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Cacharrito spa 3102

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  • #1

    Cacharrito spa 3102

    Hola x aqui de nuevo , me pasa algo peculiar con un spa 3102 lo tengo registrado a un asterisk para hacer llamadas a la pstn , recibo llamadas a través de un IVR great , pero cuando quiero sacar llamadas no me tira me descuelga al segundo ... tengo la impresión que podría ser el Dial plan del spa 3102 ..

    alguna idea , posteare mas luego el dialplan del cacharro y el debug de asterisk ...

    #### llmadas a la pstn ###

    exten => _9.,1,Dial(SIP/${EXTEN:1}@sipurafxo,60,r)
    exten => _9.,n,Hangup()

    aqui al marcar se cae al segundo
    Etiquetas: Ninguno
  • #2
    Re: Cacharrito spa 3102

    Nos ayudaria mucho si arrojas el contenido de la consola al momento en que marcas a la PSTN. Me inclino a pensar que tienes una llamada en G729 y la segunda no puede tomar el mismo codec porque el equipo no aguanta las 2 con el mismo.
    dCAP Christian Cabrera R.
    Para aprender a usar Asterisk, asiste a uno de mis cursos Asterisk
    Si deseas asesoría pagada, por favor contáctame

    Comentario

    • #3
      Re: Cacharrito spa 3102

      aqui en mi pais los numeros telefonicos son de 8 digitos y cuando marco no me jala la llamada , pero si recibo perfecto.

      no veo problema de codec , tambien lo considere , incluso mira mi salida.

      aqui una salida a la pstn

      -- Executing [9226xxxxx@netsol:1] Dial("SIP/104-0eb7e0a0", "SIP/226xxxxx@sip urafxo|50|tT") in new stack
      -- Called 226xxxxx@sipurafxo
      -- SIP/sipurafxo-0eb838c0 is circuit-busy
      == Everyone is busy/congested at this time (1:0/1/0)
      -- Executing [922686894@netsol:2] Hangup("SIP/104-0eb7e0a0", "") in new stac k
      == Spawn extension (netsol, 9226xxxxx, 2) exited non-zero on 'SIP/104-0eb7e0a0 '


      ############sip debug###########

      me da un 404 no encontrado , lo raro es que recibo llamadas desde la pstn!

      interface: any
      filter: (ip or ip6) and ( port 5060 )
      #
      U +4.398697 192.168.10.39:5060 -> 192.168.10.7:5060
      INVITE sip:9226xxxxx@192.168.10.7:5060;user=phone SIP/2.0
      Via: SIP/2.0/UDP 192.168.10.39;branch=z9hG4bKa7832d4569C1BD42
      From: "Ricardo" <sip:104@192.168.10.7>;tag=CBF87EAE-9CF4C597
      To: <sip:9226xxxxx@192.168.10.7;user=phone>
      CSeq: 1 INVITE
      Call-ID: 90ab2b1b-57e74520-5cac6561@192.168.10.39
      Contact: <sip:104@192.168.10.39>
      Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
      User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049
      Supported: 100rel,replaces
      Allow-Events: talk,hold,conference
      Max-Forwards: 70
      Content-Type: application/sdp
      Content-Length: 225

      v=0
      o=- 1321319802 1321319802 IN IP4 192.168.10.39
      s=Polycom IP Phone
      c=IN IP4 192.168.10.39
      t=0 0
      m=audio 2222 RTP/AVP 0 8 101
      a=sendrecv
      a=rtpmap:0 PCMU/8000
      a=rtpmap:8 PCMA/8000
      a=rtpmap:101 telephone-event/8000

      #
      U +0.001273 192.168.10.7:5060 -> 192.168.10.39:5060
      SIP/2.0 407 Proxy Authentication Required
      Via: SIP/2.0/UDP 192.168.10.39;branch=z9hG4bKa7832d4569C1BD42;recei ved=192.168.10.39
      From: "Ricardo" <sip:104@192.168.10.7>;tag=CBF87EAE-9CF4C597
      To: <sip:9226xxxxx@192.168.10.7;user=phone>;tag=as6244 9efb
      Call-ID: 90ab2b1b-57e74520-5cac6561@192.168.10.39
      CSeq: 1 INVITE
      User-Agent: Asterisk PBX
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
      Supported: replaces
      Proxy-Authenticate: Digest algorithm=MD5, realm="netsoluciones-org", nonce="361d51b1"
      Content-Length: 0


      #
      U +0.016807 192.168.10.39:5060 -> 192.168.10.7:5060
      ACK sip:9226xxxxx@192.168.10.7:5060 SIP/2.0
      Via: SIP/2.0/UDP 192.168.10.39;branch=z9hG4bKa7832d4569C1BD42
      From: "Ricardo" <sip:104@192.168.10.7>;tag=CBF87EAE-9CF4C597
      To: <sip:9226xxxxx@192.168.10.7;user=phone>;tag=as6244 9efb
      CSeq: 1 ACK
      Call-ID: 90ab2b1b-57e74520-5cac6561@192.168.10.39
      Contact: <sip:104@192.168.10.39>
      Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
      User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049
      Max-Forwards: 70
      Content-Length: 0


      #
      U +0.003525 192.168.10.39:5060 -> 192.168.10.7:5060
      INVITE sip:9226xxxxx@192.168.10.7:5060;user=phone SIP/2.0
      Via: SIP/2.0/UDP 192.168.10.39;branch=z9hG4bKf6b456cFBED713D
      From: "Ricardo" <sip:104@192.168.10.7>;tag=CBF87EAE-9CF4C597
      To: <sip:9226xxxxx@192.168.10.7;user=phone>
      CSeq: 2 INVITE
      Call-ID: 90ab2b1b-57e74520-5cac6561@192.168.10.39
      Contact: <sip:104@192.168.10.39>
      Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
      User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049
      Supported: 100rel,replaces
      Allow-Events: talk,hold,conference
      Proxy-Authorization: Digest username="104", realm="netsoluciones-org", nonce="361d51b1", uri="sip:9226868
      94@192.168.10.7:5060;user=phone", response="1567cc9e0db4b77d5fdf0800938c04ab", algorithm=MD5
      Max-Forwards: 70
      Content-Type: application/sdp
      Content-Length: 225

      v=0
      o=- 1321319802 1321319802 IN IP4 192.168.10.39
      s=Polycom IP Phone
      c=IN IP4 192.168.10.39
      t=0 0
      m=audio 2222 RTP/AVP 0 8 101
      a=sendrecv
      a=rtpmap:0 PCMU/8000
      a=rtpmap:8 PCMA/8000
      a=rtpmap:101 telephone-event/8000

      #
      U +0.017405 192.168.10.7:5060 -> 192.168.10.39:5060
      SIP/2.0 100 Trying
      Via: SIP/2.0/UDP 192.168.10.39;branch=z9hG4bKf6b456cFBED713D;receiv ed=192.168.10.39
      From: "Ricardo" <sip:104@192.168.10.7>;tag=CBF87EAE-9CF4C597
      To: <sip:9226xxxxx@192.168.10.7;user=phone>
      Call-ID: 90ab2b1b-57e74520-5cac6561@192.168.10.39
      CSeq: 2 INVITE
      User-Agent: Asterisk PBX
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
      Supported: replaces
      Contact: <sip:9226xxxxx@192.168.10.7>
      Content-Length: 0


      #
      U +0.001643 192.168.10.7:5060 -> 192.168.10.5:5060
      INVITE sip:226xxxxx@192.168.10.5;user=phone SIP/2.0
      Via: SIP/2.0/UDP 192.168.10.7:5060;branch=z9hG4bK11f3055e;rport
      From: "Ricardo" <sip:104@192.168.10.7>;tag=as008e4401
      To: <sip:226xxxxx@192.168.10.5;user=phone>
      Contact: <sip:104@192.168.10.7>
      Call-ID: 63f288c93b429a67428f32271331cc87@192.168.10.7
      CSeq: 102 INVITE
      User-Agent: Asterisk PBX
      Max-Forwards: 70
      Date: Wed, 16 Nov 2011 01:21:05 GMT
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
      Supported: replaces
      Content-Type: application/sdp
      Content-Length: 262

      v=0
      o=root 3619 3619 IN IP4 192.168.10.7
      s=session
      c=IN IP4 192.168.10.7
      t=0 0
      m=audio 14512 RTP/AVP 0 8 101
      a=rtpmap:0 PCMU/8000
      a=rtpmap:8 PCMA/8000
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-16
      a=silenceSuppff - - - -
      a=ptime:20
      a=sendrecv

      #
      U +0.008605 192.168.10.5:5060 -> 192.168.10.7:5060
      SIP/2.0 404 Not Found
      To: <sip:226xxxxx@192.168.10.5;user=phone>;tag=a94a2a6 175b5a4e6i0
      From: "Ricardo" <sip:104@192.168.10.7>;tag=as008e4401
      Call-ID: 63f288c93b429a67428f32271331cc87@192.168.10.7
      CSeq: 102 INVITE
      Via: SIP/2.0/UDP 192.168.10.7:5060;branch=z9hG4bK11f3055e
      Server: Linksys/SPA3102-3.3.6(GW)
      Content-Length: 0


      #
      U +0.000561 192.168.10.7:5060 -> 192.168.10.5:5060
      ACK sip:226xxxxx@192.168.10.5;user=phone SIP/2.0
      Via: SIP/2.0/UDP 192.168.10.7:5060;branch=z9hG4bK11f3055e;rport
      From: "Ricardo" <sip:104@192.168.10.7>;tag=as008e4401
      To: <sip:226xxxxx@192.168.10.5;user=phone>;tag=a94a2a6 175b5a4e6i0
      Contact: <sip:104@192.168.10.7>
      Call-ID: 63f288c93b429a67428f32271331cc87@192.168.10.7
      CSeq: 102 ACK
      User-Agent: Asterisk PBX
      Max-Forwards: 70
      Content-Length: 0


      #
      U +0.000891 192.168.10.7:5060 -> 192.168.10.39:5060
      SIP/2.0 404 Not Found
      Via: SIP/2.0/UDP 192.168.10.39;branch=z9hG4bKf6b456cFBED713D;receiv ed=192.168.10.39
      From: "Ricardo" <sip:104@192.168.10.7>;tag=CBF87EAE-9CF4C597
      To: <sip:9226xxxxx@192.168.10.7;user=phone>;tag=as46e8 ac33
      Call-ID: 90ab2b1b-57e74520-5cac6561@192.168.10.39
      CSeq: 2 INVITE
      User-Agent: Asterisk PBX
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
      Supported: replaces
      Content-Length: 0


      #
      U +0.018330 192.168.10.39:5060 -> 192.168.10.7:5060
      ACK sip:9226xxxxx@192.168.10.7:5060 SIP/2.0
      Via: SIP/2.0/UDP 192.168.10.39;branch=z9hG4bKf6b456cFBED713D
      From: "Ricardo" <sip:104@192.168.10.7>;tag=CBF87EAE-9CF4C597
      To: <sip:9226xxxxx@192.168.10.7;user=phone>;tag=as46e8 ac33
      CSeq: 2 ACK
      Call-ID: 90ab2b1b-57e74520-5cac6561@192.168.10.39
      Contact: <sip:104@192.168.10.39>
      Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
      User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049
      Proxy-Authorization: Digest username="104", realm="netsoluciones-org", nonce="361d51b1", uri="sip:9226868
      94@192.168.10.7:5060;user=phone", response="1567cc9e0db4b77d5fdf0800938c04ab", algorithm=MD5
      Max-Forwards: 70
      Content-Length: 0


      exit

      Comentario

      • #4
        Re: Cacharrito spa 3102

        En efecto, parece ser problema del dialplan. ¿Ya revisaste que tengas el XXXXXXXX dentro del DP que hayas puesto para la linea FXO?

        Tambien recuerda que para permitir llamadas VoIP-FXO necesitas especificar que DP usarás.

        Saludos,
        dCAP Christian Cabrera R.
        Para aprender a usar Asterisk, asiste a uno de mis cursos Asterisk
        Si deseas asesoría pagada, por favor contáctame

        Comentario

        • #5
          Re: Cacharrito spa 3102

          Christian te muestro el dialplan q tiene el spa3102 dentro de Line1

          (*xxx|[3469]11|0|00|[2-9]xxxxxxx|1xxx[2-9]xxxxxxxS0|xxxxxxxxxxxx.)

          aqui en mi pais los numeros convencionales son de 8 digitos, = celular

          ahora el DP como lo especifico ? , aqui quede medio perdido

          sldss

          Comentario

          • #6
            Re: Cacharrito spa 3102

            En lo que afinas, déjalo abierto para todo. Ponlo como ( x. ) , así te permitirá cualquier cantidad de al menos 1 digito.

            Si eso te funciona, ya podremos ir restringiendo el dialplan para que solo permita lo que quieres

            Saludos,
            dCAP Christian Cabrera R.
            Para aprender a usar Asterisk, asiste a uno de mis cursos Asterisk
            Si deseas asesoría pagada, por favor contáctame

            Comentario

            • #7
              Re: Cacharrito spa 3102

              Originalmente publicado por Christian Cabrera
              En lo que afinas, déjalo abierto para todo. Ponlo como ( x. ) , así te permitirá cualquier cantidad de al menos 1 digito.
              hijue me estas hablando en chiness , que modifique el DP y dejarlo x.

              seria asi:
              (*xxx|[3469]11|0|00|[2-9]xxxxxxx|1xxx[2-9]xxxxxxxS0|x.)

              Comentario

              • #8
                Re: Cacharrito spa 3102

                Con peras y manzanas:

                Actualmente tienes un cuadro que dice "dialplan" y que tiene escrito el código (*xxx|[3469]11|0|00|[2-9]xxxxxxx|1xxx[2-9]xxxxxxxS0|x.)

                Edita ese recuadro y deja escrito solamente ( x. )

                Pruébalo, y dime como te fue
                dCAP Christian Cabrera R.
                Para aprender a usar Asterisk, asiste a uno de mis cursos Asterisk
                Si deseas asesoría pagada, por favor contáctame

                Comentario

                • #9
                  Re: Cacharrito spa 3102

                  slds , fijate que lo deje como me pediste y todavia no me jala

                  Saved useragent "Grandstream BT200 1.2.3.5" for peer 103
                  -- Executing [9226xxxxx@netsol:1] Set("SIP/103-17015840", "__SIP_CODEC=ulaw") in new stack
                  -- Executing [9226xxxxx@netsol:2] Dial("SIP/103-17015840", "SIP/226xxxxx@sipurafxo|50|tT") in new stack
                  -- Called 226xxxxx@sipurafxo
                  -- SIP/sipurafxo-17088800 is circuit-busy
                  == Everyone is busy/congested at this time (1:0/1/0)
                  -- Executing [9226xxxxx@netsol:3] Hangup("SIP/103-17015840", "") in new stack
                  == Spawn extension (netsol, 9226xxxxx, 3) exited non-zero on 'SIP/103-17015840'

                  christian aqui en el foro no hay una opcion de subir imagenes?

                  sldss

                  Comentario

                  • #10
                    Re: Cacharrito spa 3102

                    Hasta donde sé si la hay, súbelas como adjunto.
                    [attachment=0:23qteg2l]asterisk.png[/attachment:23qteg2l]

                    Activa el sip debug y muestra este mismo texto para ver que te arroja
                    dCAP Christian Cabrera R.
                    Para aprender a usar Asterisk, asiste a uno de mis cursos Asterisk
                    Si deseas asesoría pagada, por favor contáctame

                    Comentario

                    • #11
                      Re: Cacharrito spa 3102

                      upss , te envio a ver que puede ser

                      Comentario

                      • #12
                        Re: Cacharrito spa 3102

                        otro screen del pst line

                        Comentario

                        • #13
                          Re: Cacharrito spa 3102

                          esta es una llamada entrante y en curso de la pstn

                          Spawn extension (netsol, 9226xxxxx, 3) exited non-zero on 'SIP/104-09d3ebb0'
                          -- Executing [s@netsol:1] GotoIfTime("SIP/sipurafxo-09d443e0", "08:00-18:00|mon-fri|*|*?s|dentro") in new stack
                          -- Executing [s@netsol:2] Dial("SIP/sipurafxo-09d443e0", "SIP/104|86|Tt") in new stack
                          -- Called 104
                          -- SIP/104-09d487c0 is ringing



                          sip show channels
                          Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message
                          192.168.10.39 104 03802bb1157 00102/00000 0x4 (ulaw) No Tx: ACK
                          192.168.10.5 sipurafxo 2d6495fe-a9 00101/00101 0x4 (ulaw) No Rx: ACK

                          = 3624)
                          Verbosity is at least 36
                          == Spawn extension (netsol, 0, 1) exited non-zero on 'SIP/sipurafxo-09d3ebb0'
                          == End MixMonitor Recording SIP/sipurafxo-09d3ebb0


                          recibo bien llamadas

                          Comentario

                          • #14
                            Re: Cacharrito spa 3102

                            necesito ver el sip debug al momento en que mandas la llamada al exterior
                            dCAP Christian Cabrera R.
                            Para aprender a usar Asterisk, asiste a uno de mis cursos Asterisk
                            Si deseas asesoría pagada, por favor contáctame

                            Comentario

                            • #15
                              Re: Cacharrito spa 3102

                              aqui va un debug del sip


                              interface: any
                              filter: (ip or ip6) and ( port 5060 )
                              #
                              U +4.028638 192.168.10.39:5060 -> 192.168.10.7:5060
                              INVITE sip:9226xxxxx@192.168.10.7:5060;user=phone SIP/2.0
                              Via: SIP/2.0/UDP 192.168.10.39;branch=z9hG4bK72c788e570E4E63A
                              From: "Ricardo" <sip:104@192.168.10.7>;tag=34DC42D6-AC6F143F
                              To: <sip:9226xxxxx@192.168.10.7;user=phone>
                              CSeq: 1 INVITE
                              Call-ID: a13ab873-fdce3e60-e8e42311@192.168.10.39
                              Contact: <sip:104@192.168.10.39>
                              Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
                              User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049
                              Supported: 100rel,replaces
                              Allow-Events: talk,hold,conference
                              Max-Forwards: 70
                              Content-Type: application/sdp
                              Content-Length: 225

                              v=0
                              o=- 1321659243 1321659243 IN IP4 192.168.10.39
                              s=Polycom IP Phone
                              c=IN IP4 192.168.10.39
                              t=0 0
                              m=audio 2224 RTP/AVP 0 8 101
                              a=sendrecv
                              a=rtpmap:0 PCMU/8000
                              a=rtpmap:8 PCMA/8000
                              a=rtpmap:101 telephone-event/8000

                              #
                              U +0.000852 192.168.10.7:5060 -> 192.168.10.39:5060
                              SIP/2.0 407 Proxy Authentication Required
                              Via: SIP/2.0/UDP 192.168.10.39;branch=z9hG4bK72c788e570E4E63A;recei ved=192.168.10.39
                              From: "Ricardo" <sip:104@192.168.10.7>;tag=34DC42D6-AC6F143F
                              To: <sip:9226xxxxx@192.168.10.7;user=phone>;tag=as6793 cfa5
                              Call-ID: a13ab873-fdce3e60-e8e42311@192.168.10.39
                              CSeq: 1 INVITE
                              User-Agent: Asterisk PBX
                              Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                              Supported: replaces
                              Proxy-Authenticate: Digest algorithm=MD5, realm="netsoluciones-org", nonce="15ab02cd"
                              Content-Length: 0


                              #
                              U +0.017305 192.168.10.39:5060 -> 192.168.10.7:5060
                              ACK sip:9226xxxxx@192.168.10.7:5060 SIP/2.0
                              Via: SIP/2.0/UDP 192.168.10.39;branch=z9hG4bK72c788e570E4E63A
                              From: "Ricardo" <sip:104@192.168.10.7>;tag=34DC42D6-AC6F143F
                              To: <sip:9226xxxxx@192.168.10.7;user=phone>;tag=as6793 cfa5
                              CSeq: 1 ACK
                              Call-ID: a13ab873-fdce3e60-e8e42311@192.168.10.39
                              Contact: <sip:104@192.168.10.39>
                              Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
                              User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049
                              Max-Forwards: 70
                              Content-Length: 0


                              #
                              U +0.002893 192.168.10.39:5060 -> 192.168.10.7:5060
                              INVITE sip:9226xxxxx@192.168.10.7:5060;user=phone SIP/2.0
                              Via: SIP/2.0/UDP 192.168.10.39;branch=z9hG4bK86fa851cFF645B7D
                              From: "Ricardo" <sip:104@192.168.10.7>;tag=34DC42D6-AC6F143F
                              To: <sip:9226xxxxx@192.168.10.7;user=phone>
                              CSeq: 2 INVITE
                              Call-ID: a13ab873-fdce3e60-e8e42311@192.168.10.39
                              Contact: <sip:104@192.168.10.39>
                              Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
                              User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049
                              Supported: 100rel,replaces
                              Allow-Events: talk,hold,conference
                              Proxy-Authorization: Digest username="104", realm="netsoluciones-org", nonce="15ab02cd", uri="sip:9226xxxxx
                              @192.168.10.7:5060;user=phone", response="9060a8371d4a3b903e90fed8a735d847", algorithm=MD5
                              Max-Forwards: 70
                              Content-Type: application/sdp
                              Content-Length: 225

                              v=0
                              o=- 1321659243 1321659243 IN IP4 192.168.10.39
                              s=Polycom IP Phone
                              c=IN IP4 192.168.10.39
                              t=0 0
                              m=audio 2224 RTP/AVP 0 8 101
                              a=sendrecv
                              a=rtpmap:0 PCMU/8000
                              a=rtpmap:8 PCMA/8000
                              a=rtpmap:101 telephone-event/8000

                              #
                              U +0.000471 192.168.10.7:5060 -> 192.168.10.39:5060
                              SIP/2.0 100 Trying
                              Via: SIP/2.0/UDP 192.168.10.39;branch=z9hG4bK86fa851cFF645B7D;recei ved=192.168.10.39
                              From: "Ricardo" <sip:104@192.168.10.7>;tag=34DC42D6-AC6F143F
                              To: <sip:9226xxxxx@192.168.10.7;user=phone>
                              Call-ID: a13ab873-fdce3e60-e8e42311@192.168.10.39
                              CSeq: 2 INVITE
                              User-Agent: Asterisk PBX
                              Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                              Supported: replaces
                              Contact: <sip:9226xxxxx@192.168.10.7>
                              Content-Length: 0


                              #
                              U +0.001871 192.168.10.7:5060 -> 192.168.10.5:5060
                              INVITE sip:226xxxxx@192.168.10.5;user=phone SIP/2.0
                              Via: SIP/2.0/UDP 192.168.10.7:5060;branch=z9hG4bK3770405c;rport
                              From: "Ricardo" <sip:104@192.168.10.7>;tag=as1a4c2c36
                              To: <sip:226xxxxx@192.168.10.5;user=phone>
                              Contact: <sip:104@192.168.10.7>
                              Call-ID: 71b7cf051f3f67111c46ddcb08ba6b3e@192.168.10.7
                              CSeq: 102 INVITE
                              User-Agent: Asterisk PBX
                              Max-Forwards: 70
                              Date: Sat, 19 Nov 2011 22:16:11 GMT
                              Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                              Supported: replaces
                              Content-Type: application/sdp
                              Content-Length: 262

                              v=0
                              o=root 3634 3634 IN IP4 192.168.10.7
                              s=session
                              c=IN IP4 192.168.10.7
                              t=0 0
                              m=audio 14296 RTP/AVP 0 8 101
                              a=rtpmap:0 PCMU/8000
                              a=rtpmap:8 PCMA/8000
                              a=rtpmap:101 telephone-event/8000
                              a=fmtp:101 0-16
                              a=silenceSuppff - - - -
                              a=ptime:20
                              a=sendrecv

                              #
                              U +0.014771 192.168.10.5:5060 -> 192.168.10.7:5060
                              SIP/2.0 404 Not Found
                              To: <sip:226xxxxx@192.168.10.5;user=phone>;tag=2c22779 f910088c2i0
                              From: "Ricardo" <sip:104@192.168.10.7>;tag=as1a4c2c36
                              Call-ID: 71b7cf051f3f67111c46ddcb08ba6b3e@192.168.10.7
                              CSeq: 102 INVITE
                              Via: SIP/2.0/UDP 192.168.10.7:5060;branch=z9hG4bK3770405c
                              Server: Linksys/SPA3102-3.3.6(GW)
                              Content-Length: 0


                              #
                              U +0.000417 192.168.10.7:5060 -> 192.168.10.5:5060
                              ACK sip:226xxxxx@192.168.10.5;user=phone SIP/2.0
                              Via: SIP/2.0/UDP 192.168.10.7:5060;branch=z9hG4bK3770405c;rport
                              From: "Ricardo" <sip:104@192.168.10.7>;tag=as1a4c2c36
                              To: <sip:226xxxxx@192.168.10.5;user=phone>;tag=2c22779 f910088c2i0
                              Contact: <sip:104@192.168.10.7>
                              Call-ID: 71b7cf051f3f67111c46ddcb08ba6b3e@192.168.10.7
                              CSeq: 102 ACK
                              User-Agent: Asterisk PBX
                              Max-Forwards: 70
                              Content-Length: 0


                              #
                              U +0.000906 192.168.10.7:5060 -> 192.168.10.39:5060
                              SIP/2.0 404 Not Found
                              Via: SIP/2.0/UDP 192.168.10.39;branch=z9hG4bK86fa851cFF645B7D;recei ved=192.168.10.39
                              From: "Ricardo" <sip:104@192.168.10.7>;tag=34DC42D6-AC6F143F
                              To: <sip:9226xxxxx@192.168.10.7;user=phone>;tag=as4ae1 fcf0
                              Call-ID: a13ab873-fdce3e60-e8e42311@192.168.10.39
                              CSeq: 2 INVITE
                              User-Agent: Asterisk PBX
                              Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                              Supported: replaces
                              Content-Length: 0


                              #
                              U +0.016114 192.168.10.39:5060 -> 192.168.10.7:5060
                              ACK sip:9226xxxxx@192.168.10.7:5060 SIP/2.0
                              Via: SIP/2.0/UDP 192.168.10.39;branch=z9hG4bK86fa851cFF645B7D
                              From: "Ricardo" <sip:104@192.168.10.7>;tag=34DC42D6-AC6F143F
                              To: <sip:9226xxxxx@192.168.10.7;user=phone>;tag=as4ae1 fcf0
                              CSeq: 2 ACK
                              Call-ID: a13ab873-fdce3e60-e8e42311@192.168.10.39
                              Contact: <sip:104@192.168.10.39>
                              Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
                              User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049
                              Proxy-Authorization: Digest username="104", realm="netsoluciones-org", nonce="15ab02cd", uri="sip:9226xxxxx
                              @192.168.10.7:5060;user=phone", response="9060a8371d4a3b903e90fed8a735d847", algorithm=MD5
                              Max-Forwards: 70
                              Content-Length: 0


                              exit
                              10 received, 0 dropped

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