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No se reproduce ningún sonido (hello-world)

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  • No se reproduce ningún sonido (hello-world)

    Hola a todos, desde ya les comento que estoy comenzando a aprender Asterisk, ando loco con un tema, cuando marco de un softphone a otro no genera ningún sonido, he realizado el típico ejemplo de Playback(hello-world) y VoiceMail y no se escucha nada, he investigado y me dicen que NAT, PRT, Codecs etc pero no encuentro solución.
    He instalado un AsteriskNOW sin FreePBX, en sip.conf tengo habilitados los codecs ulaw, alaw, gsm.
    Alguien me puede dar una ayuda, desde ya muchas gracias.

  • #2
    Activa el comando sip set debug on antes de que hagas tu llamada, y luego péganos el resultado de lo que aparece en tu consola cuando este error se presenta.
    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
      Cordial saludo Christian, muchas gracias por tu pronta respuesta, esto es lo que me envía después de activar sip set debug on al realizar la llamada. Muchas gracias por tu ayuda.

      <--- SIP read from UDP:190.14.228.30:8606 --->
      INVITE sip:102@168.61.37.87 SIP/2.0
      Via: SIP/2.0/UDP 10.254.246.30:5060;rport;branch=z9hG4bKPj13e463a7-fc64-42d7-b20e-01817071d1ba
      Max-Forwards: 70
      From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
      To: <sip:102@168.61.37.87>
      Contact: <sip:jgomez@190.14.228.30:8606>
      Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
      CSeq: 21054 INVITE
      Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE
      upported: replaces, 100rel
      Content-Type: application/sdp
      Content-Length: 363

      v=0
      o=LGX110 3600613138 0 IN IP4 10.254.246.30
      s=sflphone
      c=IN IP4 10.254.246.30
      t=0 0
      m=audio 13202 RTP/AVP 3 0 8 9 110 111 112
      a=rtpmap:3 GSM/8000
      a=rtpmap:0 PCMU/8000
      a=rtpmap:8 PCMA/8000
      a=rtpmap:9 G722/8000
      a=rtpmap:110 speex/8000
      a=rtpmap:111 speex/16000
      a=rtpmap:112 speex/32000
      a=sendrecv
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-15
      <------------->
      --- (12 headers 16 lines) ---
      Sending to 190.14.228.30:8606 (NAT)
      Using INVITE request as basis request - f6eb303b-224b-4f7a-8bc9-2d56348d28c6
      Found peer 'jgomez' for 'jgomez' from 190.14.228.30:8606

      <--- Reliably Transmitting (NAT) to 190.14.228.30:8606 --->
      SIP/2.0 401 Unauthorized
      Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj13e463a7-fc64-42d7-b20e-01817071d1ba;received=190.14.228.30;rport=8606
      From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
      To: <sip:102@168.61.37.87>;tag=as3ede4952
      Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
      CSeq: 21054 INVITE
      Server: Asterisk PBX 1.8.25.0
      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
      Supported: replaces, timer
      WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="10315bb8"
      Content-Length: 0


      <------------>
      Scheduling destruction of SIP dialog 'f6eb303b-224b-4f7a-8bc9-2d56348d28c6' in 7680 ms (Method: INVITE)

      <--- SIP read from UDP:190.14.228.30:8606 --->
      ACK sip:102@168.61.37.87 SIP/2.0
      Via: SIP/2.0/UDP 10.254.246.30:5060;rport;branch=z9hG4bKPj13e463a7-fc64-42d7-b20e-01817071d1ba
      Max-Forwards: 70
      From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
      To: <sip:102@168.61.37.87>;tag=as3ede4952
      Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
      CSeq: 21054 ACK
      Content-Length: 0

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

      <--- SIP read from UDP:190.14.228.30:8606 --->
      INVITE sip:102@168.61.37.87 SIP/2.0
      Via: SIP/2.0/UDP 10.254.246.30:5060;rport;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223
      Max-Forwards: 70
      From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
      To: <sip:102@168.61.37.87>
      Contact: <sip:jgomez@190.14.228.30:8606>
      Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
      CSeq: 21055 INVITE
      Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE
      upported: replaces, 100rel
      Authorization: Digest username="jgomez", realm="asterisk", nonce="10315bb8", uri="sip:102@168.61.37.87", response="6355e94d5ec1eabe381c268ca1afab53", algorithm=MD5
      Content-Type: application/sdp
      Content-Length: 363

      Continua...

      Comentario


      • #4
        v=0
        o=LGX110 3600613138 0 IN IP4 10.254.246.30
        s=sflphone
        c=IN IP4 10.254.246.30
        t=0 0
        m=audio 13202 RTP/AVP 3 0 8 9 110 111 112
        a=rtpmap:3 GSM/8000
        a=rtpmap:0 PCMU/8000
        a=rtpmap:8 PCMA/8000
        a=rtpmap:9 G722/8000
        a=rtpmap:110 speex/8000
        a=rtpmap:111 speex/16000
        a=rtpmap:112 speex/32000
        a=sendrecv
        a=rtpmap:101 telephone-event/8000
        a=fmtp:101 0-15
        <------------->
        --- (13 headers 16 lines) ---
        Sending to 190.14.228.30:8606 (NAT)
        Using INVITE request as basis request - f6eb303b-224b-4f7a-8bc9-2d56348d28c6
        Found peer 'jgomez' for 'jgomez' from 190.14.228.30:8606
        == Using SIP RTP CoS mark 5
        Found RTP audio format 3
        Found RTP audio format 0
        Found RTP audio format 8
        Found RTP audio format 9
        Found RTP audio format 110
        Found RTP audio format 111
        Found RTP audio format 112
        Found audio description format GSM for ID 3
        Found audio description format PCMU for ID 0
        Found audio description format PCMA for ID 8
        Found audio description format G722 for ID 9
        Found audio description format speex for ID 110
        Found audio description format speex for ID 111
        Found unknown media description format speex for ID 112
        Found audio description format telephone-event for ID 101
        Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x20000120e (gsm|ulaw|alaw|speex|speex16|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
        Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
        Peer audio RTP is at port 10.254.246.30:13202
        Looking for 102 in users (domain 168.61.37.87)
        list_route: hop: <sip:jgomez@190.14.228.30:8606>

        <--- Transmitting (NAT) to 190.14.228.30:8606 --->
        SIP/2.0 100 Trying
        Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223;received=190.14.228.30;rport=8606
        From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
        To: <sip:102@168.61.37.87>
        Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
        CSeq: 21055 INVITE
        Server: Asterisk PBX 1.8.25.0
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Contact: <sip:102@10.74.234.47:5060>
        Content-Length: 0


        <------------>
        -- Executing [102@users:1] Answer("SIP/jgomez-00000006", "") in new stack
        Audio is at 10908
        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 190.14.228.30:8606 --->
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223;received=190.14.228.30;rport=8606
        From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
        To: <sip:102@168.61.37.87>;tag=as30d2c9c8
        Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
        CSeq: 21055 INVITE
        Server: Asterisk PBX 1.8.25.0
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Contact: <sip:102@10.74.234.47:5060>
        Content-Type: application/sdp
        Content-Length: 284

        v=0
        o=root 1533468499 1533468499 IN IP4 10.74.234.47
        s=Asterisk PBX 1.8.25.0
        c=IN IP4 10.74.234.47
        t=0 0
        m=audio 10908 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=ptime:20
        a=sendrecv

        <------------>
        Retransmitting #1 (NAT) to 190.14.228.30:8606:
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223;received=190.14.228.30;rport=8606
        From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
        To: <sip:102@168.61.37.87>;tag=as30d2c9c8
        Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
        CSeq: 21055 INVITE
        Server: Asterisk PBX 1.8.25.0
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Contact: <sip:102@10.74.234.47:5060>
        Content-Type: application/sdp
        Content-Length: 284

        v=0
        o=root 1533468499 1533468499 IN IP4 10.74.234.47
        s=Asterisk PBX 1.8.25.0
        c=IN IP4 10.74.234.47
        t=0 0
        m=audio 10908 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=ptime:20
        a=sendrecv

        ---
        Retransmitting #2 (NAT) to 190.14.228.30:8606:
        SIP/2.0 200 OK
        Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223;received=190.14.228.30;rport=8606
        From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
        To: <sip:102@168.61.37.87>;tag=as30d2c9c8
        Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
        CSeq: 21055 INVITE
        Server: Asterisk PBX 1.8.25.0
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
        Supported: replaces, timer
        Contact: <sip:102@10.74.234.47:5060>
        Content-Type: application/sdp
        Content-Length: 284

        v=0
        o=root 1533468499 1533468499 IN IP4 10.74.234.47
        s=Asterisk PBX 1.8.25.0
        c=IN IP4 10.74.234.47
        t=0 0
        m=audio 10908 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=ptime:20
        a=sendrecv

        Comentario


        • #5
          ---
          -- Executing [102@users:2] Playback("SIP/jgomez-00000006", "hello-world") in new stack
          -- <SIP/jgomez-00000006> Playing 'hello-world.gsm' (language 'es')
          Retransmitting #3 (NAT) to 190.14.228.30:8606:
          SIP/2.0 200 OK
          Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223;received=190.14.228.30;rport=8606
          From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
          To: <sip:102@168.61.37.87>;tag=as30d2c9c8
          Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
          CSeq: 21055 INVITE
          Server: Asterisk PBX 1.8.25.0
          Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
          Supported: replaces, timer
          Contact: <sip:102@10.74.234.47:5060>
          Content-Type: application/sdp
          Content-Length: 284

          v=0
          o=root 1533468499 1533468499 IN IP4 10.74.234.47
          s=Asterisk PBX 1.8.25.0
          c=IN IP4 10.74.234.47
          t=0 0
          m=audio 10908 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=ptime:20
          a=sendrecv

          ---
          -- Executing [102@users:3] Hangup("SIP/jgomez-00000006", "") in new stack
          == Spawn extension (users, 102, 3) exited non-zero on 'SIP/jgomez-00000006'
          Scheduling destruction of SIP dialog 'f6eb303b-224b-4f7a-8bc9-2d56348d28c6' in 7680 ms (Method: INVITE)
          Retransmitting #4 (NAT) to 190.14.228.30:8606:
          SIP/2.0 200 OK
          Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223;received=190.14.228.30;rport=8606
          From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
          To: <sip:102@168.61.37.87>;tag=as30d2c9c8
          Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
          CSeq: 21055 INVITE
          Server: Asterisk PBX 1.8.25.0
          Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
          Supported: replaces, timer
          Contact: <sip:102@10.74.234.47:5060>
          Content-Type: application/sdp
          Content-Length: 284

          v=0
          o=root 1533468499 1533468499 IN IP4 10.74.234.47
          s=Asterisk PBX 1.8.25.0
          c=IN IP4 10.74.234.47
          t=0 0
          m=audio 10908 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=ptime:20
          a=sendrecv

          ---
          Retransmitting #5 (NAT) to 190.14.228.30:8606:
          SIP/2.0 200 OK
          Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223;received=190.14.228.30;rport=8606
          From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
          To: <sip:102@168.61.37.87>;tag=as30d2c9c8
          Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
          CSeq: 21055 INVITE
          Server: Asterisk PBX 1.8.25.0
          Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
          Supported: replaces, timer
          Contact: <sip:102@10.74.234.47:5060>
          Content-Type: application/sdp
          Content-Length: 284

          v=0
          o=root 1533468499 1533468499 IN IP4 10.74.234.47
          s=Asterisk PBX 1.8.25.0
          c=IN IP4 10.74.234.47
          t=0 0
          m=audio 10908 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=ptime:20
          a=sendrecv

          ---
          Really destroying SIP dialog 'c40fcb53-fabc-4d10-8440-a3ad56f08612' Method: REGISTER
          Retransmitting #6 (NAT) to 190.14.228.30:8606:
          SIP/2.0 200 OK
          Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj93b2b55d-0f58-4378-9a6a-0c34a8977223;received=190.14.228.30;rport=8606
          From: <sip:jgomez@168.61.37.87>;tag=65eaf3bb-a659-42cb-a6b8-aa96975df26b
          To: <sip:102@168.61.37.87>;tag=as30d2c9c8
          Call-ID: f6eb303b-224b-4f7a-8bc9-2d56348d28c6
          CSeq: 21055 INVITE
          Server: Asterisk PBX 1.8.25.0
          Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
          Supported: replaces, timer
          Contact: <sip:102@10.74.234.47:5060>
          Content-Type: application/sdp
          Content-Length: 284

          v=0
          o=root 1533468499 1533468499 IN IP4 10.74.234.47
          s=Asterisk PBX 1.8.25.0
          c=IN IP4 10.74.234.47
          t=0 0
          m=audio 10908 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=ptime:20
          a=sendrecv

          ---
          [Feb 5 18:17:36] WARNING[32155]: chan_sip.c:3983 retrans_pkt: Retransmission timeout reached on transmission f6eb303b-224b-4f7a-8bc9-2d56348d28c6 for seqno 21055 (Critical Response) -- See https://wiki.asterisk.org/wiki/displ...etransmissions
          Packet timed out after 7680ms with no response
          Really destroying SIP dialog 'f6eb303b-224b-4f7a-8bc9-2d56348d28c6' Method: INVITE

          <--- SIP read from UDP:190.14.228.30:8606 --->
          REGISTER sip:168.61.37.87 SIP/2.0
          Via: SIP/2.0/UDP 10.254.246.30:5060;rport;branch=z9hG4bKPj26835103-f13d-4c4c-a7b6-927f82b9edc5
          Max-Forwards: 70
          From: <sip:jgomez@168.61.37.87>;tag=e05afb17-c5c6-4a88-9c30-a16017684432
          To: <sip:jgomez@168.61.37.87>
          Call-ID: c40fcb53-fabc-4d10-8440-a3ad56f08612
          CSeq: 46411 REGISTER
          User-Agent: SFLphone
          Contact: <sip:jgomez@10.254.246.30:5060>
          Expires: 60
          Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE
          Content-Length: 0

          <------------->
          --- (12 headers 0 lines) ---
          Sending to 190.14.228.30:8606 (NAT)

          <--- Transmitting (NAT) to 190.14.228.30:8606 --->
          SIP/2.0 401 Unauthorized
          Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPj26835103-f13d-4c4c-a7b6-927f82b9edc5;received=190.14.228.30;rport=8606
          From: <sip:jgomez@168.61.37.87>;tag=e05afb17-c5c6-4a88-9c30-a16017684432
          To: <sip:jgomez@168.61.37.87>;tag=as7daa5eed
          Call-ID: c40fcb53-fabc-4d10-8440-a3ad56f08612
          CSeq: 46411 REGISTER
          Server: Asterisk PBX 1.8.25.0
          Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
          Supported: replaces, timer
          WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1f6c34dd"
          Content-Length: 0


          <------------>
          Scheduling destruction of SIP dialog 'c40fcb53-fabc-4d10-8440-a3ad56f08612' in 32000 ms (Method: REGISTER)

          <--- SIP read from UDP:190.14.228.30:8606 --->
          REGISTER sip:168.61.37.87 SIP/2.0
          Via: SIP/2.0/UDP 10.254.246.30:5060;rport;branch=z9hG4bKPja4fe68e8-e6cb-4b0a-ac17-7dffbc01a983
          Max-Forwards: 70
          From: <sip:jgomez@168.61.37.87>;tag=e05afb17-c5c6-4a88-9c30-a16017684432
          To: <sip:jgomez@168.61.37.87>
          Call-ID: c40fcb53-fabc-4d10-8440-a3ad56f08612
          CSeq: 46412 REGISTER
          User-Agent: SFLphone
          Contact: <sip:jgomez@10.254.246.30:5060>
          Expires: 60
          Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE
          Authorization: Digest username="jgomez", realm="asterisk", nonce="1f6c34dd", uri="sip:168.61.37.87", response="e95cac695e3f42c1c76ab840a9645b07", algorithm=MD5
          Content-Length: 0

          <------------->
          --- (13 headers 0 lines) ---
          Sending to 190.14.228.30:8606 (NAT)
          Reliably Transmitting (NAT) to 190.14.228.30:8606:
          OPTIONS sip:jgomez@10.254.246.30:5060 SIP/2.0
          Via: SIP/2.0/UDP 10.74.234.47:5060;branch=z9hG4bK367461a9;rport
          Max-Forwards: 70
          From: "asterisk" <sip:asterisk@10.74.234.47>;tag=as3d6b3703
          To: <sip:jgomez@10.254.246.30:5060>
          Contact: <sip:asterisk@10.74.234.47:5060>
          Call-ID: 39daa4b027d89787562b1ec138c49d02@10.74.234.47:5060
          CSeq: 102 OPTIONS
          User-Agent: Asterisk PBX 1.8.25.0
          Date: Wed, 05 Feb 2014 18:17:58 GMT
          Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
          Supported: replaces, timer
          Content-Length: 0


          ---

          <--- Transmitting (NAT) to 190.14.228.30:8606 --->
          SIP/2.0 200 OK
          Via: SIP/2.0/UDP 10.254.246.30:5060;branch=z9hG4bKPja4fe68e8-e6cb-4b0a-ac17-7dffbc01a983;received=190.14.228.30;rport=8606
          From: <sip:jgomez@168.61.37.87>;tag=e05afb17-c5c6-4a88-9c30-a16017684432
          To: <sip:jgomez@168.61.37.87>;tag=as7daa5eed
          Call-ID: c40fcb53-fabc-4d10-8440-a3ad56f08612
          CSeq: 46412 REGISTER
          Server: Asterisk PBX 1.8.25.0
          Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
          Supported: replaces, timer
          Expires: 60
          Contact: <sip:jgomez@10.254.246.30:5060>;expires=60
          Date: Wed, 05 Feb 2014 18:17:58 GMT
          Content-Length: 0


          <------------>
          Scheduling destruction of SIP dialog '020f770241f1c1803db96d9a4fe44b1d@10.74.234.47:506 0' in 7680 ms (Method: NOTIFY)
          Reliably Transmitting (NAT) to 190.14.228.30:8606:
          NOTIFY sip:jgomez@10.254.246.30:5060 SIP/2.0
          Via: SIP/2.0/UDP 10.74.234.47:5060;branch=z9hG4bK36a4396d;rport
          Max-Forwards: 70
          From: "asterisk" <sip:asterisk@10.74.234.47>;tag=as704a21cf
          To: <sip:jgomez@10.254.246.30:5060>
          Contact: <sip:asterisk@10.74.234.47:5060>
          Call-ID: 020f770241f1c1803db96d9a4fe44b1d@10.74.234.47:5060
          CSeq: 102 NOTIFY
          User-Agent: Asterisk PBX 1.8.25.0
          Event: message-summary
          Content-Type: application/simple-message-summary
          Content-Length: 92

          Messages-Waiting: no
          Message-Account: sip:asterisk@10.74.234.47
          Voice-Message: 0/0 (0/0)

          ---
          Scheduling destruction of SIP dialog 'c40fcb53-fabc-4d10-8440-a3ad56f08612' in 32000 ms (Method: REGISTER)

          <--- SIP read from UDP:190.14.228.30:8606 --->
          SIP/2.0 200 OK
          Via: SIP/2.0/UDP 10.74.234.47:5060;rport=5060;received=168.61.37.87 ;branch=z9hG4bK367461a9
          Call-ID: 39daa4b027d89787562b1ec138c49d02@10.74.234.47:5060
          From: "asterisk" <sip:asterisk@10.74.234.47>;tag=as3d6b3703
          To: <sip:jgomez@10.254.246.30:5060>;tag=z9hG4bK367461a 9
          CSeq: 102 OPTIONS
          Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE
          Accept: message/sipfrag;version=2.0, application/sdp, text/plain, application/sdp
          Supported: replaces, 100rel
          Allow-Events: refer
          Content-Length: 0

          <------------->
          --- (11 headers 0 lines) ---
          Really destroying SIP dialog '39daa4b027d89787562b1ec138c49d02@10.74.234.47:506 0' Method: OPTIONS

          <--- SIP read from UDP:190.14.228.30:8606 --->
          SIP/2.0 200 OK
          Via: SIP/2.0/UDP 10.74.234.47:5060;rport=5060;received=168.61.37.87 ;branch=z9hG4bK36a4396d
          Call-ID: 020f770241f1c1803db96d9a4fe44b1d@10.74.234.47:5060
          From: "asterisk" <sip:asterisk@10.74.234.47>;tag=as704a21cf
          To: <sip:jgomez@10.254.246.30:5060>;tag=z9hG4bK36a4396 d
          CSeq: 102 NOTIFY
          Content-Length: 0

          <------------->
          --- (7 headers 0 lines) ---
          Really destroying SIP dialog '020f770241f1c1803db96d9a4fe44b1d@10.74.234.47:506 0' Method: NOTIFY

          Comentario


          • #6
            La linea
            Código:
            Peer audio RTP is at port 10.254.246.30:13202
            te dice que es un problema de RTP que se está enviando a la dirección/puerto incorrecto, ya que tu usuario está en una IP pública diferente y el RTP está tratando de enviarse a una local.

            ¿Cuál es la red local donde Asterisk está?
            ¿Hacia que IP estás registrando el teléfono del usuario (normalmente es lo que pones en el campo de domain del softphone)?
            dCAP Christian Cabrera R.
            Para aprender a usar Asterisk, asiste a uno de mis cursos Asterisk
            Si deseas asesoría pagada, por favor contáctame

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