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Problema de sonido con SPA 942

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  • #1

    Problema de sonido con SPA 942

    Buenos días a todos:

    Espero me puedan resolver esta duda, dado que mi provedor de Elastix no le ha atinado.

    Tengo un elastix 2.4.0 que instalo otra compañia.

    spa8000 de 8 lineas y spa962, spa 942.

    El problema es que al intentar cascadear mi switch cisco sf200-24 a otro switch inteligente los telefonos spa pierden audio. Es decir de 10 llamadas realizadas una o dos no tienen audio.

    De codecs tengo instalados ulaw y alaw.

    Como prueba instale una version limpia en mi laptop en una maquina virtual conectado a mi switch con un spa8000 y un spa942, el spa 942 igual no tiene audio en una o dos llamadas de 10 que hago.

    Este problema lo tengo tanto en llamadas entre extensiones como a la calle.

    ya probe varios firmware para el spa942 desde el primero que salio hasta el ultimo actualizado en el site de Cisco y no se solucionó el problema.

    alguna idea?

    Agradezco de antemano sus comentarios!
    Etiquetas: Ninguno
  • #2
    Es muy raro: un switch no debe bloquear los puertos. Me ha pasado de que eso ocurre en los Cisco por la manera en que se van asignando los RTP de manera aleatoria.

    Ejecuta el SIP debug y pega el debug de una llamada de las que NO tiene audio. Tambien pon una de las que SI tiene audio para comparar las diferencias.
    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
      Hola perdon por la tardanza este es el debug cuando si tuve sonido:

      Audio is at 19716
      Adding codec 0x4 (ulaw) to SDP
      Adding codec 0x8 (alaw) to SDP
      Adding non-codec 0x1 (telephone-event) to SDP
      Reliably Transmitting (NAT) to 192.168.5.4:5060:
      INVITE sip:667@192.168.5.4:5060 SIP/2.0
      Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK04594692;rport
      Max-Forwards: 70
      From: "Sistemas" <sip:510@192.168.5.1>;tag=as31cc823c
      To: <sip:667@192.168.5.4:5060>
      Contact: <sip:510@192.168.5.1:5060>
      Call-ID: 16086eb71ec8a2e04c5bb4ac03ba372c@192.168.5.1:5060
      CSeq: 102 INVITE
      User-Agent: FPBX-2.8.1(1.8.20.0)
      Date: Fri, 04 Jul 2014 22:11:15 GMT
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
      Supported: replaces, timer
      Content-Type: application/sdp
      Content-Length: 257

      v=0
      o=root 819409044 819409044 IN IP4 192.168.5.1
      s=Asterisk PBX 1.8.20.0
      c=IN IP4 192.168.5.1
      t=0 0
      m=audio 19716 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=ptime:20
      a=sendrecv

      ---
      -- Called SIP/667

      <--- SIP read from UDP:192.168.5.4:5060 --->
      SIP/2.0 100 Trying
      To: <sip:667@192.168.5.4:5060>
      From: "Sistemas" <sip:510@192.168.5.1>;tag=as31cc823c
      Call-ID: 16086eb71ec8a2e04c5bb4ac03ba372c@192.168.5.1:5060
      CSeq: 102 INVITE
      Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK04594692
      Server: Linksys/SPA922-5.2.8
      Content-Length: 0

      <------------->
      --- (8 headers 0 lines) ---

      <--- SIP read from UDP:192.168.5.4:5060 --->
      SIP/2.0 180 Ringing
      To: <sip:667@192.168.5.4:5060>;tag=7b2a90d3ce4a98eai 0
      From: "Sistemas" <sip:510@192.168.5.1>;tag=as31cc823c
      Call-ID: 16086eb71ec8a2e04c5bb4ac03ba372c@192.168.5.1:5060
      CSeq: 102 INVITE
      Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK04594692
      Server: Linksys/SPA922-5.2.8
      Content-Length: 0

      <------------->
      --- (8 headers 0 lines) ---
      list_route: no route
      -- SIP/667-00007826 is ringing

      <--- SIP read from UDP:192.168.5.4:5060 --->
      SIP/2.0 200 OK
      To: <sip:667@192.168.5.4:5060>;tag=7b2a90d3ce4a98eai 0
      From: "Sistemas" <sip:510@192.168.5.1>;tag=as31cc823c
      Call-ID: 16086eb71ec8a2e04c5bb4ac03ba372c@192.168.5.1:5060
      CSeq: 102 INVITE
      Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK04594692
      Contact: "667" <sip:667@192.168.5.4:5060>
      Server: Linksys/SPA922-5.2.8
      Content-Length: 202
      Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
      Supported: replaces
      Content-Type: application/sdp

      v=0
      o=- 31010 31010 IN IP4 192.168.5.4
      s=-
      c=IN IP4 192.168.5.4
      t=0 0
      m=audio 16434 RTP/AVP 0 101
      a=rtpmap:0 PCMU/8000
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-15
      a=ptime:30
      a=sendrecv
      <------------->
      --- (12 headers 11 lines) ---
      Found RTP audio format 0
      Found RTP audio format 101
      Found audio description format PCMU for ID 0
      Found audio description format telephone-event for ID 101
      Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
      Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
      Peer audio RTP is at port 192.168.5.4:16434
      list_route: hop: <sip:667@192.168.5.4:5060>
      set_destination: Parsing <sip:667@192.168.5.4:5060> for address/port to send to
      set_destination: set destination to 192.168.5.4:5060
      Transmitting (NAT) to 192.168.5.4:5060:
      ACK sip:667@192.168.5.4:5060 SIP/2.0
      Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK24163926;rport
      Max-Forwards: 70
      From: "Sistemas" <sip:510@192.168.5.1>;tag=as31cc823c
      To: <sip:667@192.168.5.4:5060>;tag=7b2a90d3ce4a98eai 0
      Contact: <sip:510@192.168.5.1:5060>
      Call-ID: 16086eb71ec8a2e04c5bb4ac03ba372c@192.168.5.1:5060
      CSeq: 102 ACK
      User-Agent: FPBX-2.8.1(1.8.20.0)
      Content-Length: 0




      --------------------------------------------------------------


      Este es el debug cuando no hubo sonido



      <--- SIP read from UDP:192.168.5.4:5060 --->
      REGISTER sip:192.168.5.1:5060 SIP/2.0
      Via: SIP/2.0/UDP 192.168.5.4:5060;branch=z9hG4bK-a566268d
      From: "667" <sip:667@192.168.5.1>;tag=f64a22234c67433do0
      To: "667" <sip:667@192.168.5.1>
      Call-ID: 4e5ece5a-475829b7@192.168.5.4
      CSeq: 48655 REGISTER
      Max-Forwards: 70
      Authorization: Digest username="667",realm="asterisk",nonce="7e50ad6b",u ri="sip:192.168.5.1:5060",algorithm=MD5,response=" 4b5f72567f847715dd9add86ff79f4d4"
      Contact: "667" <sip:667@192.168.5.4:5060>;expires=90
      User-Agent: Linksys/SPA922-5.2.8
      Content-Length: 0
      Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
      Supported: replaces

      <------------->
      --- (13 headers 0 lines) ---
      Sending to 192.168.5.4:5060 (NAT)

      <--- Transmitting (NAT) to 192.168.5.4:5060 --->
      SIP/2.0 401 Unauthorized
      Via: SIP/2.0/UDP 192.168.5.4:5060;branch=z9hG4bK-a566268d;received=192.168.5.4;rport=5060
      From: "667" <sip:667@192.168.5.1>;tag=f64a22234c67433do0
      To: "667" <sip:667@192.168.5.1>;tag=as57b1d4fc
      Call-ID: 4e5ece5a-475829b7@192.168.5.4
      CSeq: 48655 REGISTER
      Server: FPBX-2.8.1(1.8.20.0)
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
      Supported: replaces, timer
      WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="664c80a8"
      Content-Length: 0


      <------------>
      Scheduling destruction of SIP dialog '4e5ece5a-475829b7@192.168.5.4' in 32000 ms (Method: REGISTER)

      <--- SIP read from UDP:192.168.5.4:5060 --->
      REGISTER sip:192.168.5.1:5060 SIP/2.0
      Via: SIP/2.0/UDP 192.168.5.4:5060;branch=z9hG4bK-26aca596
      From: "667" <sip:667@192.168.5.1>;tag=f64a22234c67433do0
      To: "667" <sip:667@192.168.5.1>
      Call-ID: 4e5ece5a-475829b7@192.168.5.4
      CSeq: 48656 REGISTER
      Max-Forwards: 70
      Authorization: Digest username="667",realm="asterisk",nonce="664c80a8",u ri="sip:192.168.5.1:5060",algorithm=MD5,response=" f134723246a4743b2e7daa634d774904"
      Contact: "667" <sip:667@192.168.5.4:5060>;expires=90
      User-Agent: Linksys/SPA922-5.2.8
      Content-Length: 0
      Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
      Supported: replaces

      <------------->
      --- (13 headers 0 lines) ---
      Sending to 192.168.5.4:5060 (NAT)
      Reliably Transmitting (NAT) to 192.168.5.4:5060:
      OPTIONS sip:667@192.168.5.4:5060 SIP/2.0
      Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK2dab329d;rport
      Max-Forwards: 70
      From: "Unknown" <sip:Unknown@192.168.5.1>;tag=as6ae4718f
      To: <sip:667@192.168.5.4:5060>
      Contact: <sip:Unknown@192.168.5.1:5060>
      Call-ID: 5aa23009085f5b112ac66a747d4a703c@192.168.5.1:5060
      CSeq: 102 OPTIONS
      User-Agent: FPBX-2.8.1(1.8.20.0)
      Date: Fri, 04 Jul 2014 22:14:03 GMT
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
      Supported: replaces, timer
      Content-Length: 0


      ---

      <--- Transmitting (NAT) to 192.168.5.4:5060 --->
      SIP/2.0 200 OK
      Via: SIP/2.0/UDP 192.168.5.4:5060;branch=z9hG4bK-26aca596;received=192.168.5.4;rport=5060
      From: "667" <sip:667@192.168.5.1>;tag=f64a22234c67433do0
      To: "667" <sip:667@192.168.5.1>;tag=as57b1d4fc
      Call-ID: 4e5ece5a-475829b7@192.168.5.4
      CSeq: 48656 REGISTER
      Server: FPBX-2.8.1(1.8.20.0)
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
      Supported: replaces, timer
      Expires: 90
      Contact: <sip:667@192.168.5.4:5060>;expires=90
      Date: Fri, 04 Jul 2014 22:14:03 GMT
      Content-Length: 0


      <------------>
      Scheduling destruction of SIP dialog '4f0056c153a1d4a67c8bebe01a109443@192.168.5.1:5060 ' in 6400 ms (Method: NOTIFY)
      Reliably Transmitting (NAT) to 192.168.5.4:5060:
      NOTIFY sip:667@192.168.5.4:5060 SIP/2.0
      Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK3a378592;rport
      Max-Forwards: 70
      From: "Unknown" <sip:Unknown@192.168.5.1>;tag=as6c285bb0
      To: <sip:667@192.168.5.4:5060>
      Contact: <sip:Unknown@192.168.5.1:5060>
      Call-ID: 4f0056c153a1d4a67c8bebe01a109443@192.168.5.1:5060
      CSeq: 102 NOTIFY
      User-Agent: FPBX-2.8.1(1.8.20.0)
      Event: message-summary
      Content-Type: application/simple-message-summary
      Content-Length: 86

      Messages-Waiting: no
      Message-Account: sip:*97@192.168.5.1
      Voice-Message: 0/0 (0/0)

      ---
      Scheduling destruction of SIP dialog '4e5ece5a-475829b7@192.168.5.4' in 32000 ms (Method: REGISTER)

      <--- SIP read from UDP:192.168.5.4:5060 --->
      SIP/2.0 200 OK
      To: <sip:667@192.168.5.4:5060>;tag=d446c09b46d27725i 0
      From: "Unknown" <sip:Unknown@192.168.5.1>;tag=as6ae4718f
      Call-ID: 5aa23009085f5b112ac66a747d4a703c@192.168.5.1:5060
      CSeq: 102 OPTIONS
      Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK2dab329d
      Server: Linksys/SPA922-5.2.8
      Content-Length: 0
      Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
      Supported: replaces

      <------------->
      --- (10 headers 0 lines) ---
      Really destroying SIP dialog '5aa23009085f5b112ac66a747d4a703c@192.168.5.1:5060 ' Method: OPTIONS

      <--- SIP read from UDP:192.168.5.4:5060 --->
      SIP/2.0 200 OK
      To: <sip:667@192.168.5.4:5060>;tag=d446c09b46d27725i 0
      From: "Unknown" <sip:Unknown@192.168.5.1>;tag=as6c285bb0
      Call-ID: 4f0056c153a1d4a67c8bebe01a109443@192.168.5.1:5060
      CSeq: 102 NOTIFY
      Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK3a378592
      Server: Linksys/SPA922-5.2.8
      Content-Length: 0



      Gracias de antemano!

      Comentario

      • #4
        Estos son los debug

        cuando si tengo sonido


        Audio is at 19716
        Adding codec 0x4 (ulaw) to SDP
        Adding codec 0x8 (alaw) to SDP
        Adding non-codec 0x1 (telephone-event) to SDP
        Reliably Transmitting (NAT) to 192.168.5.4:5060:
        INVITE sip:667@192.168.5.4:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK04594692;rport
        Max-Forwards: 70
        From: "Sistemas" <sip:510@192.168.5.1>;tag=as31cc823c
        To: <sip:667@192.168.5.4:5060>
        Contact: <sip:510@192.168.5.1:5060>
        Call-ID: 16086eb71ec8a2e04c5bb4ac03ba372c@192.168.5.1:5060
        CSeq: 102 INVITE
        User-Agent: FPBX-2.8.1(1.8.20.0)
        Date: Fri, 04 Jul 2014 22:11:15 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Content-Type: application/sdp
        Content-Length: 257

        v=0
        o=root 819409044 819409044 IN IP4 192.168.5.1
        s=Asterisk PBX 1.8.20.0
        c=IN IP4 192.168.5.1
        t=0 0
        m=audio 19716 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=ptime:20
        a=sendrecv

        ---
        -- Called SIP/667

        <--- SIP read from UDP:192.168.5.4:5060 --->
        SIP/2.0 100 Trying
        To: <sip:667@192.168.5.4:5060>
        From: "Sistemas" <sip:510@192.168.5.1>;tag=as31cc823c
        Call-ID: 16086eb71ec8a2e04c5bb4ac03ba372c@192.168.5.1:5060
        CSeq: 102 INVITE
        Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK04594692
        Server: Linksys/SPA922-5.2.8
        Content-Length: 0

        <------------->
        --- (8 headers 0 lines) ---

        <--- SIP read from UDP:192.168.5.4:5060 --->
        SIP/2.0 180 Ringing
        To: <sip:667@192.168.5.4:5060>;tag=7b2a90d3ce4a98eai 0
        From: "Sistemas" <sip:510@192.168.5.1>;tag=as31cc823c
        Call-ID: 16086eb71ec8a2e04c5bb4ac03ba372c@192.168.5.1:5060
        CSeq: 102 INVITE
        Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK04594692
        Server: Linksys/SPA922-5.2.8
        Content-Length: 0

        <------------->
        --- (8 headers 0 lines) ---
        list_route: no route
        -- SIP/667-00007826 is ringing

        <--- SIP read from UDP:192.168.5.4:5060 --->
        SIP/2.0 200 OK
        To: <sip:667@192.168.5.4:5060>;tag=7b2a90d3ce4a98eai 0
        From: "Sistemas" <sip:510@192.168.5.1>;tag=as31cc823c
        Call-ID: 16086eb71ec8a2e04c5bb4ac03ba372c@192.168.5.1:5060
        CSeq: 102 INVITE
        Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK04594692
        Contact: "667" <sip:667@192.168.5.4:5060>
        Server: Linksys/SPA922-5.2.8
        Content-Length: 202
        Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
        Supported: replaces
        Content-Type: application/sdp

        v=0
        o=- 31010 31010 IN IP4 192.168.5.4
        s=-
        c=IN IP4 192.168.5.4
        t=0 0
        m=audio 16434 RTP/AVP 0 101
        a=rtpmap:0 PCMU/8000
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-15
        a=ptime:30
        a=sendrecv
        <------------->
        --- (12 headers 11 lines) ---
        Found RTP audio format 0
        Found RTP audio format 101
        Found audio description format PCMU for ID 0
        Found audio description format telephone-event for ID 101
        Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
        Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
        Peer audio RTP is at port 192.168.5.4:16434
        list_route: hop: <sip:667@192.168.5.4:5060>
        set_destination: Parsing <sip:667@192.168.5.4:5060> for address/port to send to
        set_destination: set destination to 192.168.5.4:5060
        Transmitting (NAT) to 192.168.5.4:5060:
        ACK sip:667@192.168.5.4:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK24163926;rport
        Max-Forwards: 70
        From: "Sistemas" <sip:510@192.168.5.1>;tag=as31cc823c
        To: <sip:667@192.168.5.4:5060>;tag=7b2a90d3ce4a98eai 0
        Contact: <sip:510@192.168.5.1:5060>
        Call-ID: 16086eb71ec8a2e04c5bb4ac03ba372c@192.168.5.1:5060
        CSeq: 102 ACK
        User-Agent: FPBX-2.8.1(1.8.20.0)
        Content-Length: 0


        -----------------------------------------

        cuando no tengo sonido


        <--- SIP read from UDP:192.168.5.4:5060 --->
        REGISTER sip:192.168.5.1:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.5.4:5060;branch=z9hG4bK-a566268d
        From: "667" <sip:667@192.168.5.1>;tag=f64a22234c67433do0
        To: "667" <sip:667@192.168.5.1>
        Call-ID: 4e5ece5a-475829b7@192.168.5.4
        CSeq: 48655 REGISTER
        Max-Forwards: 70
        Authorization: Digest username="667",realm="asterisk",nonce="7e50ad6b",u ri="sip:192.168.5.1:5060",algorithm=MD5,response=" 4b5f72567f847715dd9add86ff79f4d4"
        Contact: "667" <sip:667@192.168.5.4:5060>;expires=90
        User-Agent: Linksys/SPA922-5.2.8
        Content-Length: 0
        Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
        Supported: replaces

        <------------->
        --- (13 headers 0 lines) ---
        Sending to 192.168.5.4:5060 (NAT)

        <--- Transmitting (NAT) to 192.168.5.4:5060 --->
        SIP/2.0 401 Unauthorized
        Via: SIP/2.0/UDP 192.168.5.4:5060;branch=z9hG4bK-a566268d;received=192.168.5.4;rport=5060
        From: "667" <sip:667@192.168.5.1>;tag=f64a22234c67433do0
        To: "667" <sip:667@192.168.5.1>;tag=as57b1d4fc
        Call-ID: 4e5ece5a-475829b7@192.168.5.4
        CSeq: 48655 REGISTER
        Server: FPBX-2.8.1(1.8.20.0)
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="664c80a8"
        Content-Length: 0


        <------------>
        Scheduling destruction of SIP dialog '4e5ece5a-475829b7@192.168.5.4' in 32000 ms (Method: REGISTER)

        <--- SIP read from UDP:192.168.5.4:5060 --->
        REGISTER sip:192.168.5.1:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.5.4:5060;branch=z9hG4bK-26aca596
        From: "667" <sip:667@192.168.5.1>;tag=f64a22234c67433do0
        To: "667" <sip:667@192.168.5.1>
        Call-ID: 4e5ece5a-475829b7@192.168.5.4
        CSeq: 48656 REGISTER
        Max-Forwards: 70
        Authorization: Digest username="667",realm="asterisk",nonce="664c80a8",u ri="sip:192.168.5.1:5060",algorithm=MD5,response=" f134723246a4743b2e7daa634d774904"
        Contact: "667" <sip:667@192.168.5.4:5060>;expires=90
        User-Agent: Linksys/SPA922-5.2.8
        Content-Length: 0
        Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
        Supported: replaces

        <------------->
        --- (13 headers 0 lines) ---
        Sending to 192.168.5.4:5060 (NAT)
        Reliably Transmitting (NAT) to 192.168.5.4:5060:
        OPTIONS sip:667@192.168.5.4:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK2dab329d;rport
        Max-Forwards: 70
        From: "Unknown" <sip:Unknown@192.168.5.1>;tag=as6ae4718f
        To: <sip:667@192.168.5.4:5060>
        Contact: <sip:Unknown@192.168.5.1:5060>
        Call-ID: 5aa23009085f5b112ac66a747d4a703c@192.168.5.1:5060
        CSeq: 102 OPTIONS
        User-Agent: FPBX-2.8.1(1.8.20.0)
        Date: Fri, 04 Jul 2014 22:14:03 GMT
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Content-Length: 0


        ---

        <--- Transmitting (NAT) to 192.168.5.4:5060 --->
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP 192.168.5.4:5060;branch=z9hG4bK-26aca596;received=192.168.5.4;rport=5060
        From: "667" <sip:667@192.168.5.1>;tag=f64a22234c67433do0
        To: "667" <sip:667@192.168.5.1>;tag=as57b1d4fc
        Call-ID: 4e5ece5a-475829b7@192.168.5.4
        CSeq: 48656 REGISTER
        Server: FPBX-2.8.1(1.8.20.0)
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Expires: 90
        Contact: <sip:667@192.168.5.4:5060>;expires=90
        Date: Fri, 04 Jul 2014 22:14:03 GMT
        Content-Length: 0


        <------------>
        Scheduling destruction of SIP dialog '4f0056c153a1d4a67c8bebe01a109443@192.168.5.1:5060 ' in 6400 ms (Method: NOTIFY)
        Reliably Transmitting (NAT) to 192.168.5.4:5060:
        NOTIFY sip:667@192.168.5.4:5060 SIP/2.0
        Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK3a378592;rport
        Max-Forwards: 70
        From: "Unknown" <sip:Unknown@192.168.5.1>;tag=as6c285bb0
        To: <sip:667@192.168.5.4:5060>
        Contact: <sip:Unknown@192.168.5.1:5060>
        Call-ID: 4f0056c153a1d4a67c8bebe01a109443@192.168.5.1:5060
        CSeq: 102 NOTIFY
        User-Agent: FPBX-2.8.1(1.8.20.0)
        Event: message-summary
        Content-Type: application/simple-message-summary
        Content-Length: 86

        Messages-Waiting: no
        Message-Account: sip:*97@192.168.5.1
        Voice-Message: 0/0 (0/0)

        ---
        Scheduling destruction of SIP dialog '4e5ece5a-475829b7@192.168.5.4' in 32000 ms (Method: REGISTER)

        <--- SIP read from UDP:192.168.5.4:5060 --->
        SIP/2.0 200 OK
        To: <sip:667@192.168.5.4:5060>;tag=d446c09b46d27725i 0
        From: "Unknown" <sip:Unknown@192.168.5.1>;tag=as6ae4718f
        Call-ID: 5aa23009085f5b112ac66a747d4a703c@192.168.5.1:5060
        CSeq: 102 OPTIONS
        Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK2dab329d
        Server: Linksys/SPA922-5.2.8
        Content-Length: 0
        Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
        Supported: replaces

        <------------->
        --- (10 headers 0 lines) ---
        Really destroying SIP dialog '5aa23009085f5b112ac66a747d4a703c@192.168.5.1:5060 ' Method: OPTIONS

        <--- SIP read from UDP:192.168.5.4:5060 --->
        SIP/2.0 200 OK
        To: <sip:667@192.168.5.4:5060>;tag=d446c09b46d27725i 0
        From: "Unknown" <sip:Unknown@192.168.5.1>;tag=as6c285bb0
        Call-ID: 4f0056c153a1d4a67c8bebe01a109443@192.168.5.1:5060
        CSeq: 102 NOTIFY
        Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK3a378592
        Server: Linksys/SPA922-5.2.8
        Content-Length: 0


        Saludos!

        Comentario

        • #5
          Perdon ya pegue el debug dos veces desde el viernes pero parece que todavia no aparece mi mensaje aprobado.

          Comentario

          • #6
            Este es el debug cuando tengo sonido


            Audio is at 11406
            Adding codec 0x4 (ulaw) to SDP
            Adding codec 0x8 (alaw) to SDP
            Adding non-codec 0x1 (telephone-event) to SDP
            Reliably Transmitting (NAT) to 192.168.5.4:5060:
            INVITE sip:667@192.168.5.4:5060 SIP/2.0
            Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK57ec8892;rport
            Max-Forwards: 70
            From: "Sistemas" <sip:510@192.168.5.1>;tag=as24312278
            To: <sip:667@192.168.5.4:5060>
            Contact: <sip:510@192.168.5.1:5060>
            Call-ID: 2fcf353638decd8a480c6caa19873145@192.168.5.1:5060
            CSeq: 102 INVITE
            User-Agent: FPBX-2.8.1(1.8.20.0)
            Date: Mon, 07 Jul 2014 16:32:54 GMT
            Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
            Supported: replaces, timer
            Content-Type: application/sdp
            Content-Length: 257

            v=0
            o=root 474195392 474195392 IN IP4 192.168.5.1
            s=Asterisk PBX 1.8.20.0
            c=IN IP4 192.168.5.1
            t=0 0
            m=audio 11406 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=ptime:20
            a=sendrecv

            ---
            -- Called SIP/667

            <--- SIP read from UDP:192.168.5.4:5060 --->
            SIP/2.0 100 Trying
            To: <sip:667@192.168.5.4:5060>
            From: "Sistemas" <sip:510@192.168.5.1>;tag=as24312278
            Call-ID: 2fcf353638decd8a480c6caa19873145@192.168.5.1:5060
            CSeq: 102 INVITE
            Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK57ec8892
            Server: Linksys/SPA922-5.2.8
            Content-Length: 0

            <------------->
            --- (8 headers 0 lines) ---

            <--- SIP read from UDP:192.168.5.4:5060 --->
            SIP/2.0 180 Ringing
            To: <sip:667@192.168.5.4:5060>;tag=ecbc57aa6e8fefcci 0
            From: "Sistemas" <sip:510@192.168.5.1>;tag=as24312278
            Call-ID: 2fcf353638decd8a480c6caa19873145@192.168.5.1:5060
            CSeq: 102 INVITE
            Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK57ec8892
            Server: Linksys/SPA922-5.2.8
            Content-Length: 0

            <------------->
            --- (8 headers 0 lines) ---
            list_route: no route
            -- SIP/667-00001168 is ringing
            -- <DAHDI/13-1> Playing 'vm-intro.slin' (language 'es')

            <--- SIP read from UDP:192.168.5.4:5060 --->
            SIP/2.0 200 OK
            To: <sip:667@192.168.5.4:5060>;tag=ecbc57aa6e8fefcci 0
            From: "Sistemas" <sip:510@192.168.5.1>;tag=as24312278
            Call-ID: 2fcf353638decd8a480c6caa19873145@192.168.5.1:5060
            CSeq: 102 INVITE
            Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK57ec8892
            Contact: "667" <sip:667@192.168.5.4:5060>
            Server: Linksys/SPA922-5.2.8
            Content-Length: 202
            Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
            Supported: replaces
            Content-Type: application/sdp

            v=0
            o=- 16623 16623 IN IP4 192.168.5.4
            s=-
            c=IN IP4 192.168.5.4
            t=0 0
            m=audio 16462 RTP/AVP 0 101
            a=rtpmap:0 PCMU/8000
            a=rtpmap:101 telephone-event/8000
            a=fmtp:101 0-15
            a=ptime:30
            a=sendrecv
            <------------->
            --- (12 headers 11 lines) ---
            Found RTP audio format 0
            Found RTP audio format 101
            Found audio description format PCMU for ID 0
            Found audio description format telephone-event for ID 101
            Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
            Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
            Peer audio RTP is at port 192.168.5.4:16462
            list_route: hop: <sip:667@192.168.5.4:5060>
            set_destination: Parsing <sip:667@192.168.5.4:5060> for address/port to send to
            set_destination: set destination to 192.168.5.4:5060
            Transmitting (NAT) to 192.168.5.4:5060:
            ACK sip:667@192.168.5.4:5060 SIP/2.0
            Via: SIP/2.0/UDP 192.168.5.1:5060;branch=z9hG4bK4db4df0a;rport
            Max-Forwards: 70
            From: "Sistemas" <sip:510@192.168.5.1>;tag=as24312278
            To: <sip:667@192.168.5.4:5060>;tag=ecbc57aa6e8fefcci 0
            Contact: <sip:510@192.168.5.1:5060>
            Call-ID: 2fcf353638decd8a480c6caa19873145@192.168.5.1:5060
            CSeq: 102 ACK
            User-Agent: FPBX-2.8.1(1.8.20.0)
            Content-Length: 0


            ---
            -- SIP/667-00001168 answered SIP/510-00001167

            <--- SIP read from UDP:192.168.5.4:5060 --->
            BYE sip:510@192.168.5.1:5060 SIP/2.0
            Via: SIP/2.0/UDP 192.168.5.4:5060;branch=z9hG4bK-c972bc01
            From: <sip:667@192.168.5.4>;tag=ecbc57aa6e8fefcci0
            To: "Sistemas" <sip:510@192.168.5.1>;tag=as24312278
            Call-ID: 2fcf353638decd8a480c6caa19873145@192.168.5.1:5060
            CSeq: 101 BYE
            Max-Forwards: 70
            User-Agent: Linksys/SPA922-5.2.8
            Content-Length: 0

            <------------->
            --- (9 headers 0 lines) ---
            Sending to 192.168.5.4:5060 (NAT)
            Scheduling destruction of SIP dialog '2fcf353638decd8a480c6caa19873145@192.168.5.1:5060 ' in 6400 ms (Method: BYE)

            <--- Transmitting (NAT) to 192.168.5.4:5060 --->
            SIP/2.0 200 OK
            Via: SIP/2.0/UDP 192.168.5.4:5060;branch=z9hG4bK-c972bc01;received=192.168.5.4;rport=5060
            From: <sip:667@192.168.5.4>;tag=ecbc57aa6e8fefcci0
            To: "Sistemas" <sip:510@192.168.5.1>;tag=as24312278
            Call-ID: 2fcf353638decd8a480c6caa19873145@192.168.5.1:5060
            CSeq: 101 BYE
            Server: FPBX-2.8.1(1.8.20.0)
            Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
            Supported: replaces, timer
            Content-Length: 0

            Comentario

            • #7
              Los Equipos de Cisco tienen un problema con asterisk que si los tienes en el mismo puerto, hacen cosas raras, si están en la misma subnet pon un puerto diferente a cada teléfono y listo, espero esto te sirva de ayuda, aprovecho para saludar a Christian, un fuerte abrazo desde Torreón.

              Comentario

              • #8
                Hola coloque un puerto diferente al 5060 en el telefono cisco y tambien tuve problemas de audio

                Comentario

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