-084e68f8", "TRUNKOUTCID=955XXXXXX") in new stack
        -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/601-084e68f8", "1?trunkcid") in new stack
        -- Goto (macro-outbound-callerid,s,12)
        -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/601-084e68f8", "1|Set|CALLERID(all)=955XXXXXX") in new stack
        -- Executing [s@macro-outbound-callerid:13] GotoIf("SIP/601-084e68f8", "0?exit") in new stack
        -- Executing [s@macro-outbound-callerid:14] Set("SIP/601-084e68f8", "CALLERID(all)=601") in new stack
        -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/601-084e68f8", "0|SetCallerPres|prohib_passed_screen") in new stack
        -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/601-084e68f8", "0|AGI|fixlocalprefix") in new stack
        -- Executing [s@macro-dialout-trunk:13] Set("SIP/601-084e68f8", "OUTNUM=695XXXXXX") in new stack
        -- Executing [s@macro-dialout-trunk:14] Set("SIP/601-084e68f8", "custom=SIP/Yacom") in new stack
        -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/601-084e68f8", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
        -- Executing [s@macro-dialout-trunk:16] Macro("SIP/601-084e68f8", "dialout-trunk-predial-hook|") in new stack
        -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/601-084e68f8", "") in new stack
        -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/601-084e68f8", "0?bypass|1") in new stack
        -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/601-084e68f8", "0?customtrunk") in new stack
        -- Executing [s@macro-dialout-trunk:19] Dial("SIP/601-084e68f8", "SIP/Yacom/695XXXXXX|300|") in new stack
      == Everyone is busy/congested at this time (1:0/0/1)
        -- Executing [s@macro-dialout-trunk:20] Goto("SIP/601-084e68f8", "s-CHANUNAVAIL|1") in new stack
        -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
        -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/601-084e68f8", "1?noreport") in new stack
        -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
        -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/601-084e68f8", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 20) - failing through to other trunks") in new stack
        -- Executing [0695XXXXXX@from-internal:5] Macro("SIP/601-084e68f8", "outisbusy|") in new stack
        -- Executing [s@macro-outisbusy:1] Playback("SIP/601-084e68f8", "all-circuits-busy-now|noanswer") in new stack
        -- <SIP/601-084e68f8> Playing 'all-circuits-busy-now' (language 'es')
        -- Executing [s@macro-outisbusy:2] Playback("SIP/601-084e68f8", "pls-try-call-later|noanswer") in new stack
        -- <SIP/601-084e68f8> Playing 'pls-try-call-later' (language 'es')
      == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/601-084e68f8' in macro 'outisbusy'
      == Spawn extension (from-internal, 0695XXXXXX, 5) exited non-zero on 'SIP/601-084e68f8'
        -- Executing [h@from-internal:1] Macro("SIP/601-084e68f8", "hangupcall") in new stack
        -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/601-084e68f8", "w") in new stack
        -- Executing [s@macro-hangupcall:2] NoCDR("SIP/601-084e68f8", "") in new stack
        -- Executing [s@macro-hangupcall:3] GotoIf("SIP/601-084e68f8", "1?skiprg") in new stack
        -- Goto (macro-hangupcall,s,6)
        -- Executing [s@macro-hangupcall:6] GotoIf("SIP/601-084e68f8", "1?skipblkvm") in new stack
        -- Goto (macro-hangupcall,s,9)
       -- Executing [s@macro-hangupcall:9] GotoIf("SIP/601-084e68f8", "1?theend") in new stack
        -- Goto (macro-hangupcall,s,11)
        -- Executing [s@macro-hangupcall:11] Hangup("SIP/601-084e68f8", "") in new stack
      == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/601-084e68f8' in macro 'hangupcall'
      == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/601-084e68f8'
    
     
  8. aparicio_juan

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    Mira la llamada trata de salir, te fijaste que este registrada la troncal, cambiaste algun contexto, podrias poner la configuracion de la troncal.

    Aparicio Juan Jose
    www.voipip.com.ar
    www.onlytechnology.com.ar
     
  9. Alferez

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    Según he leido en otros foros, este proveedor es muy especial con el Caller ID, así q voy a ver si tengo suerte y hablo con algún técnico competente y me puede decir el formato.
     
  10. tolengo

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    Si el tema va por el ID te recomentaria que en la configuracion del trunk pongas:

    fromuser=tunumeroid


    te lo mando todo:

    fromuser=tunumeroid
    username=tunumeroid
    canreinvite=yes
    context=from-trunk
    disallow=all
    allow=alaw <--es importante saber que codec permite tu provider
    dtmfmode=info <-- tipo de signalizacion que usa tu provider para tonos
    fromdomain=ipodnsproovedor
    host=ipodnsproovedor
    insecure=port,invite
    nat=yes
    secret=xxxx
    type=friend


    registracion:
    tunumeroid:password@ipodnsprovider/tunumeroid <-- es importante el formato


    si con esto no te funciona vas a tener que mandarnos mas info, lo mejor que hagas un "sip set debug" luego tratar de hacer una llamada y ahi vas a ver que es lo que esta fallando, es muy probable de que no tengas el codec adecuado.


    tambien cambia el logger.conf por un ratito asi te tira mas info en la consola.

    Saludos.

    J.
     
  11. tolengo

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    me olvidava.. por si las moscas pone en tu configuracion de trunk "never override callerid".
     
  12. Alferez

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    Nada, no lo consigo. Me he dado de alta en 12Voip para probar una de las gratuitas y la he puesto como seguinda opción y por esta si me sale.
    Se ve que como salida de de Yacom no me la da por válida y salta directamente hacia esta, por lo que por lo menos se que no es problema de configuración en la ruta de salida, sino de la troncal.

    Seguiré trasteando.
     
  13. tolengo

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    Asi no te vamos a poder ayudar, tendrias que pegar algun debug y configuracion o algo para poder guiarte.

    saludos.
     
  14. Alferez

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    Esta es la configuración que tengo puesta en la troncal:
    Code:
    Outbound Caller ID: +34XXXXXXXXX
    Never Override CallerID: Activado
    
    En Trunk Peer Details:
    type=peer
    secret=PASSWORD
    username=95XXXXXXX
    fromuser=+3495XXXXXXX
    fromdomain=voipd.ya.com
    realm=voipd.ya.com
    outboundproxy=proxy.voip.ya.com
    canrenvite=no
    insecure=very
    qualify=yes
    nat=yes
    dtmfmode=rfc2833
    
    En User Context (Vacio)
    
    En Register String:
    +3495XXXXXXX@voipd.ya.com:PASSWORD:95XXXXXXX@proxy.voip.ya.com
    
    Muchas gracias por la ayuda. Si necesitais algo más indicarme con que comando os lo puedo sacar más detallado.
     
  15. tolengo

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    para anticiparte ya te dije que el register string va con este formato:

    numero:password@ip_del_provider/numero

    sin esto se te va a compilicar


    luego es necesario que nos envies mas info. algun extrato de algun debug o algo
     
  16. Alferez

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    Tras probar como indicas el resultado es el mismo.

    He sacado un debug y el resultado es igual al que ya puse. De todas formas lo vuelvo a poner.

    Code:
    [root@elastix asterisk]# asterisk -rvvv
    Asterisk 1.4.25.1, 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 1.4.25.1 currently running on elastix (pid = 2523)
    Verbosity is at least 3
        -- Executing [0695XXXXXX@from-internal:1] Macro("SIP/602-089ccfa0", "user-callerid|SKIPTTL|") in new stack
        -- Executing [s@macro-user-callerid:1] Set("SIP/602-089ccfa0", "AMPUSER=602") in new stack
        -- Executing [s@macro-user-callerid:2] GotoIf("SIP/602-089ccfa0", "0?report") in new stack
        -- Executing [s@macro-user-callerid:3] ExecIf("SIP/602-089ccfa0", "1|Set|REALCALLERIDNUM=602") in new stack
        -- Executing [s@macro-user-callerid:4] Set("SIP/602-089ccfa0", "AMPUSER=602") in new stack
        -- Executing [s@macro-user-callerid:5] Set("SIP/602-089ccfa0", "AMPUSERCIDNAME=Merchy") in new stack
        -- Executing [s@macro-user-callerid:6] GotoIf("SIP/602-089ccfa0", "0?report") in new stack
        -- Executing [s@macro-user-callerid:7] Set("SIP/602-089ccfa0", "AMPUSERCID=602") in new stack
        -- Executing [s@macro-user-callerid:8] Set("SIP/602-089ccfa0", "CALLERID(all)="Merchy" <602>") in new stack
        -- Executing [s@macro-user-callerid:9] Set("SIP/602-089ccfa0", "REALCALLERIDNUM=602") in new stack
        -- Executing [s@macro-user-callerid:10] ExecIf("SIP/602-089ccfa0", "0|Set|CHANNEL(language)=") in new stack
        -- Executing [s@macro-user-callerid:11] GotoIf("SIP/602-089ccfa0", "1?continue") in new stack
        -- Goto (macro-user-callerid,s,20)
        -- Executing [s@macro-user-callerid:20] NoOp("SIP/602-089ccfa0", "Using CallerID "Merchy" <602>") in new stack
        -- Executing [0695XXXXXX@from-internal:2] Set("SIP/602-089ccfa0", "_NODEST=") in new stack
        -- Executing [0695XXXXXX@from-internal:3] Macro("SIP/602-089ccfa0", "record-enable|602|OUT|") in new stack
        -- Executing [s@macro-record-enable:1] GotoIf("SIP/602-089ccfa0", "1?check") in new stack
        -- Goto (macro-record-enable,s,4)
        -- Executing [s@macro-record-enable:4] AGI("SIP/602-089ccfa0", "recordingcheck|20090718-125903|1247914743.21") in new stack
        -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
      recordingcheck|20090718-125903|1247914743.21: Outbound recording not enabled
        -- AGI Script recordingcheck completed, returning 0
        -- Executing [s@macro-record-enable:5] MacroExit("SIP/602-089ccfa0", "") in new stack
        -- Executing [0695XXXXXX@from-internal:4] Macro("SIP/602-089ccfa0", "dialout-trunk|2|695XXXXXX||") in new stack
        -- Executing [s@macro-dialout-trunk:1] Set("SIP/602-089ccfa0", "DIAL_TRUNK=2") in new stack
        -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/602-089ccfa0", "0?sub-pincheck|s|1") in new stack
        -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/602-089ccfa0", "0?disabletrunk|1") in new stack
        -- Executing [s@macro-dialout-trunk:4] Set("SIP/602-089ccfa0", "DIAL_NUMBER=695XXXXXX") in new stack
        -- Executing [s@macro-dialout-trunk:5] Set("SIP/602-089ccfa0", "DIAL_TRUNK_OPTIONS=tr") in new stack
        -- Executing [s@macro-dialout-trunk:6] Set("SIP/602-089ccfa0", "OUTBOUND_GROUP=OUT_2") in new stack
        -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/602-089ccfa0", "0?nomax") in new stack
        -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/602-089ccfa0", "0?chanfull") in new stack
        -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/602-089ccfa0", "0?skipoutcid") in new stack
        -- Executing [s@macro-dialout-trunk:10] Set("SIP/602-089ccfa0", "DIAL_TRUNK_OPTIONS=") in new stack
        -- Executing [s@macro-dialout-trunk:11] Macro("SIP/602-089ccfa0", "outbound-callerid|2") in new stack
        -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/602-089ccfa0", "0|SetCallerPres|") in new stack
        -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/602-089ccfa0", "0|Set|REALCALLERIDNUM=602") in new stack
        -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/602-089ccfa0", "1?normcid") in new stack
        -- Goto (macro-outbound-callerid,s,6)
        -- Executing [s@macro-outbound-callerid:6] Set("SIP/602-089ccfa0", "USEROUTCID=602") in new stack
        -- Executing [s@macro-outbound-callerid:7] Set("SIP/602-089ccfa0", "EMERGENCYCID=") in new stack
        -- Executing [s@macro-outbound-callerid:8] Set("SIP/602-089ccfa0", "TRUNKOUTCID=+3495XXXXXXX") in new stack
        -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/602-089ccfa0", "1?trunkcid") in new stack
        -- Goto (macro-outbound-callerid,s,12)
        -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/602-089ccfa0", "1|Set|CALLERID(all)=+3495XXXXXXX") in new stack
        -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/602-089ccfa0", "1|Set|CALLERID(all)=602") in new stack
        -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/602-089ccfa0", "0|SetCallerPres|prohib_passed_screen") in new stack
        -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/602-089ccfa0", "0|AGI|fixlocalprefix") in new stack
        -- Executing [s@macro-dialout-trunk:13] Set("SIP/602-089ccfa0", "OUTNUM=695XXXXXX") in new stack
        -- Executing [s@macro-dialout-trunk:14] Set("SIP/602-089ccfa0", "custom=SIP/Yacom") in new stack
        -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/602-089ccfa0", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
        -- Executing [s@macro-dialout-trunk:16] Macro("SIP/602-089ccfa0", "dialout-trunk-predial-hook|") in new stack
        -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/602-089ccfa0", "") in new stack
        -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/602-089ccfa0", "0?bypass|1") in new stack
        -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/602-089ccfa0", "0?customtrunk") in new stack
        -- Executing [s@macro-dialout-trunk:19] Dial("SIP/602-089ccfa0", "SIP/Yacom/695XXXXXX|300|") in new stack
      == Everyone is busy/congested at this time (1:0/0/1)
        -- Executing [s@macro-dialout-trunk:20] Goto("SIP/602-089ccfa0", "s-CHANUNAVAIL|1") in new stack
        -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
        -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/602-089ccfa0", "1?noreport") in new stack
        -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
        -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/602-089ccfa0", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 20) - failing through to other trunks") in new stack
        -- Executing [0695XXXXXX@from-internal:5] Macro("SIP/602-089ccfa0", "dialout-trunk|3|695XXXXXX||") in new stack
        -- Executing [s@macro-dialout-trunk:1] Set("SIP/602-089ccfa0", "DIAL_TRUNK=3") in new stack
        -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/602-089ccfa0", "0?sub-pincheck|s|1") in new stack
        -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/602-089ccfa0", "0?disabletrunk|1") in new stack
        -- Executing [s@macro-dialout-trunk:4] Set("SIP/602-089ccfa0", "DIAL_NUMBER=695XXXXXX") in new stack
        -- Executing [s@macro-dialout-trunk:5] Set("SIP/602-089ccfa0", "DIAL_TRUNK_OPTIONS=tr") in new stack
        -- Executing [s@macro-dialout-trunk:6] Set("SIP/602-089ccfa0", "OUTBOUND_GROUP=OUT_3") in new stack
        -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/602-089ccfa0", "0?nomax") in new stack
        -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/602-089ccfa0", "0?chanfull") in new stack
        -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/602-089ccfa0", "0?skipoutcid") in new stack
        -- Executing [s@macro-dialout-trunk:10] Set("SIP/602-089ccfa0", "DIAL_TRUNK_OPTIONS=") in new stack
        -- Executing [s@macro-dialout-trunk:11] Macro("SIP/602-089ccfa0", "outbound-callerid|3") in new stack
        -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/602-089ccfa0", "0|SetCallerPres|") in new stack
        -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/602-089ccfa0", "0|Set|REALCALLERIDNUM=602") in new stack
        -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/602-089ccfa0", "1?normcid") in new stack
        -- Goto (macro-outbound-callerid,s,6)
        -- Executing [s@macro-outbound-callerid:6] Set("SIP/602-089ccfa0", "USEROUTCID=602") in new stack
        -- Executing [s@macro-outbound-callerid:7] Set("SIP/602-089ccfa0", "EMERGENCYCID=") in new stack
        -- Executing [s@macro-outbound-callerid:8] Set("SIP/602-089ccfa0", "TRUNKOUTCID=+3495XXXXXXX") in new stack
        -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/602-089ccfa0", "1?trunkcid") in new stack
        -- Goto (macro-outbound-callerid,s,12)
        -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/602-089ccfa0", "1|Set|CALLERID(all)=+3495XXXXXXX") in new stack
        -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/602-089ccfa0", "1|Set|CALLERID(all)=602") in new stack
        -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/602-089ccfa0", "0|SetCallerPres|prohib_passed_screen") in new stack
        -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/602-089ccfa0", "0|AGI|fixlocalprefix") in new stack
        -- Executing [s@macro-dialout-trunk:13] Set("SIP/602-089ccfa0", "OUTNUM=695XXXXXX") in new stack
        -- Executing [s@macro-dialout-trunk:14] Set("SIP/602-089ccfa0", "custom=SIP/12Voip") in new stack
        -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/602-089ccfa0", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
        -- Executing [s@macro-dialout-trunk:16] Macro("SIP/602-089ccfa0", "dialout-trunk-predial-hook|") in new stack
        -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/602-089ccfa0", "") in new stack
        -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/602-089ccfa0", "0?bypass|1") in new stack
        -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/602-089ccfa0", "0?customtrunk") in new stack
        -- Executing [s@macro-dialout-trunk:19] Dial("SIP/602-089ccfa0", "SIP/12Voip/695XXXXXX|300|") in new stack
        -- Called 12Voip/695XXXXXX
        -- SIP/12Voip-08a25d00 is making progress passing it to SIP/602-089ccfa0
      == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/602-089ccfa0' in macro 'dialout-trunk'
      == Spawn extension (from-internal, 0695XXXXXX, 5) exited non-zero on 'SIP/602-089ccfa0'
        -- Executing [h@macro-dialout-trunk:1] Macro("SIP/602-089ccfa0", "hangupcall|") in new stack
        -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/602-089ccfa0", "vw") in new stack
        -- Executing [s@macro-hangupcall:2] NoCDR("SIP/602-089ccfa0", "") in new stack
        -- Executing [s@macro-hangupcall:3] GotoIf("SIP/602-089ccfa0", "1?skiprg") in new stack
        -- Goto (macro-hangupcall,s,6)
        -- Executing [s@macro-hangupcall:6] GotoIf("SIP/602-089ccfa0", "1?skipblkvm") in new stack
        -- Goto (macro-hangupcall,s,9)
        -- Executing [s@macro-hangupcall:9] GotoIf("SIP/602-089ccfa0", "1?theend") in new stack
        -- Goto (macro-hangupcall,s,11)
        -- Executing [s@macro-hangupcall:11] Hangup("SIP/602-089ccfa0", "") in new stack
      == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/602-089ccfa0' in macro 'hangupcall'
      == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/602-089ccfa0'
        -- Remote UNIX connection
        -- Remote UNIX connection disconnected
    elastix*CLI>
    
    
    Si deseas algún tipo distinto dime como obtenerlo.

    Lo que veo es que la línea de Yacom (es como la he llamado) da el siguiente error al intentar usarla:

    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/602-089ccfa0", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 20) - failing through to other trunks") in new stack

    Por lo que tras ese intento fallido lo intenta por la línea de 12Voip y lo realiza con exito.
     
  17. tolengo

    Joined:
    Oct 31, 2008
    Messages:
    117
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    OK, para saber porque motivo tu provider rechaza esta llamada tenes que hacer un :

    sip set debug

    y despues manda el log. Ahi vas las headers con el motivo de rechazo.

    cualquier cosa avisa.

    suerte!

    J.
     
  18. jcastellanos

    Joined:
    Feb 10, 2009
    Messages:
    2,404
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    0
    no, si tu carier no deja pasar audio es por puertosm estas tras de un proxy?????????
     
  19. Alferez

    Joined:
    Jul 8, 2009
    Messages:
    83
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    0
    No, la centralita está conectada directamente al router, y las llamadas entrantes si deja pasar audio perfectamente.
     
  20. aparicio_juan

    Joined:
    Mar 6, 2008
    Messages:
    109
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    0
    prueba poniendo esta en la troncal:

    En Trunk Peer Details:
    disallow=all
    allow=g729&gsm&ulaw&alaw
    type=peer
    secret=PASSWORD
    username=95XXXXXXX
    fromuser=95XXXXXXX
    fromdomain=voipd.ya.com
    host=voipd.ya.com
    canrenvite=no
    insecure=very
    qualify=yes
    port=5060
    nat=yes
    dtmfmode=rfc2833

    En User Context (Vacio)

    En Register String:
    95XXXXXXX:pASSWORD@voipd.ya.com

    si no funciona prueba sacando:
    fromuser=95XXXXXXX
    fromdomain=voipd.ya.com

    chequea antes de probar que la troncal se este registrando, con el comando sip show registry.
    y luego de cada cambio para estar seguros que los tomo desde la consola reinicia asterisk poniendo
    /etc/init.d/asterisk restart asi estas seguro que los cambios fueron tomados.
    por como se ve estas marcando asi 695XXXXXX supongo que las X las pusiste tu, pero la pregunta es cual es el dial plan de tu proveedor, ya que la mayoria te pude que pongas un digito antes y que la llamada salga como si fuera internacional, eso tambien puede estar rebotando la llamada, como tambien el tipo de codec que usa tu proveedor cual es, arriba te declare 4 pero el 729 se instala manual en la maquina y es el que usa casi todos los proveedores de ip. prueba con esa configuracion y sino anda averigua estos datos, saludos.

    Aparicio Juan Jose
    www.voipip.com.ar
    www.onlytechnology.com.ar
     

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