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[RESUELTO] Ayuda para conectar dos centrales asterisk con troncal sip

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  • [RESUELTO] Ayuda para conectar dos centrales asterisk con troncal sip

    Buenas tarde,

    necesito conectar dos centrales asterisk a traves de sip, lei que para que se registren hace falta poner

    register => numero: pass@ipdelacentral

    y deberia registrarse pero no lo hace

    alguien sabe como??
    Editado por última vez por luquetty; https://asteriskmx.org/foros/member/1776-luquetty en 04-27-2013, 04:11 PM.

  • #2
    Creo que es más fácil si haces troncales y los pones como friend:

    en sip.conf o sip_additional.conf

    En servidor A
    Código:
    [servidorB]
    host=(la ip del servidor B)
    name=servidorA
    secret=password compartido
    type=friend
    allow=all
    context=el contexto al que permitirás acceso // si vas a homologarlos, probablemente quieras utilizar from-internal
    En el servidor B
    Código:
    [servidorA]
    host=(ip del servidor A)
    name=servidorB
    secret=password compartido
    type=friend
    allow=all
    context=el contexto al que permitirás acceso // mismo caso que el anterior
    Chécalo y dinos si pudiste.

    Saludos
    IT Specialist

    Comentario


    • #3
      En sip.conf cree


      central1

      [192.168.1.202]
      host=192.168.1.202
      name=192.168.1.201
      secret=password
      type=friend
      allow=all
      context=default


      central2

      [192.168.1.201]
      host=192.168.1.201
      name=192.168.1.202
      secret=password
      type=friend
      allow=all
      context=default


      en extension.conf
      [default]
      exten = _XX.,1,Dial(sip/${EXTEN}@192.168.1.202,30,Ttm)



      en que le estoy errando ???
      Editado por última vez por luquetty; https://asteriskmx.org/foros/member/1776-luquetty en 04-28-2013, 04:25 PM.

      Comentario


      • #4
        El dial correcto sería así:

        exten= _XX.,1,Dial(SIP/192.168.1.202/${EXTEN},30,Ttm)

        aunque te recomendaría que utilizaras nombres en vez de direcciones en los nombres de las troncales.

        tu dial podría quedar así:

        exten= _XX.,1,Dial(SIP/central2/${EXTEN},30,Ttm)
        IT Specialist

        Comentario


        • #5
          Desde ya gracias, creo que vamos avansando pero me responde congestion



          -- Executing [2000@default:1] Dial("SIP/3000-00000002", "SIP/192.168.1.201/2000,30,Ttm") in new stack
          == Using SIP RTP CoS mark 5
          -- Called SIP/192.168.1.201/2000
          -- Started music on hold, class 'default', on SIP/3000-00000002
          [Apr 28 19:07:24] NOTICE[1179]: chan_sip.c:20959 handle_response_invite: Failed to authenticate on INVITE to '"3000" <sip:3000@192.168.1.202>;tag=as166f1e29'
          -- SIP/192.168.1.201-00000003 is circuit-busy
          == Everyone is busy/congested at this time (1:0/1/0)
          -- Stopped music on hold on SIP/3000-00000002
          -- Auto fallthrough, channel 'SIP/3000-00000002' status is 'CONGESTION'

          Comentario


          • #6
            Tienes declarado el parámetro localnet en [general] de sip.conf?
            IT Specialist

            Comentario


            • #7
              no lo creo, no lo hice, como se hace??



              [general]
              localnet=192.168.1.0/255.255.255.0
              Editado por última vez por luquetty; https://asteriskmx.org/foros/member/1776-luquetty en 04-28-2013, 05:35 PM.

              Comentario


              • #8
                Si, así es como se hace, pruébalo y pega aquí un debug sip de ambos servidores cuando intentar realizar la llamada.
                IT Specialist

                Comentario


                • #9
                  si ya lo hice,



                  intento llamar de la central1 a la central2


                  en la central1 me dice

                  [Apr 28 14:06:48] WARNING[1859]: chan_sip.c:10067 process_sdp_a_audio: Got Siren7 offer at 240 00 bps, but only 32000 bps supported; ignoring.
                  [Apr 28 14:06:48] WARNING[2194]: pbx.c:4250 pbx_substitute_variables_helper_full: Error in ext ension logic (missing '}')
                  -- Executing [3000@default:1] Dial("SIP/2001-0000002c", "sip/central2/") in new stack
                  == Using SIP RTP CoS mark 5
                  -- Called sip/central2/
                  [Apr 28 14:06:48] NOTICE[1859]: chan_sip.c:20959 handle_response_invite: Failed to authenticat e on INVITE to '"2001" <sip:2001@192.168.1.201>;tag=as1c817489'
                  -- SIP/central2-0000002d is circuit-busy
                  == Everyone is busy/congested at this time (1:0/1/0)
                  -- Auto fallthrough, channel 'SIP/2001-0000002c' status is 'CONGESTION'


                  en la central2 nada,
                  Editado por última vez por luquetty; https://asteriskmx.org/foros/member/1776-luquetty en 04-28-2013, 06:52 PM.

                  Comentario


                  • #10
                    central 1

                    sip.conf

                    [plantilla](!)
                    type=friend
                    secret=lucas
                    qualify=yes
                    nat=yes
                    host=dynamic
                    canreinvite=no
                    mailbox= prueba@hotmail.com
                    context= default



                    [2000](plantilla)
                    [2001](plantilla)


                    [central2]
                    host=192.168.1.202
                    name=central1
                    secret=password
                    type=friend
                    allow=all
                    context=default
                    qualify=yes


                    [general]
                    localnet=192.168.1.0/255.255.255.0



                    extensions.conf

                    [default]
                    ; A internos
                    exten = _2XX.,1,Dial(sip/${EXTEN},30,Ttm)
                    exten = _2XX.,2,Playback(ss-noservice)
                    exten = _2XX.,3,Hangup
                    exten = _3XX.,1,Dial(sip/central2/${EXTEN],30,tTM)


                    central2

                    sip.conf

                    [3000]
                    type=friend
                    secret=lucas
                    qualify=yes
                    nat=no
                    host=dynamic
                    canreinvite=no
                    mailbox= prueba@hotmail.com



                    [central1]
                    host=192.168.1.201
                    name=central2
                    secret=password
                    type=friend
                    allow=all
                    context=default
                    qualify=yes



                    [general]
                    localnet=192.168.1.0/255.255.255.0



                    extensions.conf

                    [default]
                    exten= _2XX.,1,Dial(SIP/central1/${EXTEN},30,Ttm)
                    exten= _3XX.,1,Dial(SIP/${EXTEN},30,Ttm)

                    Comentario


                    • #11
                      Ejecuta

                      Código:
                      sip set debug on
                      En ambos servidores y pega lo que te salga en el cli de Asterisk.
                      IT Specialist

                      Comentario


                      • #12
                        Central 1



                        <--- Reliably Transmitting (NAT) to 192.168.1.4:5070 --->
                        SIP/2.0 503 Service Unavailable
                        Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPjf12010730d344c828 bb1af043bc90bcb;received=19 2.168.1.4;rport=5070
                        From: sip:2001@192.168.1.201;tag=94fa21886bff4db39110993 2020f7652
                        To: sip:3000@192.168.1.201;tag=as2f5f21fe
                        Call-ID: 13a4f2a9a52447db9d63f8faa04ec2bb
                        CSeq: 13950 INVITE
                        Server: Asterisk PBX 1.8.21.0
                        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
                        Supported: replaces, timer
                        X-Asterisk-HangupCause: Call Rejected
                        X-Asterisk-HangupCauseCode: 21
                        Content-Length: 0


                        <------------>
                        Really destroying SIP dialog '69de00a801ce6cc443dce0e674aee3a0@192.168.1.201:50 60' Method: INV ITE
                        Retransmitting #1 (NAT) to 192.168.1.4:5070:
                        SIP/2.0 503 Service Unavailable
                        Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPjf12010730d344c828 bb1af043bc90bcb;received=19 2.168.1.4;rport=5070
                        From: sip:2001@192.168.1.201;tag=94fa21886bff4db39110993 2020f7652
                        To: sip:3000@192.168.1.201;tag=as2f5f21fe
                        Call-ID: 13a4f2a9a52447db9d63f8faa04ec2bb
                        CSeq: 13950 INVITE
                        Server: Asterisk PBX 1.8.21.0
                        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
                        Supported: replaces, timer
                        -Asterisk-HangupCause: Call Rejected
                        X-Asterisk-HangupCauseCode: 21
                        Content-Length: 0


                        ---

                        <--- SIP read from UDP:192.168.1.4:5070 --->
                        ACK sip:3000@192.168.1.201 SIP/2.0
                        Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPjf12010730d3 44c828bb1af043bc90bcb
                        Max-Forwards: 70
                        From: sip:2001@192.168.1.201;tag=94fa21886bff4db39110993 2020f7652
                        To: sip:3000@192.168.1.201;tag=as2f5f21fe
                        Call-ID: 13a4f2a9a52447db9d63f8faa04ec2bb
                        CSeq: 13950 ACK
                        Content-Length: 0

                        <------------->
                        --- (8 headers 0 lines) ---
                        Really destroying SIP dialog '13a4f2a9a52447db9d63f8faa04ec2bb' Method: ACK

                        <--- SIP read from UDP:192.168.1.4:5070 --->





                        CENTRAL2

                        <--- SIP read from UDP:192.168.1.201:5060 --->
                        SIP/2.0 200 OK
                        Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK1916b532;received =192.168.1.202;rport=5060
                        From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as68311fa5
                        To: <sip:192.168.1.201>;tag=as441827c4
                        Call-ID: 6b0f7b3e11623852480d12653a74a8c9@192.168.1.202:506 0
                        CSeq: 102 OPTIONS
                        Server: Asterisk PBX 1.8.21.0
                        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
                        Supported: replaces, timer
                        Contact: <sip:192.168.1.201:5060>
                        Accept: application/sdp
                        Content-Length: 0

                        <------------->
                        --- (12 headers 0 lines) ---
                        Really destroying SIP dialog '6b0f7b3e11623852480d12653a74a8c9@192.168.1.202:50 60' Method: OPTIONS
                        Retransmitting #3 (no NAT) to 190.122.72.253:53321:
                        OPTIONS sip:3000@190.122.72.253:53321;rinstance=5add4181bb f19751;transport=UDP SIP/2.0
                        Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK26a4ec60
                        Max-Forwards: 70
                        From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as0d2ae947
                        To: <sip:3000@190.122.72.253:53321;rinstance=5add4181b bf19751;transport=UDP>
                        Contact: <sip:asterisk@192.168.1.202:5060>
                        Call-ID: 7bcecbf206c7bc9320e04376542d5b78@192.168.1.202:506 0
                        CSeq: 102 OPTIONS
                        User-Agent: Asterisk PBX 1.8.21.0
                        Date: Mon, 29 Apr 2013 00:00:52 GMT
                        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
                        Supported: replaces, timer
                        Content-Length: 0


                        ---
                        Retransmitting #4 (no NAT) to 190.122.72.253:53321:
                        OPTIONS sip:3000@190.122.72.253:53321;rinstance=5add4181bb f19751;transport=UDP SIP/2.0
                        Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK26a4ec60
                        Max-Forwards: 70
                        From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as0d2ae947
                        To: <sip:3000@190.122.72.253:53321;rinstance=5add4181b bf19751;transport=UDP>
                        Contact: <sip:asterisk@192.168.1.202:5060>
                        Call-ID: 7bcecbf206c7bc9320e04376542d5b78@192.168.1.202:506 0
                        CSeq: 102 OPTIONS
                        User-Agent: Asterisk PBX 1.8.21.0
                        Date: Mon, 29 Apr 2013 00:00:52 GMT
                        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
                        Supported: replaces, timer
                        Content-Length: 0


                        ---
                        Really destroying SIP dialog '7bcecbf206c7bc9320e04376542d5b78@192.168.1.202:50 60' Method: OPTIONS

                        Comentario


                        • #13
                          Tienes varias cosas ahí:

                          En tus trunk usa disallow=all y luego usa allow=ulaw y añade alas y GSM.
                          En tu dial plan de la central 1 tienes un corchete en lugar de una llave en el dial.
                          Y finalmente al parecer esta fallando al utenttificar el Invite. Checa las contraseñas y que ambos realms sean los mismos si usas tipo friend.

                          El debug parece incompleto trata de capturarlo todo.

                          Comentario


                          • #14
                            Ya hice los cambios
                            [central2]
                            host=192.168.1.202
                            name=central1
                            secret=password
                            type=friend
                            context=default
                            qualify=yes
                            disallow=all
                            allow=ulaw
                            allow=gsm

                            [central1]
                            host=192.168.1.201
                            name=central2
                            secret=password
                            type=friend
                            context=default
                            qualify=yes
                            disallow=all
                            allow=ulaw
                            allow=gsm

                            Central1

                            <--- SIP read from UDP:192.168.1.202:5060 --->
                            ACK sip:2001@192.168.1.201 SIP/2.0
                            Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK3b42e3cb;rport
                            Max-Forwards: 70
                            From: "3000" <sip:3000@192.168.1.202>;tag=as198d19bd
                            To: <sip:2001@192.168.1.201>;tag=as0df947c0
                            Contact: <sip:3000@192.168.1.202:5060>
                            Call-ID: 3c036b6a5b32dddb58548e0e6613a673@192.168.1.202:506 0
                            CSeq: 102 ACK
                            User-Agent: Asterisk PBX 1.8.21.0
                            Content-Length: 0

                            <------------->
                            --- (10 headers 0 lines) ---
                            Really destroying SIP dialog '3c036b6a5b32dddb58548e0e6613a673@192.168.1.202:50 60' Method: ACK

                            <--- SIP read from UDP:192.168.1.4:5070 --->
                            REGISTER sip:2001@192.168.1.201 SIP/2.0
                            Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPj4f61ce1ec83 b4ec59106cd34987a5cd7
                            Max-Forwards: 70
                            From: <sip:2001@192.168.1.201>;tag=d29741af4fcf41978fb7e 7670384ee08
                            To: <sip:2001@192.168.1.201>
                            Call-ID: 802c20bc482943299346cb4130d258d6
                            CSeq: 18203 REGISTER
                            User-Agent: AdoreSoftphone
                            Contact: <sip:2001@192.168.1.4:5070>
                            Expires: 300
                            Content-Length: 0

                            <------------->
                            --- (11 headers 0 lines) ---
                            Sending to 192.168.1.4:5070 (NAT)

                            <--- Transmitting (NAT) to 192.168.1.4:5070 --->
                            SIP/2.0 401 Unauthorized
                            Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPj4f61ce1ec83b4ec59 106cd34987a5cd7;received=19 2.168.1.4;rport=5070
                            From: <sip:2001@192.168.1.201>;tag=d29741af4fcf41978fb7e 7670384ee08
                            To: <sip:2001@192.168.1.201>;tag=as5c4a7869
                            Call-ID: 802c20bc482943299346cb4130d258d6
                            CSeq: 18203 REGISTER
                            Server: Asterisk PBX 1.8.21.0
                            Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
                            Supported: replaces, timer
                            WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2399f341"
                            Content-Length: 0

                            Scheduling destruction of SIP dialog '802c20bc482943299346cb4130d258d6' in 32000 ms (Method: R EGISTER)
                            <--- SIP read from UDP:192.168.1.4:5070 --->
                            REGISTER sip:2001@192.168.1.201 SIP/2.0
                            Via: SIP/2.0/UDP 192.168.1.4:5070;rport;branch=z9hG4bKPjb493013deff 340c6a4b9da8e0fb5bac0
                            Max-Forwards: 70
                            From: <sip:2001@192.168.1.201>;tag=d29741af4fcf41978fb7e 7670384ee08
                            To: <sip:2001@192.168.1.201>
                            Call-ID: 802c20bc482943299346cb4130d258d6
                            CSeq: 18204 REGISTER
                            User-Agent: AdoreSoftphone
                            Contact: <sip:2001@192.168.1.4:5070>
                            Expires: 300
                            Authorization: Digest username="2001", realm="asterisk", nonce="2399f341", uri="sip:2001@192.1 68.1.201", response="8717306df7315613303d000e4f037aee", algorithm=MD5
                            Content-Length: 0

                            --- (12 headers 0 lines) ---
                            Sending to 192.168.1.4:5070 (NAT)
                            Reliably Transmitting (NAT) to 192.168.1.4:5070:
                            OPTIONS sip:2001@192.168.1.4:5070 SIP/2.0
                            Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK51f7ae44;rport
                            Max-Forwards: 70
                            From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as7106eed6
                            To: <sip:2001@192.168.1.4:5070>
                            Contact: <sip:asterisk@192.168.1.201:5060>
                            Call-ID: 0780b7270634f0755f44a67a3844ebee@192.168.1.201:506 0
                            CSeq: 102 OPTIONS
                            User-Agent: Asterisk PBX 1.8.21.0
                            Date: Sun, 28 Apr 2013 17:56:27 GMT
                            Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
                            Supported: replaces, timer
                            Content-Length: 0


                            ---

                            <--- Transmitting (NAT) to 192.168.1.4:5070 --->
                            SIP/2.0 200 OK
                            Via: SIP/2.0/UDP 192.168.1.4:5070;branch=z9hG4bKPjb493013deff340c6a 4b9da8e0fb5bac0;received=19 2.168.1.4;rport=5070
                            From: <sip:2001@192.168.1.201>;tag=d29741af4fcf41978fb7e 7670384ee08
                            To: <sip:2001@192.168.1.201>;tag=as5c4a7869
                            Call-ID: 802c20bc482943299346cb4130d258d6
                            CSeq: 18204 REGISTER
                            Server: Asterisk PBX 1.8.21.0
                            Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
                            Supported: replaces, timer
                            Expires: 300
                            Contact: <sip:2001@192.168.1.4:5070>;expires=300
                            Date: Sun, 28 Apr 2013 17:56:27 GMT
                            Content-Length: 0


                            <------------>
                            Scheduling destruction of SIP dialog '1ec8f97c6c7357e05e64368135353add@192.168.1.201:50 60' in 6400 ms (Method: NOTIFY)
                            Reliably Transmitting (NAT) to 192.168.1.4:5070:
                            NOTIFY sip:2001@192.168.1.4:5070 SIP/2.0
                            Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK2bff97fd;rport
                            Max-Forwards: 70
                            From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as512a745c
                            To: <sip:2001@192.168.1.4:5070>
                            Contact: <sip:asterisk@192.168.1.201:5060>
                            Call-ID: 1ec8f97c6c7357e05e64368135353add@192.168.1.201:506 0
                            CSeq: 102 NOTIFY
                            User-Agent: Asterisk PBX 1.8.21.0
                            Event: message-summary
                            Content-Type: application/simple-message-summary
                            Content-Length: 93

                            Messages-Waiting: no
                            Message-Account: sip:asterisk@192.168.1.201
                            Voice-Message: 0/0 (0/0)

                            ---
                            Scheduling destruction of SIP dialog '802c20bc482943299346cb4130d258d6' in 32000 ms (Method: R EGISTER)

                            <--- SIP read from UDP:192.168.1.4:5070 --->
                            SIP/2.0 200 OK
                            Via: SIP/2.0/UDP 192.168.1.201:5060;rport=5060;received=192.168.1.2 01;branch=z9hG4bK51f7ae44
                            Call-ID: 0780b7270634f0755f44a67a3844ebee@192.168.1.201:506 0
                            From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as7106eed6
                            To: <sip:2001@192.168.1.4>
                            CSeq: 102 OPTIONS
                            Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
                            Accept: application/sdp, application/pidf+xml, application/xpidf+xml, message/sipfrag;version= 2.0, application/im-iscomposing+xml, text/plain
                            Supported: 100rel, norefersub
                            Allow-Events: presence, refer
                            User-Agent: AdoreSoftphone
                            Content-Type: application/sdp
                            Content-Length: 718

                            v=0
                            o=- 3576174092 3576174092 IN IP4 192.168.1.4
                            s=pjmedia
                            c=IN IP4 192.168.1.4
                            t=0 0
                            m=audio 4000 RTP/AVP 103 102 104 117 3 0 8 9 118 119 18 4 2 15 125 126 101
                            a=rtcp:4001 IN IP4 192.168.1.4
                            a=rtpmap:103 speex/16000
                            a=rtpmap:102 speex/8000
                            a=rtpmap:104 speex/32000
                            a=rtpmap:117 iLBC/8000
                            a=fmtp:117 mode=30
                            a=rtpmap:3 GSM/8000
                            a=rtpmap:0 PCMU/8000
                            a=rtpmap:8 PCMA/8000
                            a=rtpmap:9 G722/8000
                            a=rtpmap:118 AMR/8000
                            a=rtpmap:119 AMR-WB/16000
                            a=rtpmap:18 G729/8000
                            a=rtpmap:4 G723/8000
                            a=rtpmap:2 G726-32/8000
                            a=rtpmap:15 G728/8000
                            a=rtpmap:125 G7221/16000
                            a=fmtp:125 bitrate=24000
                            a=rtpmap:126 G7221/16000
                            a=fmtp:126 bitrate=32000
                            a=sendrecv
                            a=rtpmap:101 telephone-event/8000
                            a=fmtp:101 0-15
                            <------------->
                            --- (13 headers 29 lines) ---
                            Really destroying SIP dialog '0780b7270634f0755f44a67a3844ebee@192.168.1.201:50 60' Method: OPT IONS

                            <--- SIP read from UDP:192.168.1.4:5070 --->
                            SIP/2.0 200 OK
                            Via: SIP/2.0/UDP 192.168.1.201:5060;rport=5060;received=192.168.1.2 01;branch=z9hG4bK2bff97fd
                            Call-ID: 1ec8f97c6c7357e05e64368135353add@192.168.1.201:506 0
                            From: "asterisk" <sip:asterisk@192.168.1.201>;tag=as512a745c
                            To: <sip:2001@192.168.1.4>
                            CSeq: 102 NOTIFY
                            Content-Length: 0

                            <------------->
                            --- (7 headers 0 lines) ---
                            Really destroying SIP dialog '1ec8f97c6c7357e05e64368135353add@192.168.1.201:50 60' Method: NOT IFY
                            home*CLI> sip set debug off

                            Central2

                            <--- Transmitting (NAT) to 192.168.1.9:5066 --->
                            SIP/2.0 200 OK
                            Via: SIP/2.0/UDP 192.168.1.9:5066;branch=z9hG4bK-e126f7e4;received=192.168.1.9;rport=5066
                            From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
                            To: "3001" <sip:3001@192.168.1.202>;tag=as2fe25d1e
                            Call-ID: ccf9627c-fa28d39f@192.168.1.9
                            CSeq: 84 REGISTER
                            Server: Asterisk PBX 1.8.21.0
                            Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
                            Supported: replaces, timer
                            Expires: 60
                            Contact: <sip:3001@192.168.1.9:5066>;expires=60
                            Date: Mon, 29 Apr 2013 00:41:29 GMT
                            Content-Length: 0


                            <------------>
                            Scheduling destruction of SIP dialog '52fc475669f24fce125544f32519c96e@192.168.1.202:50 60' in 6400 ms (Method: NOTIFY)
                            Reliably Transmitting (NAT) to 192.168.1.9:5066:
                            NOTIFY sip:3001@192.168.1.9:5066 SIP/2.0
                            Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK49b80ec6;rport
                            Max-Forwards: 70
                            From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as53b7d78c
                            To: <sip:3001@192.168.1.9:5066>
                            Contact: <sip:asterisk@192.168.1.202:5060>
                            Call-ID: 52fc475669f24fce125544f32519c96e@192.168.1.202:506 0
                            CSeq: 102 NOTIFY
                            User-Agent: Asterisk PBX 1.8.21.0
                            Event: message-summary
                            Content-Type: application/simple-message-summary
                            Content-Length: 93

                            <--- SIP read from UDP:192.168.1.9:5066 --->
                            SIP/2.0 200 OK
                            To: <sip:3001@192.168.1.9:5066>;tag=3730351cb34f217f i0
                            From: "asterisk" <sip:asterisk@192.168.1.202>;tag=as53b7d78c
                            Call-ID: 52fc475669f24fce125544f32519c96e@192.168.1.202:506 0
                            CSeq: 102 NOTIFY
                            Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK49b80ec6
                            Server: Sipura/SPA941-4.1.8
                            Content-Length: 0


                            <--- Transmitting (NAT) to 192.168.1.9:5066 --->
                            SIP/2.0 200 OK
                            Via: SIP/2.0/UDP 192.168.1.9:5066;branch=z9hG4bK-9081e9f;received=192.168.1.9;rport=5066
                            From: "3001" <sip:3001@192.168.1.202>;tag=5138e8786dd0a5d3o0
                            To: <sip:192.168.1.202>;tag=as6d44df28
                            Call-ID: 554ddcd8-5f93cf3@192.168.1.9
                            CSeq: 93 NOTIFY
                            Server: Asterisk PBX 1.8.21.0
                            Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
                            Supported: replaces, timer
                            Content-Length: 0

                            Comentario


                            • #15
                              Siguen incompletos esos debugs no se ven los invites. Para que los tengas completos en un archivo has:

                              asterisk -rnvvvvdd | tee debugc1.txt

                              En ambos pbx y sube los archivos resultantes a pastebin.com

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