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Llamadas salientes se cortan

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  • #1

    Llamadas salientes se cortan

    Saludos cordiales:

    Después de muchos problemas con el firewall he conseguido conectarme a mi proveedor IP. Ya estoy registrado...el problema ahora es que ni siquiera llega a dar la llamada. Es decir intenta llamar y se corta. Paso un ejemplo con un teléfono al azar:

    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    -- Executing [0034954597000@from-internal:1] Macro("SIP/20-00000014", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/20-00000014", "AMPUSER=20") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/20-00000014", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/20-00000014", "1?Set(REALCALLERIDNUM=20)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/20-00000014", "AMPUSER=20") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/20-00000014", "AMPUSERCIDNAME=Rafa Ramos") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/20-00000014", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/20-00000014", "AMPUSERCID=20") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/20-00000014", "CALLERID(all)="Rafa Ramos" <20>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/20-00000014", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/20-00000014", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] Set("SIP/20-00000014", "CALLERID(number)=20") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/20-00000014", "CALLERID(name)=Rafa Ramos") in new stack
    -- Executing [s@macro-user-callerid:21] NoOp("SIP/20-00000014", "Using CallerID "Rafa Ramos" <20>") in new stack
    -- Executing [0034954597000@from-internal:2] NoOp("SIP/20-00000014", "Calling Out Route: Fuera") in new stack
    -- Executing [0034954597000@from-internal:3] Set("SIP/20-00000014", "MOHCLASS=default") in new stack
    -- Executing [0034954597000@from-internal:4] Set("SIP/20-00000014", "_NODEST=") in new stack
    -- Executing [0034954597000@from-internal:5] Macro("SIP/20-00000014", "record-enable,20,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/20-00000014", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/20-00000014", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/20-00000014", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/20-00000014", "0?IN") in new stack
    -- Executing [s@macro-record-enable:16] ExecIf("SIP/20-00000014", "1?MacroExit()") in new stack
    -- Executing [0034954597000@from-internal:6] Macro("SIP/20-00000014", "dialout-trunk,3,4954597000,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/20-00000014", "DIAL_TRUNK=3") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/20-00000014", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/20-00000014", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/20-00000014", "DIAL_NUMBER=4954597000") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/20-00000014", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/20-00000014", "OUTBOUND_GROUP=OUT_3") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/20-00000014", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/20-00000014", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/20-00000014", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/20-00000014", "outbound-callerid,3") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/20-00000014", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/20-00000014", "0?Set(REALCALLERIDNUM=20)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/20-00000014", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/20-00000014", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/20-00000014", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/20-00000014", "TRUNKOUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/20-00000014", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/20-00000014", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/20-00000014", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/20-00000014", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/20-00000014", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/20-00000014", "0?sub-flp-3,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/20-00000014", "OUTNUM=4954597000") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/20-00000014", "custom=SIP/FONWORLD") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/20-00000014", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)) ") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/20-00000014", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/20-00000014", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/20-00000014", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/20-00000014", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/20-00000014", "SIP/FONWORLD/4954597000,300,") in new stack
    == Using SIP RTP TOS bits 184
    == Using SIP RTP CoS mark 5
    -- Called SIP/FONWORLD/4954597000
    -- SIP/FONWORLD-00000015 answered SIP/20-00000014
    -- Locally bridging SIP/20-00000014 and SIP/FONWORLD-00000015
    -- Executing [h@macro-dialout-trunk:1] Macro("SIP/20-00000014", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/20-00000014", "1?endmixmoncheck") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/20-00000014", "End of MIXMON check") in new stack
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/20-00000014", "1?nomeetmemon") in new stack
    -- Goto (macro-hangupcall,s,15)
    -- Executing [s@macro-hangupcall:15] NoOp("SIP/20-00000014", "MEETME_RECORDINGFILE=") in new stack
    -- Executing [s@macro-hangupcall:16] GotoIf("SIP/20-00000014", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,18)
    -- Executing [s@macro-hangupcall:18] NoOp("SIP/20-00000014", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:19] GotoIf("SIP/20-00000014", "1?noautomon2") in new stack
    -- Goto (macro-hangupcall,s,25)
    -- Executing [s@macro-hangupcall:25] NoOp("SIP/20-00000014", "MONITOR_FILENAME=") in new stack
    -- Executing [s@macro-hangupcall:26] GotoIf("SIP/20-00000014", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,29)
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/20-00000014", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,32)
    -- Executing [s@macro-hangupcall:32] GotoIf("SIP/20-00000014", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,34)
    -- Executing [s@macro-hangupcall:34] Hangup("SIP/20-00000014", "") in new stack
    == Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/20-00000014' in macro 'hangupcall'
    == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/20-00000014'
    == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/20-00000014' in macro 'dialout-trunk'
    == Spawn extension (from-internal, 0034954597000, 6) exited non-zero on 'SIP/20-00000014'
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected

    A ver si alguien me puede echar una mano.

    Muchísimas gracias.
    Etiquetas: Ninguno
  • #2
    Re: Llamadas salientes se cortan

    Puede ser problema del codec en uso o incluso un problema de retransmision. Para saber mejor pega el sip debug de una llamada.

    Comentario

    • #3
      Re: Llamadas salientes se cortan

      Ostras, ahí me ha pillado. No sé cómo hacer eso. Lo estoy intentando subiendo el nivel de verbosidad, pero me sale lo mismo que arriba....

      Comentario

      • #4
        Re: Llamadas salientes se cortan

        Creo que he conseguido el sip.debug. He cambiado los datos de la IP por motivos de seguridad.

        == Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/20-0000000f' in macro 'hangupcall'
        == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/20-0000000f'
        == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/20-0000000f' in macro 'dialout-trunk'
        == Spawn extension (from-internal, 0034657222797, 6) exited non-zero on 'SIP/20-0000000f'
        Really destroying SIP dialog '363830d922e3eb57101a85df61c5d275@192.168.1.123:50 60' Method: OPTIONS
        Reliably Transmitting (NAT) to 98.16.244.133:5060:
        OPTIONS sip:25@192.168.1.5:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK5fa99ab5;rport
        Max-Forwards: 70
        From: "Unknown" <sip:Unknown@192.168.1.123>;tag=as5fe63eab
        To: <sip:25@192.168.1.5:5060>
        Contact: <sip:Unknown@192.168.1.123:5060>
        Call-ID: 4cc91af406e9d161032b6c0045c9ba14@192.168.1.123:506 0
        CSeq: 102 OPTIONS
        User-Agent: FPBX-2.8.1(1.8.7.0)
        Date: Sat, 14 Jul 2012 10:36:06 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Content-Length: 0


        ---

        <--- SIP read from UDP:98.16.244.133:5060 --->
        OPTIONS sip:25@192.168.1.5:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK5fa99ab5;rport
        Max-Forwards: 70
        From: "Unknown" <sip:Unknown@192.168.1.123>;tag=as5fe63eab
        To: <sip:25@192.168.1.5:5060>
        Contact: <sip:Unknown@192.168.1.123:5060>
        Call-ID: 4cc91af406e9d161032b6c0045c9ba14@192.168.1.123:506 0
        CSeq: 102 OPTIONS
        User-Agent: FPBX-2.8.1(1.8.7.0)
        Date: Sat, 14 Jul 2012 10:36:06 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Content-Length: 0

        <------------->
        --- (13 headers 0 lines) ---
        Looking for 25 in from-sip-external (domain 192.168.1.5:5060)

        <--- Transmitting (NAT) to 98.16.244.133:5060 --->
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK5fa99ab5;received =98.16.244.133;rport=5060
        From: "Unknown" <sip:Unknown@192.168.1.123>;tag=as5fe63eab
        To: <sip:25@192.168.1.5:5060>;tag=as2b03f6bd
        Call-ID: 4cc91af406e9d161032b6c0045c9ba14@192.168.1.123:506 0
        CSeq: 102 OPTIONS
        Server: FPBX-2.8.1(1.8.7.0)
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Contact: <sip:192.168.1.123:5060>
        Accept: application/sdp
        Content-Length: 0


        <------------>
        Scheduling destruction of SIP dialog '4cc91af406e9d161032b6c0045c9ba14@192.168.1.123:50 60' in 32000 ms (Method: OPTIONS)

        <--- SIP read from UDP:98.16.244.133:5060 --->
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK5fa99ab5;received =98.16.244.133;rport=5060
        From: "Unknown" <sip:Unknown@192.168.1.123>;tag=as5fe63eab
        To: <sip:25@192.168.1.5:5060>;tag=as2b03f6bd
        Call-ID: 4cc91af406e9d161032b6c0045c9ba14@192.168.1.123:506 0
        CSeq: 102 OPTIONS
        Server: FPBX-2.8.1(1.8.7.0)
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Contact: <sip:192.168.1.123:5060>
        Accept: application/sdp
        Content-Length: 0

        <------------->
        --- (12 headers 0 lines) ---
        Really destroying SIP dialog '4cc91af406e9d161032b6c0045c9ba14@192.168.1.123:50 60' Method: OPTIONS
        Really destroying SIP dialog '4cc91af406e9d161032b6c0045c9ba14@192.168.1.123:50 60' Method: OPTIONS
        Reliably Transmitting (NAT) to 98.16.244.133:5060:
        OPTIONS sip:25@192.168.1.5:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK6e509c17;rport
        Max-Forwards: 70
        From: "Unknown" <sip:Unknown@192.168.1.123>;tag=as127aeb1b
        To: <sip:25@192.168.1.5:5060>
        Contact: <sip:Unknown@192.168.1.123:5060>
        Call-ID: 7297b2ef5a77dc6a0d7edfba56926e54@192.168.1.123:506 0
        CSeq: 102 OPTIONS
        User-Agent: FPBX-2.8.1(1.8.7.0)
        Date: Sat, 14 Jul 2012 10:37:06 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Content-Length: 0


        ---

        <--- SIP read from UDP:98.16.244.133:5060 --->
        OPTIONS sip:25@192.168.1.5:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK6e509c17;rport
        Max-Forwards: 70
        From: "Unknown" <sip:Unknown@192.168.1.123>;tag=as127aeb1b
        To: <sip:25@192.168.1.5:5060>
        Contact: <sip:Unknown@192.168.1.123:5060>
        Call-ID: 7297b2ef5a77dc6a0d7edfba56926e54@192.168.1.123:506 0
        CSeq: 102 OPTIONS
        User-Agent: FPBX-2.8.1(1.8.7.0)
        Date: Sat, 14 Jul 2012 10:37:06 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Content-Length: 0

        <------------->
        --- (13 headers 0 lines) ---
        Looking for 25 in from-sip-external (domain 192.168.1.5:5060)

        <--- Transmitting (NAT) to 98.16.244.133:5060 --->
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK6e509c17;received =98.16.244.133;rport=5060
        From: "Unknown" <sip:Unknown@192.168.1.123>;tag=as127aeb1b
        To: <sip:25@192.168.1.5:5060>;tag=as1589c556
        Call-ID: 7297b2ef5a77dc6a0d7edfba56926e54@192.168.1.123:506 0
        CSeq: 102 OPTIONS
        Server: FPBX-2.8.1(1.8.7.0)
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Contact: <sip:192.168.1.123:5060>
        Accept: application/sdp
        Content-Length: 0


        <------------>
        Scheduling destruction of SIP dialog '7297b2ef5a77dc6a0d7edfba56926e54@192.168.1.123:50 60' in 32000 ms (Method: OPTIONS)

        <--- SIP read from UDP:98.16.244.133:5060 --->
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK6e509c17;received =98.16.244.133;rport=5060
        From: "Unknown" <sip:Unknown@192.168.1.123>;tag=as127aeb1b
        To: <sip:25@192.168.1.5:5060>;tag=as1589c556
        Call-ID: 7297b2ef5a77dc6a0d7edfba56926e54@192.168.1.123:506 0
        CSeq: 102 OPTIONS
        Server: FPBX-2.8.1(1.8.7.0)
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Contact: <sip:192.168.1.123:5060>
        Accept: application/sdp
        Content-Length: 0

        <------------->
        --- (12 headers 0 lines) ---
        Really destroying SIP dialog '7297b2ef5a77dc6a0d7edfba56926e54@192.168.1.123:50 60' Method: OPTIONS

        Comentario

        • #5
          Re: Llamadas salientes se cortan

          Hola ese parece el debug de registro y acks, ademas que parece que por ahi hay una llamada anonima entrante por lo del contexto from-sip-external. Asegurate de tener la opcion allowguest=no en la seccion de sip settings.

          sobre el debug necesitamos el de la llamada completa. Si te cuesta trabajo encontrarlo fijate en la hora y depespues buscas esa hora en el log full de asterisk o bien ejecuta el comando "asterisk -rnvvvvvddddd | tee asterisk.log" sin comillas, esto creara un archivo llamado asterisk.log el cual tendra toda la salida de tu consola, para terminar la captura solo sal de asterisk.

          Comentario

          • #6
            Re: Llamadas salientes se cortan

            Muchísimas gracias.

            Se solucionó de la forma más rara: borré los trunks y las salidas, reinicié, las volví a crear...y magia-potagia.

            Muchas gracias por la atención.

            Comentario

            • #7
              Re: Llamadas salientes se cortan

              Vaya tela...hoy lo vuelvo a poner...y mi proveedor de VOIP está registrado, pero aparece en sip show peers como UNREACHABLE....

              No termino de pillarle el truco a esto....

              Comentario

              • #8
                Re: Llamadas salientes se cortan

                Hoy ha sido un día de pruebas, y el problema está en el cortafuegos y el NAT, pienso. Ahora las llamadas internas/externas están solucionadas, pero el voicemail no envía al e-mail ni manda señal de mensaje al teléfono (se le ponía antes un sobrecito).

                En fin, que esto lo pondré en otro post.

                Gracias y el tema de las llamadas solucionados gracias al direccionamiento de puertos.

                Saludos.

                Comentario

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