ayuda con ruta entrate sip

Discussion in 'Elastix 2.x' started by axeellcevallos, Sep 13, 2009.

  1. axeellcevallos

    Joined:
    Nov 14, 2008
    Messages:
    78
    Likes Received:
    0
    amigos mios tengo el siguiente problema

    nesesito configurar numeros DID de USA en la elastix compre algunos en
    didww y configure para que estos sean redirigidos al elastix el problema es que cuando la llamada entra al elastix el elastix no reconose el peer del que le estan mandando la llamada y me da el mensage de no servicio

    estos son los dato que me dio el probedor de didww

    Outgoing Settings

    Trunk Name: didww_us

    Incoming Settings
    USER Context: from-didww1
    USER Details:
    dtmf=rfc2833
    dtmfmode=rfc2833
    host=204.11.194.34
    insecure=very
    type=peer

    context=from-trunk

    tambien cree la ruta entrante con el did y la extencion donde quiero que termine la llamada

    esta tiene un did zap y funciona bien

    los configuro pero la llamada entra a l asterisk y asterisk da el tono de no servicio dice que es un peer desconocido

    de antemano gracias por su alluda
     
  2. gamba47

    Joined:
    May 28, 2009
    Messages:
    595
    Likes Received:
    0
    Pusistes algo en el registrer string ?

    gamba47
     
  3. axeellcevallos

    Joined:
    Nov 14, 2008
    Messages:
    78
    Likes Received:
    0
    laverdad no no puse nada lo configure segun el provedor de las dids
    pero si me dices que devo de poner en el register string para que se registre lo intentare

    este es el log de la consola cuando entra la llamada

    -- Executing [13055078443@from-sip-external:1] NoOp("SIP/204.11.194.35-08b94e58", "Received incoming SIP connection from unknown peer to 13055078443") in new stack
    -- Executing [13055078443@from-sip-external:2] Set("SIP/204.11.194.35-08b94e58", "DID=13055078443") in new stack
    -- Executing [13055078443@from-sip-external:3] Goto("SIP/204.11.194.35-08b94e58", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/204.11.194.35-08b94e58", "0?from-trunk|13055078443|1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/204.11.194.35-08b94e58", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2009-09-14 21:03:58 UTC.
    -- Executing [s@from-sip-external:3] Answer("SIP/204.11.194.35-08b94e58", "") in new stack
    -- Executing [s@from-sip-external:4] Wait("SIP/204.11.194.35-08b94e58", "2") in new stack
    -- Executing [s@from-sip-external:5] Playback("SIP/204.11.194.35-08b94e58", "ss-noservice") in new stack
    -- <SIP/204.11.194.35-08b94e58> Playing 'ss-noservice' (language 'en')
    -- Executing [s@from-sip-external:6] PlayTones("SIP/204.11.194.35-08b94e58", "congestion") in new stack
    -- Executing [s@from-sip-external:7] Congestion("SIP/204.11.194.35-08b94e58", "5") in new stack
    == Spawn extension (from-sip-external, s, 7) exited non-zero on 'SIP/204.11.194.35-08b94e58'
    -- Executing [h@from-sip-external:1] NoOp("SIP/204.11.194.35-08b94e58", "Hangup") in new stack
    -- Executing [h@from-sip-external:2] Set("SIP/204.11.194.35-08b94e58", "DID=s") in new stack
    -- Executing [h@from-sip-external:3] Goto("SIP/204.11.194.35-08b94e58", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/204.11.194.35-08b94e58", "0?from-trunk|s|1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/204.11.194.35-08b94e58", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2009-09-14 21:04:11 UTC.
    -- Executing [s@from-sip-external:3] Answer("SIP/204.11.194.35-08b94e58", "") in new stack
    == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/204.11.194.35-08b94e58'
     
  4. jgutierrez

    Joined:
    Feb 28, 2008
    Messages:
    5,737
    Likes Received:
    0
    Te recomiendo que edites el archivo:
    /etc/asterisk/sip_general_custom.conf
    y le pongas:
    context=from-pstn
    Luego ejecutas desde la consola:
    asterisk -rx "reload"
     
  5. axeellcevallos

    Joined:
    Nov 14, 2008
    Messages:
    78
    Likes Received:
    0
    muchas garcias despues de colocar

    context=from-pstn

    tambien agregue en el extensions_custom.conf

    [from-didww]
    exten=> 13055078443,1,Dial(SIP/203)

    muchisimas gracias amigos por su ayuda
     

Share This Page