No puedo configurar las llamadas

Discussion in 'Elastix 2.x' started by inforemp, Sep 16, 2009.

  1. inforemp

    Joined:
    Sep 8, 2009
    Messages:
    10
    Likes Received:
    0
    Me canse de leer posts y manuales, y ya estoy muy confundido

    No puedo hacer llamadas desde mi extension configurada como "3000" hacia algun numero interno de mi voip, por ejemplo "100000", cada vez q marco me dice "todas las lineas se encuentran ocupadas"

    y si marco mi extension "3000" cuyo numero de mi voip seria el "100123", nunca recibo la llamada.


    Aca pego todo lo que tengo hasta ahora, espero que alguien pueda iluminarme en lo que estoy haciendo mal. PD: 0 experiencia en esto, hace 1 semana instale el elastix y recien entro en el "mundo voip"

    [​IMG]
    [​IMG]

    Los puertos abiertos en mi Linksys PAP2 son Linea 1: 5060 y Linea 2: 5061

    sip.conf

    externip=cuenxxxx.ath.xx
    localnet=192.168.1.0/255.255.255.0

    sip_nat.conf

    nat=yes
    externip=cuentxxx.ath.cx
    localnet=192.168.1.0/255.255.255.0
    externrefresh=5




    trunk:

    username= ? (extension user, o voip?)
    type=peer
    secret=xxxxx
    insecure=very
    host= ? (router ip, elastix, o my voip?)
    dtmfmode=rfc2833
    disallow=all
    allow=alaw&ulaw&gsm
    canredirect=no
    canreinvite=no

    user details

    canreinvite=no
    context=from-trunk
    fromuser= ? (usuario extension o voip?)
    qualify=no
    secret=xxxx
    type=user
    username=xxxx


    Register String


    usuariovoip:passwordvoip@200.69.159.33 (ip voip) no?


    outbound routes:

    dial pattern: 8|x.

    trunk sequence: trunk


    my sip debug:
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/3000-09f58aa8", "0?bypass|1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/3000-09f58aa8", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/3000-09f58aa8", "SIP/citarella/100000|300|") in new stack
    Audio is at 192.168.1.100 port 13526
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x2 (gsm) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (NAT) to 192.168.1.50:5060:
    INVITE sip:100000@192.168.1.50 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1f6352ed;rport
    From: "1003000" <sip:1003000@192.168.1.100>;tag=as50c053f6
    To: <sip:100000@192.168.1.50>
    Contact: <sip:1003000@192.168.1.100>
    Call-ID: 21483b1906c22465184755a9611a97c7@192.168.1.100
    CSeq: 102 INVITE
    User-Agent: Elastix
    Max-Forwards: 70
    Date: Wed, 16 Sep 2009 12:24:28 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 287

    v=0
    o=root 3125 3125 IN IP4 192.168.1.100
    s=session
    c=IN IP4 192.168.1.100
    t=0 0
    m=audio 13526 RTP/AVP 0 8 3 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:eek:ff - - - -
    a=ptime:20
    a=sendrecv

    ---
    -- Called citarella/100000
    cuentaip*CLI>
    <--- SIP read from 192.168.1.50:5060 --->
    SIP/2.0 404 Not Found
    To: <sip:100000@192.168.1.50>;tag=b81f97b570e4f770i0
    From: "1003000" <sip:1003000@192.168.1.100>;tag=as50c053f6
    Call-ID: 21483b1906c22465184755a9611a97c7@192.168.1.100
    CSeq: 102 INVITE
    Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1f6352ed
    Server: Linksys/PAP2T-3.1.15(LS)
    Content-Length: 0


    <------------->
    --- (8 headers 0 lines) ---
    Transmitting (NAT) to 192.168.1.50:5060:
    ACK sip:100000@192.168.1.50 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1f6352ed;rport
    From: "1003000" <sip:1003000@192.168.1.100>;tag=as50c053f6
    To: <sip:100000@192.168.1.50>;tag=b81f97b570e4f770i0
    Contact: <sip:1003000@192.168.1.100>
    Call-ID: 21483b1906c22465184755a9611a97c7@192.168.1.100
    CSeq: 102 ACK
    User-Agent: Elastix
    Max-Forwards: 70
    Content-Length: 0


    ---
    -- SIP/citarella-09f5fa88 is circuit-busy
    == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s@macro-dialout-trunk:20] Goto("SIP/3000-09f58aa8", "s-CONGESTION|1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/3000-09f58aa8", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,3)
    -- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/3000-09f58aa8", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
    -- Executing [8100000@from-internal:5] Macro("SIP/3000-09f58aa8", "outisbusy|") in new stack
    -- Executing [s@macro-outisbusy:1] Playback("SIP/3000-09f58aa8", "all-circuits-busy-now|noanswer") in new stack
    Audio is at 192.168.1.100 port 11100
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x100 (g729) to SDP
    Adding codec 0x1 (g723) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    cuentaip*CLI>
    <--- Transmitting (NAT) to 192.168.1.50:5060 --->
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61;received=192.168.1.50
    From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
    To: <sip:8100000@192.168.1.100>;tag=as6734574c
    Call-ID: c4612e0e-bb4b4c81@192.168.1.50
    CSeq: 102 INVITE
    User-Agent: Elastix
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Contact: <sip:8100000@192.168.1.100>
    Content-Type: application/sdp
    Content-Length: 355

    v=0
    o=root 3125 3125 IN IP4 192.168.1.100
    s=session
    c=IN IP4 192.168.1.100
    t=0 0
    m=audio 11100 RTP/AVP 0 8 18 4 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:4 G723/8000
    a=fmtp:4 annexa=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:eek:ff - - - -
    a=ptime:20
    a=sendrecv

    <------------>
    -- <SIP/3000-09f58aa8> Playing 'all-circuits-busy-now' (language 'es')
    Really destroying SIP dialog '21483b1906c22465184755a9611a97c7@192.168.1.100' Method: INVITE
    -- Executing [s@macro-outisbusy:2] Playback("SIP/3000-09f58aa8", "pls-try-call-later|noanswer") in new stack
    -- <SIP/3000-09f58aa8> Playing 'pls-try-call-later' (language 'es')
    cuentaip*CLI>
    <--- SIP read from 192.168.1.50:5060 --->
    CANCEL sip:8100000@192.168.1.100 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61
    From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
    To: <sip:8100000@192.168.1.100>
    Call-ID: c4612e0e-bb4b4c81@192.168.1.50
    CSeq: 102 CANCEL
    Max-Forwards: 70
    Proxy-Authorization: Digest username="3000",realm="asterisk",nonce="65ea3d5a",uri="sip:8100000@192.168.1.100",algorithm=MD5,response="4afdab4c474409369b72ce7ae707f132"
    User-Agent: Linksys/PAP2T-3.1.15(LS)
    Content-Length: 0


    <------------->
    --- (10 headers 0 lines) ---
    Sending to 192.168.1.50 : 5060 (NAT)

    <--- Reliably Transmitting (NAT) to 192.168.1.50:5060 --->
    SIP/2.0 487 Request Terminated
    Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61;received=192.168.1.50
    From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
    To: <sip:8100000@192.168.1.100>;tag=as6734574c
    Call-ID: c4612e0e-bb4b4c81@192.168.1.50
    CSeq: 102 INVITE
    User-Agent: Elastix
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Length: 0


    <------------>

    <--- Transmitting (NAT) to 192.168.1.50:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61;received=192.168.1.50
    From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
    To: <sip:8100000@192.168.1.100>;tag=as6734574c
    Call-ID: c4612e0e-bb4b4c81@192.168.1.50
    CSeq: 102 CANCEL
    User-Agent: Elastix
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Length: 0


    <------------>
    == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/3000-09f58aa8' in macro 'outisbusy'
    == Spawn extension (from-internal, 8100000, 5) exited non-zero on 'SIP/3000-09f58aa8'
    -- Executing [h@from-internal:1] Macro("SIP/3000-09f58aa8", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/3000-09f58aa8", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/3000-09f58aa8", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/3000-09f58aa8", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/3000-09f58aa8", "") in new stack
    == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/3000-09f58aa8' in macro 'hangupcall'
    == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/3000-09f58aa8'
    cuentaip*CLI>
    <--- SIP read from 192.168.1.50:5060 --->
    ACK sip:8100000@192.168.1.100 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61
    From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
    To: <sip:8100000@192.168.1.100>;tag=as6734574c
    Call-ID: c4612e0e-bb4b4c81@192.168.1.50
    CSeq: 102 ACK
    Max-Forwards: 70
    Proxy-Authorization: Digest username="3000",realm="asterisk",nonce="65ea3d5a",uri="sip:8100000@192.168.1.100",algorithm=MD5,response="fc9351980db59df69276c8325bb1ebff"
    Contact: Linea 1 <sip:3000@192.168.1.50:5060>
    User-Agent: Linksys/PAP2T-3.1.15(LS)
    Content-Length: 0


    <------------->
    --- (11 headers 0 lines) ---
    Really destroying SIP dialog 'c4612e0e-bb4b4c81@192.168.1.50' Method: ACK
     
  2. telecomtechnician

    Joined:
    Jan 8, 2008
    Messages:
    422
    Likes Received:
    0
    Hola, empecemos por el principio

    1) Las llamadas que quieres hacer son internas? (de una extensión IP a otra dentro de la misma red lan?)

    2) Si una o ambas extensiones IP son remotas (fuera de la red lan) abriste los puertos correctos en el router?, configuraste el archivo sip_nat.conf?

    2) Verificaste que cuando creaste las extensiones, tienen los datos importantes como creación de buzón de voz y que se encuentran en el mismo contexto?

    Espero tus comentarios

    David Medina
     
  3. inforemp

    Joined:
    Sep 8, 2009
    Messages:
    10
    Likes Received:
    0
    ya lo solucione, ahora tengo otra duda, creo otro post, por favor, respondeme ahi :) gracias.
     
  4. gamba47

    Joined:
    May 28, 2009
    Messages:
    595
    Likes Received:
    0
    Sería muy bueno que comentes en que te equivocastes y como lo solucionastes, más que nada para que quede el problema y la solución, para futuros usuarios que pasen por lo mismo que vos, aunque te parezca algo fácil ahora, a otros le puede pasar lo mismo!

    gracias.

    gamba47
     
  5. ElasMex

    Joined:
    Oct 22, 2007
    Messages:
    493
    Likes Received:
    0
    inforemp

    Coloca tu solución o el error que tenías.

    Hay que dar seguimiento a los Topic que uno genera.

    Saludos
     
  6. inforemp

    Joined:
    Sep 8, 2009
    Messages:
    10
    Likes Received:
    0
    Recien vuelvo a entrar a los foros, sepan disculpar, pongo mi solucion, estaban mal hecho los trunks... les pego como los deje.

    Peer Details
    username=XXXXX (Aqui va el nombre de usuario del servicio VOIP)
    type=peer
    secret=1234 (password del servicio voip)
    insecure=very
    host=200.69.159.33 (ip del servicio voip)
    dtmfmode=rfc2833
    disallow=all
    allow=alaw&ulaw&gsm
    canredirect=no
    canreinvite=no

    User Details
    canreinvite=no
    context=from-trunk
    fromuser=XXXX (nombre de usuario de una extension creada)
    qualify=no
    secret=1234 (password extension creada)
    type=user
    username=3000 (nuevamente nombre de usuario)

    Register String:

    usuario:1234@200.xxx.xxx.xxx (esto es lo mas importante, la forma de conectarse al servicio voip es "NOMBREDUSUARIOVOIP:pASSWORDVOIP@IPDELSERVIDORVOIP
     

Share This Page