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llamadas entre extensiones remotas no se escuchan

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  • #1

    llamadas entre extensiones remotas no se escuchan

    Que tal les platico el panorama

    el PBX tiene Elastix 2.3 y esta conectado a un modem telmex en modo DMZ

    el archivo de sip_general_custom.conf esta modificado con los valores necesarios para que funcione con una cuenta dyndns la ip publica

    port = 5060
    bindaddr = 0.0.0.0
    nat=yes
    ;externip = 200.76.98.157
    localnet = 192.168.2.0/255.255.255.0
    externhost= nombre_de_dominio.dyndns.org
    externrefresh=10
    disallow=all
    allow=g729
    allow=gsm
    allow=ulaw
    allow=alaw

    defaultexpirey=60
    maxexpirey=60
    registerattempts=0

    pedantic=yes

    context = from-pstn ; Send unknown SIP callers to this context
    srvlookup=yes
    callerid = Externa
    tos=0x68
    language=es

    lo que pasa es que cuando intento hacer llamadas entre extensiones no se escucha nada entre los teléfonos IP, sin embargo si uso un softphone con zoiper usa el códec GSM pero se escuchan solo ruidos robotizados (hago la marcación de 1234 para el audio de prueba) y al cavo de unos segundos las llamadas se cortan

    Por cierto que hice pruebas con netcat para ver si los puertos UDP están abiertos y salieron positivas

    Saludos
    Etiquetas: Ninguno
  • #2
    Re: llamadas entre extensiones remotas no se escuchan

    Las capturas del sip debug serian de mucha utilidad. Otra cosa que puedes hacer es usar el programa wireshark para capturar la llamada una vez finalizada, analizala para ver si tienes audio en esa llamada, si no lo hay es un problema de ruteo si lo hay es un problema con tu server.

    Comentario

    • #3
      Re: llamadas entre extensiones remotas no se escuchan

      después del sip set debug a la extension remota salio esto que yo creo es el error
      al parecer es algo de ruteo pero no se donde corregir el error
      saludos


      [Jul 18 12:17:18] ERROR[9855] netsock2.c: getaddrinfo("189.169.174.1:15033:5060", "(null)", ...): No address associated with hostname
      [Jul 18 12:17:18] WARNING[9855] chan_sip.c: Can't find address for host '189.169.174.1:15033:5060'
      [Jul 18 12:17:18] VERBOSE[20116] app_dial.c: -- SIP/210-00000282 answered SIP/105-00000281
      [Jul 18 12:17:18] ERROR[9855] netsock2.c: getaddrinfo("189.169.174.1:15033:5060", "(null)", ...): No address associated with hostname
      [Jul 18 12:17:18] WARNING[9855] chan_sip.c: Can't find address for host '189.169.174.1:15033:5060'
      [Jul 18 12:17:19] NOTICE[9855] chan_sip.c: Correct auth, but based on stale nonce received from '"Emilio" <sip:108@192.168.2.2>;tag=300086053'
      [Jul 18 12:17:19] ERROR[9855] netsock2.c: getaddrinfo("189.169.174.1:15033:5060", "(null)", ...): No address associated with hostname
      [Jul 18 12:17:19] WARNING[9855] chan_sip.c: Can't find address for host '189.169.174.1:15033:5060'
      [Jul 18 12:17:21] ERROR[9855] netsock2.c: getaddrinfo("189.169.174.1:15033:5060", "(null)", ...): No address associated with hostname
      [Jul 18 12:17:21] WARNING[9855] chan_sip.c: Can't find address for host '189.169.174.1:15033:5060'
      [Jul 18 12:17:21] VERBOSE[9855] netsock2.c: == Using SIP RTP TOS bits 184
      [Jul 18 12:17:21] VERBOSE[9855] netsock2.c: == Using SIP RTP CoS mark 5

      Comentario

      • #4
        Re: llamadas entre extensiones remotas no se escuchan

        La dirección: 189.169.174.1:15033:5060 es inválida, en el SIP debug se sigue viendo esta dirección?

        Usas Csimple o algún otro softphone?

        Comentario

        • #5
          Re: llamadas entre extensiones remotas no se escuchan

          si de echo es la ip en la que actualmente están las extensiones remotas cuando uso zoiper ya puedo oír la grabación sin embargo se cuelga automáticamente a los 5 seg.

          Comentario

          • #6
            Re: llamadas entre extensiones remotas no se escuchan

            Puedes poner el SIP debug completo de una llamada que se corta a los 5segs?

            Comentario

            • #7
              Re: llamadas entre extensiones remotas no se escuchan

              este es la parte cuando cuelga la llamada
              Código:
              ---
Retransmitting #6 (NAT) to 189.169.174.1:14235:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-518a85c79c045b45-1---d8754z-;received=189.169.174.1;rport=14235
From: "209"<sip:209@trav.org;transport=UDP>;tag=5d166a20
To: <sip:1234@tra.org;transport=UDP>;tag=as3417eff3
Call-ID: YmFkMTkzYTE3ZWUxZmI4NWJmZTk0ZTRhNjhlY2IxNTQ.
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1234@189.161.219.47:5060>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 1460775096 1460775096 IN IP4 189.161.219.47
s=Asterisk PBX 1.8.11.0
c=IN IP4 189.161.219.47
t=0 0
m=audio 14744 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
  == Spawn extension (from-internal, 1234, 1) exited non-zero on 'SIP/209-0000000c'
    -- Executing [h@from-internal:1] Macro("SIP/209-0000000c", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/209-0000000c", "1?endmixmoncheck") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/209-0000000c", "End of MIXMON check") in new stack
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/209-0000000c", "1?nomeetmemon") in new stack
    -- Goto (macro-hangupcall,s,15)
    -- Executing [s@macro-hangupcall:15] NoOp("SIP/209-0000000c", "MEETME_RECORDINGFILE=") in new stack
    -- Executing [s@macro-hangupcall:16] GotoIf("SIP/209-0000000c", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,18)
    -- Executing [s@macro-hangupcall:18] NoOp("SIP/209-0000000c", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:19] GotoIf("SIP/209-0000000c", "1?noautomon2") in new stack
    -- Goto (macro-hangupcall,s,25)
    -- Executing [s@macro-hangupcall:25] NoOp("SIP/209-0000000c", "MONITOR_FILENAME=") in new stack
    -- Executing [s@macro-hangupcall:26] GotoIf("SIP/209-0000000c", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,29)
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/209-0000000c", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,32)
    -- Executing [s@macro-hangupcall:32] GotoIf("SIP/209-0000000c", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,34)
    -- Executing [s@macro-hangupcall:34] Hangup("SIP/209-0000000c", "") in new stack
  == Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/209-0000000c' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/209-0000000c'
Scheduling destruction of SIP dialog 'YmFkMTkzYTE3ZWUxZmI4NWJmZTk0ZTRhNjhlY2IxNTQ.' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:209@189.169.174.1:14235;transport=UDP> for address/port to send to
set_destination: set destination to 189.169.174.1:14235
Reliably Transmitting (NAT) to 189.169.174.1:14235:
BYE sip:209@189.169.174.1:14235;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 189.161.219.47:5060;branch=z9hG4bK4fd3e919;rport
Max-Forwards: 70
From: <sip:1234@tra.org;transport=UDP>;tag=as3417eff3
To: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=5d166a20
Call-ID: YmFkMTkzYTE3ZWUxZmI4NWJmZTk0ZTRhNjhlY2IxNTQ.
CSeq: 102 BYE
User-Agent: FPBX-2.8.1(1.8.11.0)
Proxy-Authorization: Digest username="209", realm="asterisk", algorithm=MD5, uri="sip:tra.org", nonce="", response="a78195072c90506c578d39f457cd0f9d"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0


---
Retransmitting #1 (NAT) to 189.169.174.1:14235:
BYE sip:209@189.169.174.1:14235;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 189.161.219.47:5060;branch=z9hG4bK4fd3e919;rport
Max-Forwards: 70
From: <sip:1234@tra.org;transport=UDP>;tag=as3417eff3
To: "209"<sip:209@trav.org;transport=UDP>;tag=5d166a20
Call-ID: YmFkMTkzYTE3ZWUxZmI4NWJmZTk0ZTRhNjhlY2IxNTQ.
CSeq: 102 BYE
User-Agent: FPBX-2.8.1(1.8.11.0)
Proxy-Authorization: Digest username="209", realm="asterisk", algorithm=MD5, uri="sip:trav.org", nonce="", response="a78195072c90506c578d39f457cd0f9d"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0


---

<--- SIP read from UDP:189.169.174.1:14235 --->
SIP/2.0 200 OK

              Comentario

              • #8
                Re: llamadas entre extensiones remotas no se escuchan

                Para poder ayudarte necesito(amos) el debug completo desde que inciia la llamada hasta que cuelga.

                Comentario

                • #9
                  Re: llamadas entre extensiones remotas no se escuchan

                  como lo puedo sacar lo que pasa es que en la consola de putty no me aparece el log completo el inicio se borra

                  Comentario

                  • #10
                    Re: llamadas entre extensiones remotas no se escuchan

                    Con el comando: "asterisk -rnvvvvdddd | tee clilog.txt"

                    Cuando te salgas del cli de asterisk en el directorio donde estas veras un archivo llamado clilog.txt el cual tendrá toda la información

                    Comentario

                    • #11
                      Re: llamadas entre extensiones remotas no se escuchan

                      son 2 llamadas la primera con un softphone con la que escucho una parte de la agravación y la 2 con un telefono ip con la que no se escuhca nada

                      Código:
                      Asterisk 1.8.11.0, Copyright (C) 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
  == Parsing '/etc/asterisk/asterisk.conf': Parsing /etc/asterisk/asterisk.conf
  == Found
Seeding global EID '00:14:c2:54:c7:62' from 'eth0' using 'siocgifhwaddr'
  == Parsing '/etc/asterisk/extconfig.conf': Parsing /etc/asterisk/extconfig.conf
  == Found
Connected to Asterisk 1.8.11.0 currently running on travelpbx (pid = 3406)
travelpbx*CLI> 
Verbosity was 3 and is now 4
Core debug was 0 and is now 4

travelpbx*CLI> sip set debug peer 209
travelpbx*CLI> 
SIP Debugging Enabled for IP: 189.169.174.1

travelpbx*CLI> 

<--- SIP read from UDP:189.169.174.1:14235 --->
REGISTER sip:travelpbx.dyndns.org;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-03b65a8354a39804-1---d8754z-
Max-Forwards: 70
Contact: <sip:209@189.169.174.1:14235;rinstance=8c5f315a8591c3b0;transport=UDP>
To: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=6626391e
Call-ID: ZjMwZTlhYzM1MTI3MmY5OWZjODAzNzE4OTE5NWYzYjk.
CSeq: 476 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Authorization: Digest username="209",realm="asterisk",nonce="13b99994",uri="sip:travelpbx.dyndns.org;transport=UDP",response="9e73a63b2b554eadc8b28dacfad53ae7",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to 189.169.174.1:14235 (NAT)

travelpbx*CLI> 

<--- Transmitting (NAT) to 189.169.174.1:14235 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-03b65a8354a39804-1---d8754z-;received=189.169.174.1;rport=14235
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=6626391e
To: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=as4e57bb7f
Call-ID: ZjMwZTlhYzM1MTI3MmY5OWZjODAzNzE4OTE5NWYzYjk.
CSeq: 476 REGISTER
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="58ca6619"
Content-Length: 0


<------------>

travelpbx*CLI> 
Scheduling destruction of SIP dialog 'ZjMwZTlhYzM1MTI3MmY5OWZjODAzNzE4OTE5NWYzYjk.' in 32000 ms (Method: REGISTER)

travelpbx*CLI> 

<--- SIP read from UDP:189.169.174.1:14235 --->
REGISTER sip:travelpbx.dyndns.org;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-98d6804acca828f7-1---d8754z-
Max-Forwards: 70
Contact: <sip:209@189.169.174.1:14235;rinstance=8c5f315a8591c3b0;transport=UDP>
To: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=6626391e
Call-ID: ZjMwZTlhYzM1MTI3MmY5OWZjODAzNzE4OTE5NWYzYjk.
CSeq: 477 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Authorization: Digest username="209",realm="asterisk",nonce="58ca6619",uri="sip:travelpbx.dyndns.org;transport=UDP",response="e17a4d78fb51b9872adcba8b3899bdce",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 0

<------------->

travelpbx*CLI> 
--- (15 headers 0 lines) ---
Sending to 189.169.174.1:14235 (NAT)
Reliably Transmitting (NAT) to 189.169.174.1:14235:
OPTIONS sip:209@189.169.174.1:14235;rinstance=8c5f315a8591c3b0;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK60fb180a;rport
Max-Forwards: 70
From: "Externa" <sip:Externa@127.0.0.1>;tag=as3db65f8f
To: <sip:209@189.169.174.1:14235;rinstance=8c5f315a8591c3b0;transport=UDP>
Contact: <sip:Externa@127.0.0.1:5060>
Call-ID: 0697c9327731a9df54d4d96472000c90@127.0.0.1:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.11.0)
Date: Wed, 18 Jul 2012 21:09:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 189.169.174.1:14235 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-98d6804acca828f7-1---d8754z-;received=189.169.174.1;rport=14235
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=6626391e
To: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=as4e57bb7f
Call-ID: ZjMwZTlhYzM1MTI3MmY5OWZjODAzNzE4OTE5NWYzYjk.
CSeq: 477 REGISTER
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:209@189.169.174.1:14235;rinstance=8c5f315a8591c3b0;transport=UDP>;expires=60
Date: Wed, 18 Jul 2012 21:09:39 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ZjMwZTlhYzM1MTI3MmY5OWZjODAzNzE4OTE5NWYzYjk.' in 32000 ms (Method: REGISTER)

travelpbx*CLI> 

<--- SIP read from UDP:189.169.174.1:14235 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK60fb180a;rport=5060;received=189.192.79.10
Contact: <sip:189.169.174.1:14235>
To: <sip:209@189.169.174.1:14235;rinstance=8c5f315a8591c3b0;transport=UDP>;tag=e47fb458
From: "Externa"<sip:Externa@127.0.0.1>;tag=as3db65f8f
Call-ID: 0697c9327731a9df54d4d96472000c90@127.0.0.1:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper rev.11137
Allow-Events: presence, kpml
Content-Length: 0

<------------->

travelpbx*CLI> 
--- (14 headers 0 lines) ---
Really destroying SIP dialog '0697c9327731a9df54d4d96472000c90@127.0.0.1:5060' Method: OPTIONS

travelpbx*CLI> 

<--- SIP read from UDP:189.169.174.1:14235 --->
INVITE sip:1234@travelpbx.dyndns.org;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-2a18d5e0e0443eb5-1---d8754z-
Max-Forwards: 70
Contact: <sip:209@189.169.174.1:14235;transport=UDP>
To: <sip:1234@travelpbx.dyndns.org;transport=UDP>
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
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.11137
Allow-Events: presence, kpml
Content-Length: 330

v=0
o=Zoiper_user 0 0 IN IP4 189.169.174.1
s=Zoiper_session
c=IN IP4 189.169.174.1
t=0 0
m=audio 29922 RTP/AVP 3 8 0 98 110 101
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->

travelpbx*CLI> 
--- (14 headers 15 lines) ---

travelpbx*CLI> 
Sending to 189.169.174.1:14235 (NAT)

travelpbx*CLI> 
Using INVITE request as basis request - YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.

travelpbx*CLI> 
Found peer '209' for '209' from 189.169.174.1:14235

<--- Reliably Transmitting (NAT) to 189.169.174.1:14235 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-2a18d5e0e0443eb5-1---d8754z-;received=189.169.174.1;rport=14235
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
To: <sip:1234@travelpbx.dyndns.org;transport=UDP>;tag=as37f1e11f
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0e3ee712"
Content-Length: 0


<------------>

travelpbx*CLI> 
Scheduling destruction of SIP dialog 'YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.' in 6400 ms (Method: INVITE)

travelpbx*CLI> 
Retransmitting #1 (NAT) to 189.169.174.1:14235:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-2a18d5e0e0443eb5-1---d8754z-;received=189.169.174.1;rport=14235
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
To: <sip:1234@travelpbx.dyndns.org;transport=UDP>;tag=as37f1e11f
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0e3ee712"
Content-Length: 0


---

travelpbx*CLI> 

<--- SIP read from UDP:189.169.174.1:14235 --->
ACK sip:1234@travelpbx.dyndns.org;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-2a18d5e0e0443eb5-1---d8754z-
Max-Forwards: 70
To: <sip:1234@travelpbx.dyndns.org;transport=UDP>;tag=as37f1e11f
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 1 ACK
Content-Length: 0

<------------->

travelpbx*CLI> 
--- (8 headers 0 lines) ---

travelpbx*CLI> 

<--- SIP read from UDP:189.169.174.1:14235 --->
INVITE sip:1234@travelpbx.dyndns.org;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-b03edb215b07ae51-1---d8754z-
Max-Forwards: 70
Contact: <sip:209@189.169.174.1:14235;transport=UDP>
To: <sip:1234@travelpbx.dyndns.org;transport=UDP>
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
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.11137
Authorization: Digest username="209",realm="asterisk",nonce="0e3ee712",uri="sip:1234@travelpbx.dyndns.org;transport=UDP",response="b0bdeecf0afb1e9eb84e9163a9a1be7a",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 330

v=0
o=Zoiper_user 0 0 IN IP4 189.169.174.1
s=Zoiper_session
c=IN IP4 189.169.174.1
t=0 0
m=audio 29922 RTP/AVP 3 8 0 98 110 101
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->

travelpbx*CLI> 
--- (15 headers 15 lines) ---

travelpbx*CLI> 
Sending to 189.169.174.1:14235 (NAT)
Using INVITE request as basis request - YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.

travelpbx*CLI> 
Found peer '209' for '209' from 189.169.174.1:14235
  == Using SIP RTP TOS bits 184

travelpbx*CLI> 
  == Using SIP RTP CoS mark 5

travelpbx*CLI> 
Found RTP audio format 3
Found RTP audio format 8

travelpbx*CLI> 
Found RTP audio format 0
Found RTP audio format 98

travelpbx*CLI> 
Found RTP audio format 110
Found RTP audio format 101
Found audio description format GSM for ID 3

travelpbx*CLI> 
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0

travelpbx*CLI> 
Found audio description format iLBC for ID 98
Found audio description format speex for ID 110

travelpbx*CLI> 
Found audio description format telephone-event for ID 101

travelpbx*CLI> 
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)

travelpbx*CLI> 
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

travelpbx*CLI> 
Peer audio RTP is at port 189.169.174.1:29922

travelpbx*CLI> 
Looking for 1234 in from-internal (domain travelpbx.dyndns.org)

travelpbx*CLI> 
list_route: hop: <sip:209@189.169.174.1:14235;transport=UDP>

travelpbx*CLI> 

<--- Transmitting (NAT) to 189.169.174.1:14235 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-b03edb215b07ae51-1---d8754z-;received=189.169.174.1;rport=14235
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
To: <sip:1234@travelpbx.dyndns.org;transport=UDP>
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1234@127.0.0.1:5060>
Content-Length: 0


<------------>

travelpbx*CLI> 
    -- Executing [1234@from-internal:1] Playback("SIP/209-0000000f", "demo-congrats") in new stack

travelpbx*CLI> 
Audio is at 13486

travelpbx*CLI> 
Adding codec 0x2 (gsm) to SDP

travelpbx*CLI> 
Adding codec 0x4 (ulaw) to SDP

travelpbx*CLI> 
Adding codec 0x8 (alaw) to SDP

travelpbx*CLI> 
Adding non-codec 0x1 (telephone-event) to SDP

travelpbx*CLI> 

<--- Reliably Transmitting (NAT) to 189.169.174.1:14235 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-b03edb215b07ae51-1---d8754z-;received=189.169.174.1;rport=14235
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
To: <sip:1234@travelpbx.dyndns.org;transport=UDP>;tag=as36efb26f
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1234@127.0.0.1:5060>
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1116010417 1116010417 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.11.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 13486 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

travelpbx*CLI> 

<--- SIP read from UDP:189.169.174.1:14235 --->
ACK sip:1234@travelpbx.dyndns.org;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-2a18d5e0e0443eb5-1---d8754z-
Max-Forwards: 70
To: <sip:1234@travelpbx.dyndns.org;transport=UDP>;tag=as37f1e11f
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 1 ACK
Content-Length: 0

<------------->

travelpbx*CLI> 
--- (8 headers 0 lines) ---

travelpbx*CLI> 
Retransmitting #1 (NAT) to 189.169.174.1:14235:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-b03edb215b07ae51-1---d8754z-;received=189.169.174.1;rport=14235
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
To: <sip:1234@travelpbx.dyndns.org;transport=UDP>;tag=as36efb26f
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1234@127.0.0.1:5060>
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1116010417 1116010417 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.11.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 13486 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

travelpbx*CLI> 
Retransmitting #2 (NAT) to 189.169.174.1:14235:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-b03edb215b07ae51-1---d8754z-;received=189.169.174.1;rport=14235
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
To: <sip:1234@travelpbx.dyndns.org;transport=UDP>;tag=as36efb26f
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1234@127.0.0.1:5060>
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1116010417 1116010417 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.11.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 13486 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

travelpbx*CLI> 
    -- <SIP/209-0000000f> Playing 'demo-congrats.gsm' (language 'es')

travelpbx*CLI> 
Retransmitting #3 (NAT) to 189.169.174.1:14235:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-b03edb215b07ae51-1---d8754z-;received=189.169.174.1;rport=14235
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
To: <sip:1234@travelpbx.dyndns.org;transport=UDP>;tag=as36efb26f
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1234@127.0.0.1:5060>
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1116010417 1116010417 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.11.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 13486 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

travelpbx*CLI> 
Retransmitting #4 (NAT) to 189.169.174.1:14235:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-b03edb215b07ae51-1---d8754z-;received=189.169.174.1;rport=14235
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
To: <sip:1234@travelpbx.dyndns.org;transport=UDP>;tag=as36efb26f
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1234@127.0.0.1:5060>
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1116010417 1116010417 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.11.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 13486 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

travelpbx*CLI> 
Retransmitting #5 (NAT) to 189.169.174.1:14235:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-b03edb215b07ae51-1---d8754z-;received=189.169.174.1;rport=14235
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
To: <sip:1234@travelpbx.dyndns.org;transport=UDP>;tag=as36efb26f
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1234@127.0.0.1:5060>
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1116010417 1116010417 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.11.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 13486 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

travelpbx*CLI> 
Retransmitting #6 (NAT) to 189.169.174.1:14235:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-b03edb215b07ae51-1---d8754z-;received=189.169.174.1;rport=14235
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
To: <sip:1234@travelpbx.dyndns.org;transport=UDP>;tag=as36efb26f
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1234@127.0.0.1:5060>
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1116010417 1116010417 IN IP4 127.0.0.1
s=Asterisk PBX 1.8.11.0
c=IN IP4 127.0.0.1
t=0 0
m=audio 13486 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

travelpbx*CLI> 
  == Spawn extension (from-internal, 1234, 1) exited non-zero on 'SIP/209-0000000f'

travelpbx*CLI> 
    -- Executing [h@from-internal:1] Macro("SIP/209-0000000f", "hangupcall") in new stack

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/209-0000000f", "1?endmixmoncheck") in new stack

travelpbx*CLI> 
    -- Goto (macro-hangupcall,s,9)

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/209-0000000f", "End of MIXMON check") in new stack

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/209-0000000f", "1?nomeetmemon") in new stack

travelpbx*CLI> 
    -- Goto (macro-hangupcall,s,15)

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:15] NoOp("SIP/209-0000000f", "MEETME_RECORDINGFILE=") in new stack

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:16] GotoIf("SIP/209-0000000f", "1?noautomon") in new stack

travelpbx*CLI> 
    -- Goto (macro-hangupcall,s,18)

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:18] NoOp("SIP/209-0000000f", "TOUCH_MONITOR_OUTPUT=") in new stack

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:19] GotoIf("SIP/209-0000000f", "1?noautomon2") in new stack

travelpbx*CLI> 
    -- Goto (macro-hangupcall,s,25)

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:25] NoOp("SIP/209-0000000f", "MONITOR_FILENAME=") in new stack

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:26] GotoIf("SIP/209-0000000f", "1?skiprg") in new stack

travelpbx*CLI> 
    -- Goto (macro-hangupcall,s,29)
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/209-0000000f", "1?skipblkvm") in new stack

travelpbx*CLI> 
    -- Goto (macro-hangupcall,s,32)

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:32] GotoIf("SIP/209-0000000f", "1?theend") in new stack

travelpbx*CLI> 
    -- Goto (macro-hangupcall,s,34)

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:34] Hangup("SIP/209-0000000f", "") in new stack

travelpbx*CLI> 
  == Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/209-0000000f' in macro 'hangupcall'

travelpbx*CLI> 
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/209-0000000f'

travelpbx*CLI> 
Scheduling destruction of SIP dialog 'YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.' in 6400 ms (Method: INVITE)

travelpbx*CLI> 
set_destination: Parsing <sip:209@189.169.174.1:14235;transport=UDP> for address/port to send to
set_destination: set destination to 189.169.174.1:14235

travelpbx*CLI> 
Reliably Transmitting (NAT) to 189.169.174.1:14235:
BYE sip:209@189.169.174.1:14235;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK3c956ca2;rport
Max-Forwards: 70
From: <sip:1234@travelpbx.dyndns.org;transport=UDP>;tag=as36efb26f
To: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 102 BYE
User-Agent: FPBX-2.8.1(1.8.11.0)
Proxy-Authorization: Digest username="209", realm="asterisk", algorithm=MD5, uri="sip:travelpbx.dyndns.org", nonce="", response="a78195072c90506c578d39f457cd0f9d"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0


---

travelpbx*CLI> 
Retransmitting #1 (NAT) to 189.169.174.1:14235:
BYE sip:209@189.169.174.1:14235;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK3c956ca2;rport
Max-Forwards: 70
From: <sip:1234@travelpbx.dyndns.org;transport=UDP>;tag=as36efb26f
To: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 102 BYE
User-Agent: FPBX-2.8.1(1.8.11.0)
Proxy-Authorization: Digest username="209", realm="asterisk", algorithm=MD5, uri="sip:travelpbx.dyndns.org", nonce="", response="a78195072c90506c578d39f457cd0f9d"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0


---

travelpbx*CLI> 

<--- SIP read from UDP:189.169.174.1:14235 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK3c956ca2;rport=5060;received=189.192.79.10
Contact: <sip:209@189.169.174.1:14235;transport=UDP>
To: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
From: <sip:1234@travelpbx.dyndns.org;transport=UDP>;tag=as36efb26f
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 102 BYE
User-Agent: Zoiper rev.11137
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.' Method: INVITE

travelpbx*CLI> 

<--- SIP read from UDP:189.169.174.1:14235 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK3c956ca2;rport=5060;received=189.192.79.10
Contact: <sip:209@189.169.174.1:14235;transport=UDP>
To: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
From: <sip:1234@travelpbx.dyndns.org;transport=UDP>;tag=as36efb26f
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 102 BYE
User-Agent: Zoiper rev.11137
Content-Length: 0

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

travelpbx*CLI> 
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5

travelpbx*CLI> 
    -- Executing [1234@from-internal:1] Playback("SIP/105-00000010", "demo-congrats") in new stack

travelpbx*CLI> 
    -- <SIP/105-00000010> Playing 'demo-congrats.gsm' (language 'es')

travelpbx*CLI> 
    -- Remote UNIX connection

travelpbx*CLI> 
    -- Remote UNIX connection disconnected

travelpbx*CLI> 

<--- SIP read from UDP:189.169.174.1:14235 --->


<------------->

travelpbx*CLI> 
  == Spawn extension (from-internal, 1234, 1) exited non-zero on 'SIP/105-00000010'

travelpbx*CLI> 
    -- Executing [h@from-internal:1] Macro("SIP/105-00000010", "hangupcall") in new stack

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/105-00000010", "1?endmixmoncheck") in new stack

travelpbx*CLI> 
    -- Goto (macro-hangupcall,s,9)

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/105-00000010", "End of MIXMON check") in new stack

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/105-00000010", "1?nomeetmemon") in new stack

travelpbx*CLI> 
    -- Goto (macro-hangupcall,s,15)

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:15] NoOp("SIP/105-00000010", "MEETME_RECORDINGFILE=") in new stack

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:16] GotoIf("SIP/105-00000010", "1?noautomon") in new stack

travelpbx*CLI> 
    -- Goto (macro-hangupcall,s,18)

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:18] NoOp("SIP/105-00000010", "TOUCH_MONITOR_OUTPUT=") in new stack

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:19] GotoIf("SIP/105-00000010", "1?noautomon2") in new stack

travelpbx*CLI> 
    -- Goto (macro-hangupcall,s,25)

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:25] NoOp("SIP/105-00000010", "MONITOR_FILENAME=") in new stack

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:26] GotoIf("SIP/105-00000010", "1?skiprg") in new stack

travelpbx*CLI> 
    -- Goto (macro-hangupcall,s,29)

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/105-00000010", "1?skipblkvm") in new stack

travelpbx*CLI> 
    -- Goto (macro-hangupcall,s,32)

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:32] GotoIf("SIP/105-00000010", "1?theend") in new stack

travelpbx*CLI> 
    -- Goto (macro-hangupcall,s,34)

travelpbx*CLI> 
    -- Executing [s@macro-hangupcall:34] Hangup("SIP/105-00000010", "") in new stack

travelpbx*CLI> 
  == Spawn extension (macro-hangupcall, s, 34) exited non-zero on 'SIP/105-00000010' in macro 'hangupcall'

travelpbx*CLI> 
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/105-00000010'

travelpbx*CLI> 
Really destroying SIP dialog 'ZjMwZTlhYzM1MTI3MmY5OWZjODAzNzE4OTE5NWYzYjk.' Method: REGISTER

travelpbx*CLI> 
Really destroying SIP dialog 'NDg5ZTRlZjg2ZTMyODY4N2MxZTE5YTY1NDEyMmI1Mzg.' Method: SUBSCRIBE

travelpbx*CLI> exit
Executing last minute cleanups
Asterisk ending (0).

                      Comentario

                      • #12
                        Re: llamadas entre extensiones remotas no se escuchan

                        Tu zoiper tiene habilitado el STUN?

                        Por otro lado veo que tu PBX retransmite 6 veces la voz antes de colgar lo que quiere decir que cuelga porque pierde el contacto. Tengo la impresión de que se debe a que en la negociación SDP la IP de tu asterisk la toma como localhost:

                        Código:
                        Retransmitting #6 (NAT) to 189.169.174.1:14235:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 189.169.174.1:14235;branch=z9hG4bK-d8754z-b03edb215b07ae51-1---d8754z-;received=189.169.174.1;rport=14235
From: "209"<sip:209@travelpbx.dyndns.org;transport=UDP>;tag=2148d42c
To: <sip:1234@travelpbx.dyndns.org;transport=UDP>;tag=as36efb26f
Call-ID: YTU0ZDUyMTlhYWM3NmZhM2ZiNTlhZGRmM2NhMGY0MTY.
CSeq: 2 INVITE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1234@127.0.0.1:5060>
Content-Type: application/sdp
Content-Length: 278

v=0
o=root 1116010417 1116010417 [b]IN IP4 127.0.0.1[/b]
s=Asterisk PBX 1.8.11.0
[b]c=IN IP4 127.0.0.1[/b]
t=0 0
m=audio 13486 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
                        En tu archivo /etc/hosts añade también el dominio de tu maquina con la IP publica, o asegúrate que tu DNS resuelva bien tu Dominio(IP Publica) para que no la envié como localhost.

                        Comentario

                        • #13
                          Re: llamadas entre extensiones remotas no se escuchan

                          O gracias al parecer ya funciono el archivo lo modifique de esta forma

                          solo tengo una pregunta si mi ip es dinamica que pasa cuando cambie? ¿hay forma de que la ip se actualice automáticamente en el archivos hosts?

                          Muchas Gracias XD

                          Código:
                          # Do not remove the following line, or various programs
# that require network functionality will fail.
189.192.79.10   tra.dyn.org
127.0.0.1       localhost tra.dyn.org localhost.localdomain

                          Comentario

                          • #14
                            Re: llamadas entre extensiones remotas no se escuchan

                            Tendrias que crear un script que la actualice manualmente. Usando awk o sed deberia ser posible.

                            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
                              hola buen dia con todos. tengo un problema monte un servidor virtual con virtualbox y le instale elastix todo funciona bien menos las llamadas sip que se conecten al ip publica de la elastix ya que no se escuha nada pero entre internos de la red privada si, que estoy haciendo mal?? espero que me puedan ayudar

                              Comentario

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