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Error softphone en android al realizar una llamada

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  • #1

    Error softphone en android al realizar una llamada

    Hola!

    Me encuentro con este problema a ver si me podrian apoyar.

    Lo que pasa es que quiero instalar un softphone en mi telefono movil y descargue el Zoiper para android lo configure y se logro registrar en el asterisk con protocolo SIP. Lo que pasa es que cuando realizo una llamada a un telefono cisco, si suena y todo pero al contestar cuando hablo del telefono cisco se escucha en la bocina de mi telefono movil pero del movil al telefono no se escucha, se escucha puro ruido en la bocina y hace que cuelgue y termine la llamada. A y tambien cuando quiero llamar del cisco no entra la llamada en mi celular.

    Anexo lo que muestra en la consola de asterisk:

    == Using SIP RTP CoS mark 5
    -- Executing [201@interno:1] Answer("SIP/204-00000042", "") in new stack
    -- Executing [201@interno:2] Dial("SIP/204-00000042", "Sip/201,25,Tt") in new stack
    == Using SIP RTP CoS mark 5
    -- Called 201
    -- SIP/201-00000043 is ringing
    -- SIP/201-00000043 answered SIP/204-00000042
    [Jan 16 17:44:51] WARNING[15591]: chan_sip.c:4051 retrans_pkt: Maximum retries exceeded on transmission YTIyNjY4MmUxYjU1MTNjOTAzZjFkMjY5OGM2ZmYwYTg. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/displ...etransmissions
    [Jan 16 17:44:51] WARNING[15591]: chan_sip.c:4078 retrans_pkt: Hanging up call YTIyNjY4MmUxYjU1MTNjOTAzZjFkMjY5OGM2ZmYwYTg. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/displ...etransmissions ).
    == Spawn extension (interno, 201, 2) exited non-zero on 'SIP/204-00000042'


    Asi tengo mi cuentas SIP:


    [201] ; teléfono cisco 7911
    context=interno
    callerid="201" <201>
    secret=201
    mailbox=201@default
    type=friend
    host=dynamic
    canreinvite=no
    call-limit=4
    callgroup=1
    pickupgroup=1
    disallow=all
    allow=gsm
    allow=g729
    allow=ulaw
    allow=alaw
    nat=no
    dtfmode=rfc2833

    [204] ;telefono celular con android
    context=interno
    callerid="204" <204>
    secret=204
    mailbox=204@default
    type=friend
    host=dynamic
    canreinvite=no
    call-limit=4
    callgroup=1
    pickupgroup=1
    disallow=all
    allow=gsm
    allow=ulaw
    allow=alaw
    nat=yes

    Y este es mi dialplan:

    [interno]
    exten => 201,1,Answer()
    exten => 201,n,Dial(Sip/201,25,Tt)
    exten => 201,n,Voicemail(201@default)
    exten => 201,n,Hangup()

    exten => 204,1,Answer()
    exten => 204,n,Dial(Sip/204,25,Tt)
    exten => 204,n,Voicemail(204@default)
    exten => 204,n,Hangup()

    Que podrá ser lo que pueda estar haciendo conflicto?

    Gracias, espero respuestas!
    Etiquetas: Ninguno
  • #2
    ¿Podrias generar una llamada, descolgarla, y mientras escuchas el ruido ejecutar sip show channels para ver como se establece la llamada entre los teléfonos?
    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
      Este error es muy común cuando no tienes configurado correctamente los settings de nat:

      Jan 16 17:44:51] WARNING[15591]: chan_sip.c:4051 retrans_pkt: Maximum retries exceeded on transmission YTIyNjY4MmUxYjU1MTNjOTAzZjFkMjY5OGM2ZmYwYTg. for seqno 2 (Critical Response) -- See
      Checa que tengas configurado externhost o externip, localnet etc.

      Comentario

      • #4
        Christian, ya ejecute el sip show channels y este fue el resultado:

        Peer User/ANR Call ID Format Hold Last Message Expiry
        10.1.2.7 (None) NDZjYzg0YTIxODc 0x0 (nothing) No Rx: REGISTER
        10.1.2.101 (None) 0024c445-815300 0x0 (nothing) No Rx: REGISTER
        10.1.2.7 204 NzJiZmY3NWI2OWN 0x8 (alaw) No Rx: INVITE
        10.1.2.102 (None) 0024c444-ad7d00 0x0 (nothing) No Rx: REGISTER
        10.1.2.117 201 129624ca26f6215 0x4 (ulaw) No Tx: ACK

        Navaismo que valores debe de llevar los parametros externhost o externip, localnet?

        Comentario

        • #5
          Los datos de tu red, externip es tu IP pública, si es dinámica entonces debes usar externhost tu dominio publico que apunta al pbx y localnet debe ser tu red local por ejemplo: 192.168.1.0/255.255.255.0

          Comentario

          • #6
            ok navaismo... ya se los puse pero no cambio en nada. :/

            Aprovechando tengo una pregunta.. como le puedo hacer para que cuando yo marque una extensión salga el nombre de dicha extensión, ya que cuando marco por el ejemplo 201 en la pantalla nada mas dice 201 y quiero que aparezca el nombre del usuario. Se te cambiar la configuracion en el asterisk o en el teléfono? Tengo teléfonos cisco 7911.

            Comentario

            • #7
              Si estas en un ambiente externo a tu LAN tambien tienes que checar los puertos de tu firewall en tu modem/router e incluso en tu PBX. Támbien sería bueno que pegaras la salida del SIP debug completo de la llamada faliida.

              Comentario

              • #8
                Ejecute el rtp y este fue el resultado

                Sent RTP packet to 10.1.2.117:25098 (type 00, seq 035183, ts 958402496, len 000160)
                Got RTP packet from 10.1.2.117:25098 (type 00, seq 035540, ts 3763232469, len 000160)
                Sent RTP packet to 10.1.2.7:33000 (type 00, seq 014244, ts 3763232464, len 000160)
                Got RTP packet from 10.1.2.117:25098 (type 00, seq 035541, ts 3763232629, len 000160)
                Sent RTP packet to 10.1.2.7:33000 (type 00, seq 014245, ts 3763232624, len 000160)
                Got RTP packet from 10.1.2.117:25098 (type 00, seq 035542, ts 3763232789, len 000160)
                Sent RTP packet to 10.1.2.7:33000 (type 00, seq 014246, ts 3763232784, len 000160)
                Got RTP packet from 10.1.2.117:25098 (type 00, seq 035543, ts 3763232949, len 000160)
                Sent RTP packet to 10.1.2.7:33000 (type 00, seq 014247, ts 3763232944, len 000160)
                Got RTP packet from 10.1.2.117:25098 (type 00, seq 035544, ts 3763233109, len 000160)
                Sent RTP packet to 10.1.2.7:33000 (type 00, seq 014248, ts 3763233104, len 000160)
                Got RTP packet from 10.1.2.117:25098 (type 00, seq 035545, ts 3763233269, len 000160)
                Sent RTP packet to 10.1.2.7:33000 (type 00, seq 014249, ts 3763233264, len 000160)
                Got RTP packet from 10.1.2.117:25098 (type 00, seq 035546, ts 3763233429, len 000160)
                Sent RTP packet to 10.1.2.7:33000 (type 00, seq 014250, ts 3763233424, len 000160)
                Got RTP packet from 10.1.2.117:25098 (type 00, seq 035547, ts 3763233589, len 000160)
                Sent RTP packet to 10.1.2.7:33000 (type 00, seq 014251, ts 3763233584, len 000160)
                == Spawn extension (interno, 201, 2) exited non-zero on 'SIP/204-0000008f'

                y esto es cuando ejecute el sip debug:

                ---
                -- SIP/201-00000094 answered SIP/204-00000093
                Retransmitting #4 (no NAT) to 10.1.2.7:5060:
                SIP/2.0 200 OK
                Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-c4740348d1a40479-1---d8754z-;received=10.1.2.7
                From: <sip:204@10.1.2.37;transport=UDP>;tag=d084c77b
                To: <sip:201@10.1.2.37;transport=UDP>;tag=as44fff995
                Call-ID: OGIxZDcwY2ExMmY3NmZmM2Y4MmM5NjUyYWJlMzNmYmU.
                CSeq: 2 INVITE
                Server: Asterisk PBX 1.6.2.19
                Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                Supported: replaces, timer
                Contact: <sip:201@10.1.2.37>
                Content-Type: application/sdp
                Content-Length: 199

                v=0
                o=root 1930319287 1930319287 IN IP4 10.1.2.37
                s=Asterisk PBX 1.6.2.19
                c=IN IP4 10.1.2.37
                t=0 0
                m=audio 19448 RTP/AVP 0 8
                a=rtpmap:0 PCMU/8000
                a=rtpmap:8 PCMA/8000
                a=ptime:20
                a=sendrecv

                ---
                Retransmitting #5 (no NAT) to 10.1.2.7:5060:
                SIP/2.0 200 OK
                Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-c4740348d1a40479-1---d8754z-;received=10.1.2.7
                From: <sip:204@10.1.2.37;transport=UDP>;tag=d084c77b
                To: <sip:201@10.1.2.37;transport=UDP>;tag=as44fff995
                Call-ID: OGIxZDcwY2ExMmY3NmZmM2Y4MmM5NjUyYWJlMzNmYmU.
                CSeq: 2 INVITE
                Server: Asterisk PBX 1.6.2.19
                Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                Supported: replaces, timer
                Contact: <sip:201@10.1.2.37>
                Content-Type: application/sdp
                Content-Length: 199

                v=0
                o=root 1930319287 1930319287 IN IP4 10.1.2.37
                s=Asterisk PBX 1.6.2.19
                c=IN IP4 10.1.2.37
                t=0 0
                m=audio 19448 RTP/AVP 0 8
                a=rtpmap:0 PCMU/8000
                a=rtpmap:8 PCMA/8000
                a=ptime:20
                a=sendrecv

                ---
                Retransmitting #6 (no NAT) to 10.1.2.7:5060:
                SIP/2.0 200 OK
                Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-c4740348d1a40479-1---d8754z-;received=10.1.2.7
                From: <sip:204@10.1.2.37;transport=UDP>;tag=d084c77b
                To: <sip:201@10.1.2.37;transport=UDP>;tag=as44fff995
                Call-ID: OGIxZDcwY2ExMmY3NmZmM2Y4MmM5NjUyYWJlMzNmYmU.
                CSeq: 2 INVITE
                Server: Asterisk PBX 1.6.2.19
                Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                Supported: replaces, timer
                Contact: <sip:201@10.1.2.37>
                Content-Type: application/sdp
                Content-Length: 199

                v=0
                o=root 1930319287 1930319287 IN IP4 10.1.2.37
                s=Asterisk PBX 1.6.2.19
                c=IN IP4 10.1.2.37
                t=0 0
                m=audio 19448 RTP/AVP 0 8
                a=rtpmap:0 PCMU/8000
                a=rtpmap:8 PCMA/8000
                a=ptime:20
                a=sendrecv

                ---
                [Jan 17 13:27:47] WARNING[15591]: chan_sip.c:4051 retrans_pkt: Maximum retries exceeded on transmission OGIxZDcwY2ExMmY3NmZmM2Y4MmM5NjUyYWJlMzNmYmU. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/displ...etransmissions
                [Jan 17 13:27:47] WARNING[15591]: chan_sip.c:4078 retrans_pkt: Hanging up call OGIxZDcwY2ExMmY3NmZmM2Y4MmM5NjUyYWJlMzNmYmU. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/displ...etransmissions ).
                Scheduling destruction of SIP dialog '5b58928a5e4b16034bb3ee9b38330657@10.1.2.37' in 32000 ms (Method: INVITE)
                set_destination: Parsing <sip:201@10.1.2.117:5060;transport=udp> for address/port to send to
                set_destination: set destination to 10.1.2.117, port 5060
                Reliably Transmitting (no NAT) to 10.1.2.117:5060:
                BYE sip:201@10.1.2.117:5060;transport=udp SIP/2.0
                Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK229cae3f;rport
                Max-Forwards: 70
                From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as6d31e3b1
                To: <sip:201@10.1.2.117:5060;transport=udp>;tag=0024c4 44a2fa0456879b915a-46ce895b
                Call-ID: 5b58928a5e4b16034bb3ee9b38330657@10.1.2.37
                CSeq: 103 BYE
                User-Agent: Asterisk PBX 1.6.2.19
                X-Asterisk-HangupCause: Normal Clearing
                X-Asterisk-HangupCauseCode: 16
                Content-Length: 0


                ---
                == Spawn extension (interno, 201, 2) exited non-zero on 'SIP/204-00000093'

                <--- SIP read from UDP:10.1.2.117:51723 --->
                SIP/2.0 200 OK
                Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK229cae3f;rport
                From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as6d31e3b1
                To: <sip:201@10.1.2.117:5060;transport=udp>;tag=0024c4 44a2fa0456879b915a-46ce895b
                Call-ID: 5b58928a5e4b16034bb3ee9b38330657@10.1.2.37
                Date: Thu, 17 Jan 2013 19:27:48 GMT
                CSeq: 103 BYE
                Server: Cisco-CP7911G/8.3.0
                Content-Length: 0


                <------------->
                --- (9 headers 0 lines) ---
                Really destroying SIP dialog '5b58928a5e4b16034bb3ee9b38330657@10.1.2.37' Method: INVITE
                Really destroying SIP dialog 'OGIxZDcwY2ExMmY3NmZmM2Y4MmM5NjUyYWJlMzNmYmU.' Method: INVITE
                Really destroying SIP dialog 'ODk1MTQzYzY3ZGY5ODhkNTI3YzFmOGQ4MGVkMTEzOTk.' Method: REGISTER

                Comentario

                • #9
                  Los mensajes que ves que empiezan con

                  Retransmitting #
                  Son las veces que el asterisk ha intentado hacer contacto con tu peer, y como no recibe respuesta asterisk cuelga la llamada. Para poder un poco mas a fondo se necesita el sip debug completo de la llamada.

                  Intenta también estableciendo: pedantic=no en el sip.conf ya que android tiene algunos detalles.

                  Comentario

                  • #10
                    Ya le agregue pedantic=no y sigue igual, aqui esta el sip debug completo:

                    <--- SIP read from UDP:10.1.2.7:5060 --->
                    INVITE sip:201@10.1.2.37;transport=UDP SIP/2.0
                    Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-f144f5b432752fa3-1---d8754z-
                    Max-Forwards: 70
                    Contact: <sip:204@10.1.2.7:5060;transport=UDP>
                    To: <sip:201@10.1.2.37;transport=UDP>
                    From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
                    Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
                    CSeq: 1 INVITE
                    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
                    Content-Type: application/sdp
                    Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
                    User-Agent: Zoiper rev.16688
                    Allow-Events: presence, kpml
                    Content-Length: 251

                    v=0
                    o=Zoiper 0 0 IN IP4 10.1.2.7
                    s=Zoiper
                    c=IN IP4 10.1.2.7
                    t=0 0
                    m=audio 33000 RTP/AVP 0 110 3 8 98
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:110 speex/8000
                    a=rtpmap:3 GSM/8000
                    a=rtpmap:8 PCMA/8000
                    a=rtpmap:98 iLBC/8000
                    a=fmtp:98 mode=30
                    a=sendrecv

                    <------------->
                    --- (14 headers 13 lines) ---
                    == Using SIP RTP CoS mark 5
                    Sending to 10.1.2.7 : 5060 (no NAT)
                    Using INVITE request as basis request - NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
                    Found peer '204' for '204' from 10.1.2.7:5060

                    <--- Reliably Transmitting (no NAT) to 10.1.2.7:5060 --->
                    SIP/2.0 401 Unauthorized
                    Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-f144f5b432752fa3-1---d8754z-;received=10.1.2.7
                    From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
                    To: <sip:201@10.1.2.37;transport=UDP>;tag=as1e901650
                    Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
                    CSeq: 1 INVITE
                    Server: Asterisk PBX 1.6.2.19
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                    Supported: replaces, timer
                    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3e91df2d"
                    Content-Length: 0


                    <------------>
                    Scheduling destruction of SIP dialog 'NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.' in 32000 ms (Method: INVITE)

                    <--- SIP read from UDP:10.1.2.7:5060 --->
                    ACK sip:201@10.1.2.37;transport=UDP SIP/2.0
                    Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-f144f5b432752fa3-1---d8754z-
                    Max-Forwards: 70
                    To: <sip:201@10.1.2.37;transport=UDP>;tag=as1e901650
                    From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
                    Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
                    CSeq: 1 ACK
                    Content-Length: 0


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

                    <--- SIP read from UDP:10.1.2.7:5060 --->
                    INVITE sip:201@10.1.2.37;transport=UDP SIP/2.0
                    Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-
                    Max-Forwards: 70
                    Contact: <sip:204@10.1.2.7:5060;transport=UDP>
                    To: <sip:201@10.1.2.37;transport=UDP>
                    From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
                    Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
                    CSeq: 2 INVITE
                    Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
                    Content-Type: application/sdp
                    Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
                    User-Agent: Zoiper rev.16688
                    Authorization: Digest username="204",realm="asterisk",nonce="3e91df2d",u ri="sip:201@10.1.2.37;transport=UDP",response="5b7 c0ea6230464ad2bc6d194e1729f16",algorithm=MD5
                    Allow-Events: presence, kpml
                    Content-Length: 251

                    v=0
                    o=Zoiper 0 0 IN IP4 10.1.2.7
                    s=Zoiper
                    c=IN IP4 10.1.2.7
                    t=0 0
                    m=audio 33000 RTP/AVP 0 110 3 8 98
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:110 speex/8000
                    a=rtpmap:3 GSM/8000
                    a=rtpmap:8 PCMA/8000
                    a=rtpmap:98 iLBC/8000
                    a=fmtp:98 mode=30
                    a=sendrecv

                    <------------->
                    --- (15 headers 13 lines) ---
                    Sending to 10.1.2.7 : 5060 (no NAT)
                    Using INVITE request as basis request - NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
                    Found peer '204' for '204' from 10.1.2.7:5060
                    Found RTP audio format 0
                    Found RTP audio format 110
                    Found RTP audio format 3
                    Found RTP audio format 8
                    Found RTP audio format 98
                    Found audio description format PCMU for ID 0
                    Found audio description format speex for ID 110
                    Found audio description format GSM for ID 3
                    Found audio description format PCMA for ID 8
                    Found audio description format iLBC for ID 98
                    Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
                    Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
                    Peer audio RTP is at port 10.1.2.7:33000
                    Looking for 201 in interno (domain 10.1.2.37)
                    list_route: hop: <sip:204@10.1.2.7:5060;transport=UDP>

                    <--- Transmitting (no NAT) to 10.1.2.7:5060 --->
                    SIP/2.0 100 Trying
                    Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-;received=10.1.2.7
                    From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
                    To: <sip:201@10.1.2.37;transport=UDP>
                    Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
                    CSeq: 2 INVITE
                    Server: Asterisk PBX 1.6.2.19
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                    Supported: replaces, timer
                    Contact: <sip:201@10.1.2.37>
                    Content-Length: 0


                    <------------>
                    -- Executing [201@interno:1] Answer("SIP/204-000000a6", "") in new stack
                    Audio is at 10.1.2.37 port 14344
                    Adding codec 0x4 (ulaw) to SDP
                    Adding codec 0x8 (alaw) to SDP

                    <--- Reliably Transmitting (no NAT) to 10.1.2.7:5060 --->
                    SIP/2.0 200 OK
                    Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-;received=10.1.2.7
                    From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
                    To: <sip:201@10.1.2.37;transport=UDP>;tag=as0eb08b33
                    Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
                    CSeq: 2 INVITE
                    Server: Asterisk PBX 1.6.2.19
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                    Supported: replaces, timer
                    Contact: <sip:201@10.1.2.37>
                    Content-Type: application/sdp
                    Content-Length: 199

                    v=0
                    o=root 1256691965 1256691965 IN IP4 10.1.2.37
                    s=Asterisk PBX 1.6.2.19
                    c=IN IP4 10.1.2.37
                    t=0 0
                    m=audio 14344 RTP/AVP 0 8
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:8 PCMA/8000
                    a=ptime:20
                    a=sendrecv

                    <------------>
                    -- Executing [201@interno:2] Dial("SIP/204-000000a6", "Sip/201,20,Tt") in new stack
                    == Using SIP RTP CoS mark 5
                    Audio is at 10.1.2.37 port 18656
                    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 (no NAT) to 10.1.2.117:5060:
                    INVITE sip:201@10.1.2.117:5060;transport=udp SIP/2.0
                    Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK71ab2aee;rport
                    Max-Forwards: 70
                    From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as65415c05
                    To: <sip:201@10.1.2.117:5060;transport=udp>
                    Contact: <sip:204@10.1.2.37>
                    Call-ID: 6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37
                    CSeq: 102 INVITE
                    User-Agent: Asterisk PBX 1.6.2.19
                    Date: Thu, 17 Jan 2013 23:27:21 GMT
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                    Supported: replaces, timer
                    Content-Type: application/sdp
                    Content-Length: 278

                    v=0
                    o=root 1858409230 1858409230 IN IP4 10.1.2.37
                    s=Asterisk PBX 1.6.2.19
                    c=IN IP4 10.1.2.37
                    t=0 0
                    m=audio 18656 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

                    ---
                    -- Called 201

                    <--- SIP read from UDP:10.1.2.117:51919 --->
                    SIP/2.0 100 Trying
                    Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK71ab2aee;rport
                    From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as65415c05
                    To: <sip:201@10.1.2.117:5060;transport=udp>
                    Call-ID: 6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37
                    Date: Thu, 17 Jan 2013 23:27:20 GMT
                    CSeq: 102 INVITE
                    Server: Cisco-CP7911G/8.3.0
                    Contact: <sip:201@10.1.2.117:5060;transport=udp>
                    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTE R,UPDATE,SUBSCRIBE
                    Allow-Events: dialog
                    Content-Length: 0


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

                    <--- SIP read from UDP:10.1.2.117:51920 --->
                    SIP/2.0 180 Ringing
                    Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK71ab2aee;rport
                    From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as65415c05
                    To: <sip:201@10.1.2.117:5060;transport=udp>;tag=0024c4 44a2fa04b09933761d-88196ad5
                    Call-ID: 6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37
                    Date: Thu, 17 Jan 2013 23:27:21 GMT
                    CSeq: 102 INVITE
                    Server: Cisco-CP7911G/8.3.0
                    Contact: <sip:201@10.1.2.117:5060;transport=udp>
                    Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTE R,UPDATE,SUBSCRIBE
                    Allow-Events: dialog
                    Content-Length: 0


                    <------------->
                    --- (12 headers 0 lines) ---
                    -- SIP/201-000000a7 is ringing
                    Retransmitting #1 (no NAT) to 10.1.2.7:5060:
                    SIP/2.0 200 OK
                    Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-;received=10.1.2.7
                    From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
                    To: <sip:201@10.1.2.37;transport=UDP>;tag=as0eb08b33
                    Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
                    CSeq: 2 INVITE
                    Server: Asterisk PBX 1.6.2.19
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                    Supported: replaces, timer
                    Contact: <sip:201@10.1.2.37>
                    Content-Type: application/sdp
                    Content-Length: 199

                    v=0
                    o=root 1256691965 1256691965 IN IP4 10.1.2.37
                    s=Asterisk PBX 1.6.2.19
                    c=IN IP4 10.1.2.37
                    t=0 0
                    m=audio 14344 RTP/AVP 0 8
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:8 PCMA/8000
                    a=ptime:20
                    a=sendrecv

                    ---
                    Retransmitting #2 (no NAT) to 10.1.2.7:5060:
                    SIP/2.0 200 OK
                    Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-;received=10.1.2.7
                    From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
                    To: <sip:201@10.1.2.37;transport=UDP>;tag=as0eb08b33
                    Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
                    CSeq: 2 INVITE
                    Server: Asterisk PBX 1.6.2.19
                    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                    Supported: replaces, timer
                    Contact: <sip:201@10.1.2.37>
                    Content-Type: application/sdp
                    Content-Length: 199

                    v=0
                    o=root 1256691965 1256691965 IN IP4 10.1.2.37
                    s=Asterisk PBX 1.6.2.19
                    c=IN IP4 10.1.2.37
                    t=0 0
                    m=audio 14344 RTP/AVP 0 8
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:8 PCMA/8000
                    a=ptime:20
                    a=sendrecv

                    ---

                    Comentario

                    • #11
                      <--- SIP read from UDP:10.1.2.117:51921 --->
                      SIP/2.0 200 OK
                      Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK71ab2aee;rport
                      From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as65415c05
                      To: <sip:201@10.1.2.117:5060;transport=udp>;tag=0024c4 44a2fa04b09933761d-88196ad5
                      Call-ID: 6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37
                      Date: Thu, 17 Jan 2013 23:27:22 GMT
                      CSeq: 102 INVITE
                      Server: Cisco-CP7911G/8.3.0
                      Contact: <sip:201@10.1.2.117:5060;transport=udp>
                      Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTE R,UPDATE,SUBSCRIBE
                      Supported: replaces,join,norefersub
                      Allow-Events: dialog
                      Content-Length: 201
                      Content-Type: application/sdp
                      Content-Disposition: session;handling=optional

                      v=0
                      o=Cisco-SIPUA 16520 0 IN IP4 10.1.2.117
                      s=SIP Call
                      t=0 0
                      m=audio 27448 RTP/AVP 0 101
                      c=IN IP4 10.1.2.117
                      a=rtpmap:0 PCMU/8000
                      a=rtpmap:101 telephone-event/8000
                      a=fmtp:101 0-15
                      a=sendrecv

                      <------------->
                      --- (15 headers 10 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 - 0x10e (gsm|ulaw|alaw|g729), 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 10.1.2.117:27448
                      list_route: hop: <sip:201@10.1.2.117:5060;transport=udp>
                      set_destination: Parsing <sip:201@10.1.2.117:5060;transport=udp> for address/port to send to
                      set_destination: set destination to 10.1.2.117, port 5060
                      Transmitting (no NAT) to 10.1.2.117:5060:
                      ACK sip:201@10.1.2.117:5060;transport=udp SIP/2.0
                      Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK5cc4cb74;rport
                      Max-Forwards: 70
                      From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as65415c05
                      To: <sip:201@10.1.2.117:5060;transport=udp>;tag=0024c4 44a2fa04b09933761d-88196ad5
                      Contact: <sip:204@10.1.2.37>
                      Call-ID: 6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37
                      CSeq: 102 ACK
                      User-Agent: Asterisk PBX 1.6.2.19
                      Content-Length: 0


                      ---
                      -- SIP/201-000000a7 answered SIP/204-000000a6
                      Retransmitting #3 (no NAT) to 10.1.2.7:5060:
                      SIP/2.0 200 OK
                      Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-;received=10.1.2.7
                      From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
                      To: <sip:201@10.1.2.37;transport=UDP>;tag=as0eb08b33
                      Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
                      CSeq: 2 INVITE
                      Server: Asterisk PBX 1.6.2.19
                      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                      Supported: replaces, timer
                      Contact: <sip:201@10.1.2.37>
                      Content-Type: application/sdp
                      Content-Length: 199

                      v=0
                      o=root 1256691965 1256691965 IN IP4 10.1.2.37
                      s=Asterisk PBX 1.6.2.19
                      c=IN IP4 10.1.2.37
                      t=0 0
                      m=audio 14344 RTP/AVP 0 8
                      a=rtpmap:0 PCMU/8000
                      a=rtpmap:8 PCMA/8000
                      a=ptime:20
                      a=sendrecv

                      ---
                      Retransmitting #4 (no NAT) to 10.1.2.7:5060:
                      SIP/2.0 200 OK
                      Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-;received=10.1.2.7
                      From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
                      To: <sip:201@10.1.2.37;transport=UDP>;tag=as0eb08b33
                      Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
                      CSeq: 2 INVITE
                      Server: Asterisk PBX 1.6.2.19
                      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                      Supported: replaces, timer
                      Contact: <sip:201@10.1.2.37>
                      Content-Type: application/sdp
                      Content-Length: 199

                      v=0
                      o=root 1256691965 1256691965 IN IP4 10.1.2.37
                      s=Asterisk PBX 1.6.2.19
                      c=IN IP4 10.1.2.37
                      t=0 0
                      m=audio 14344 RTP/AVP 0 8
                      a=rtpmap:0 PCMU/8000
                      a=rtpmap:8 PCMA/8000
                      a=ptime:20
                      a=sendrecv

                      ---
                      Retransmitting #5 (no NAT) to 10.1.2.7:5060:
                      SIP/2.0 200 OK
                      Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-;received=10.1.2.7
                      From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
                      To: <sip:201@10.1.2.37;transport=UDP>;tag=as0eb08b33
                      Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
                      CSeq: 2 INVITE
                      Server: Asterisk PBX 1.6.2.19
                      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                      Supported: replaces, timer
                      Contact: <sip:201@10.1.2.37>
                      Content-Type: application/sdp
                      Content-Length: 199

                      v=0
                      o=root 1256691965 1256691965 IN IP4 10.1.2.37
                      s=Asterisk PBX 1.6.2.19
                      c=IN IP4 10.1.2.37
                      t=0 0
                      m=audio 14344 RTP/AVP 0 8
                      a=rtpmap:0 PCMU/8000
                      a=rtpmap:8 PCMA/8000
                      a=ptime:20
                      a=sendrecv

                      ---
                      Really destroying SIP dialog 'MDU2YTcxNmNmZWRmNjhjOWEwNzdiYTRkYzg0MmIyOTU.' Method: REGISTER
                      Really destroying SIP dialog 'MjFhZDlkZmNmZmFmYTg0ZDE0NjdjZGM4MGFhYTE5Y2M.' Method: REGISTER
                      Retransmitting #6 (no NAT) to 10.1.2.7:5060:
                      SIP/2.0 200 OK
                      Via: SIP/2.0/UDP 10.1.2.7:5060;branch=z9hG4bK-d8754z-2bf536b143c76348-1---d8754z-;received=10.1.2.7
                      From: <sip:204@10.1.2.37;transport=UDP>;tag=d1fae21b
                      To: <sip:201@10.1.2.37;transport=UDP>;tag=as0eb08b33
                      Call-ID: NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.
                      CSeq: 2 INVITE
                      Server: Asterisk PBX 1.6.2.19
                      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
                      Supported: replaces, timer
                      Contact: <sip:201@10.1.2.37>
                      Content-Type: application/sdp
                      Content-Length: 199

                      v=0
                      o=root 1256691965 1256691965 IN IP4 10.1.2.37
                      s=Asterisk PBX 1.6.2.19
                      c=IN IP4 10.1.2.37
                      t=0 0
                      m=audio 14344 RTP/AVP 0 8
                      a=rtpmap:0 PCMU/8000
                      a=rtpmap:8 PCMA/8000
                      a=ptime:20
                      a=sendrecv

                      ---
                      [Jan 17 17:27:40] WARNING[15591]: chan_sip.c:4051 retrans_pkt: Maximum retries exceeded on transmission NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/displ...etransmissions
                      [Jan 17 17:27:40] WARNING[15591]: chan_sip.c:4078 retrans_pkt: Hanging up call NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/displ...etransmissions ).
                      Scheduling destruction of SIP dialog '6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37' in 32000 ms (Method: INVITE)
                      set_destination: Parsing <sip:201@10.1.2.117:5060;transport=udp> for address/port to send to
                      set_destination: set destination to 10.1.2.117, port 5060
                      Reliably Transmitting (no NAT) to 10.1.2.117:5060:
                      BYE sip:201@10.1.2.117:5060;transport=udp SIP/2.0
                      Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK7bde824d;rport
                      Max-Forwards: 70
                      From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as65415c05
                      To: <sip:201@10.1.2.117:5060;transport=udp>;tag=0024c4 44a2fa04b09933761d-88196ad5
                      Call-ID: 6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37
                      CSeq: 103 BYE
                      User-Agent: Asterisk PBX 1.6.2.19
                      X-Asterisk-HangupCause: Normal Clearing
                      X-Asterisk-HangupCauseCode: 16
                      Content-Length: 0


                      ---
                      == Spawn extension (interno, 201, 2) exited non-zero on 'SIP/204-000000a6'

                      <--- SIP read from UDP:10.1.2.117:51922 --->
                      SIP/2.0 200 OK
                      Via: SIP/2.0/UDP 10.1.2.37:5060;branch=z9hG4bK7bde824d;rport
                      From: "Cel Rafa" <sip:204@10.1.2.37>;tag=as65415c05
                      To: <sip:201@10.1.2.117:5060;transport=udp>;tag=0024c4 44a2fa04b09933761d-88196ad5
                      Call-ID: 6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37
                      Date: Thu, 17 Jan 2013 23:27:40 GMT
                      CSeq: 103 BYE
                      Server: Cisco-CP7911G/8.3.0
                      Content-Length: 0


                      <------------->
                      --- (9 headers 0 lines) ---
                      Really destroying SIP dialog '6ceb3a4e4ee24a9910523ebe57277968@10.1.2.37' Method: INVITE
                      Really destroying SIP dialog 'NWQzMTZmMzA0ZmE5Yzc4ZGZjZjlkMzM5NjI1NTRiZDM.' Method: INVITE

                      Comentario

                      • #12
                        Hola, no logro ver que haya algo mal en el sip debug. Cuando el pbx te manda el mensaje SIP de "sonando"(ringing) tu teléfono jamás contesta con el ack y de ahi se derivan los problemas y por eso cuelga.

                        En lo que otros compañeros lo revisan mas a fondo podrías intentar con el softphone Linphone en tu celular solo para verificar.

                        Comentario

                        • #13
                          Bueno el Linphone no lo encontré disponible para mi teléfono pero instale otra aplicación y funciono perfectamente. Esto quiere decir que el problema era la aplicación. Muchas gracias. Ahora por que cuando marco a la extension de mi android no marca.. no sale la llamada?

                          Comentario

                          • #14
                            ¿Estás seguro de que tu android está bien registrado? Pasa frecuentemente de que los teléfonos móviles se "duermen" para ahorrar energía y datos y por ello, las llamadas nunca entran.

                            Asegúrate de que esté bien registrado antes de marcarle, solo para evitar esta posibilidad.

                            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

                            • #15
                              Originalmente publicado por rvazkez Ver Mensaje
                              Bueno el Linphone no lo encontré disponible para mi teléfono pero instale otra aplicación y funciono perfectamente. Esto quiere decir que el problema era la aplicación. Muchas gracias. Ahora por que cuando marco a la extension de mi android no marca.. no sale la llamada?
                              Hola,

                              Si marca, pero tu aplicación(softphone) de android no esta respondiendo correctamente, como dice Christian puede que no este bien registrado o de plano si sea un bug de la aplicación.

                              A grandes rasgos y sin ser el protocolo detallado lo que pasa es:

                              Peer-------------------------PBX
                              Quiero marcar -------------->
                              <----------------------------- Identificate primero
                              Me identifico ----------------->
                              <----------------------------- Ok, identificado correctamente
                              <----------------------------- Intentando Marcar
                              OK ------------------------------>
                              <----------------------------- Ya esta sonando el lado remoto
                              Ok ------------------------------->
                              <----------------------------- Contesto
                              OK ------------------------------>
                              RTP MEDIA <-----------------> PEER/PBX
                              hangup--------------------------->
                              <-----------------------------OK

                              Y el error viene cuando el PBX te envia lo del color azul, esta esperando el ACK de tu peer(color rojo), pero nunca llega y empieza a retransmitir el paquete hasta que se da por vencido y cuelga la llamada al no haber respueta desde el peer(tu softphone).

                              Comentario

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