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  • #1

    Problema en escuchar algunas llamadas

    Hola que tal!

    Pues tengo un problema que no he sabido resolver, espero por favor me puedan guiar para poder resolverlo.

    Pues bien, resulta que tengo un servidor montado con Asterisk 1.8 el cual alimenta varias extensiones IP en telefonos CISCO, algunas localmente y otras a través de una VPN en otros sitios.

    El problema que presento es que a veces al momento de realizar una llamada a alguna de las extensiones conectadas a través de la VPN, ellos si pueden escucharme, pero yo a veces no los escucho y viceversa.

    Creí que era un problema con la NAT, así que en el archivo sip.conf agregué la siguiente información:

    Código:
    nat = yes

localnet=192.168.0.1/255.255.255.0
localnet=192.168.1.1/255.255.255.0
localnet=192.168.2.1/255.255.255.0
localnet=192.168.3.1/255.255.255.0
localnet=192.168.4.1/255.255.255.0
localnet=192.168.5.1/255.255.255.0
localnet=192.168.6.1/255.255.255.0
localnet=192.168.7.1/255.255.255.0
localnet=192.168.8.1/255.255.255.0
localnet=192.168.10.1/255.255.255.0

externaddr = 187.177.181.224
externhost = xxxxxxxxx.dyndns.org
externrefresh = 120
externip = 187.177.181.224
    Donde el externaddr y el externip es mi puerta de enlace de mi servicio (AXTEL)

    despues de eso el problema pareció solucionarse un día, pero al día siguiente volvió a suceder. Así que procedí a revisar el log de asterisk durante una llamada con problema, pero solo me arroja lo siguiente:

    Código:
    == Spawn extension (macro-llamada, s, 2) exited non-zero on 'SIP/1012-00000d52' in macro 'llamada'
  == Spawn extension (digitalife, 1201, 1) exited non-zero on 'SIP/1012-00000d52'
  == Using SIP RTP CoS mark 5
    -- Executing [1301@digitalife:1] Macro("SIP/1012-00000d56", "llamada,SIP/1301")
    -- Executing [s@macro-llamada:1] NoOp("SIP/1012-00000d56", "Llamada a SIP/1301 en proceso") in new stack
    -- Executing [s@macro-llamada:2] Dial("SIP/1012-00000d56", "SIP/1301") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/1301
    -- SIP/1301-00000d57 is ringing
    -- SIP/1301-00000d57 answered SIP/1012-00000d56
    -- Remotely bridging SIP/1012-00000d56 and SIP/1301-00000d57
    No veo ningun problema en el enlace, asi que procedí a ver los clientes conectados y me arroja lo siguiente:

    Código:
    srvTelIP*CLI> sip show peers
Name/username              Host                                    Dyn Forcerport ACL Port     Status     Realtime
1001/1001                  192.168.0.161                            D   N      5060     Unmonitored Cached RT
1002/1002                  192.168.0.164                            D   N      5060     Unmonitored Cached RT
1003/1003                  192.168.0.163                            D   N      5060     Unmonitored Cached RT
1004/1004                  192.168.0.165                            D   N      5060     Unmonitored Cached RT
1005/1005                  192.168.0.167                            D   N      5060     Unmonitored Cached RT
1006/1006                  192.168.0.168                            D   N      5060     Unmonitored Cached RT
1007/1007                  192.168.0.171                            D   N      5060     Unmonitored Cached RT
1008/1008                  192.168.0.166                            D   N      5060     Unmonitored Cached RT
1009/1009                  192.168.0.172                            D   N      5060     Unmonitored Cached RT
1010/1010                  192.168.0.173                            D   N      5060     Unmonitored Cached RT
1011/1011                  192.168.0.180                            D   N      5060     Unmonitored Cached RT
1012/1012                  192.168.0.162                            D   N      5060     Unmonitored Cached RT
1013/1013                  192.168.0.174                            D   N      5060     Unmonitored Cached RT
1014/1014                  192.168.0.175                            D   N      5060     Unmonitored Cached RT
1015/1015                  192.168.0.176                            D   N      5060     Unmonitored Cached RT
1016                       (Unspecified)                            D   N      0        Unmonitored Cached RT
1018/1018                  192.168.0.169                            D   N      5060     Unmonitored Cached RT
1019                       (Unspecified)                            D   N      0        Unmonitored Cached RT
1020/1020                  192.168.0.177                            D   N      5060     Unmonitored Cached RT
1022                       (Unspecified)                            D   N      0        Unmonitored Cached RT
1026/1026                  192.168.0.1                              D   N      5061     Unmonitored Cached RT
1101/1101                  187.163.25.92                            D   N      1026     Unmonitored Cached RT
1102/1102                  187.163.25.92                            D   N      1027     Unmonitored Cached RT
1103/1103                  187.163.25.92                            D   N      5060     Unmonitored Cached RT
1104/1104                  187.163.25.92                            D   N      5061     Unmonitored Cached RT
1105/1105                  187.163.25.92                            D   N      1024     Unmonitored Cached RT
1106/1106                  192.168.1.121                            D   N      5060     Unmonitored Cached RT
1108/1108                  187.163.25.92                            D   N      1025     Unmonitored Cached RT
1201/1201                  192.168.2.124                            D   N      5060     Unmonitored Cached RT
1202/1202                  187.163.25.171                           D   N      5060     Unmonitored Cached RT
1301/1301                  187.163.26.52                            D   N      1040     Unmonitored Cached RT
1302/1302                  187.163.26.52                            D   N      5060     Unmonitored Cached RT
1401/1401                  187.163.10.252                           D   N      5060     Unmonitored Cached RT
1402/1402                  187.163.10.252                           D   N      1245     Unmonitored Cached RT
1501/1501                  187.133.175.181                          D   N      5060     Unmonitored Cached RT
1502/1502                  192.168.5.120                            D   N      5060     Unmonitored Cached RT
1601/1601                  192.168.6.125                            D   N      5060     Unmonitored Cached RT
1602/1602                  192.168.6.120                            D   N      5060     Unmonitored Cached RT
38 sip peers [Monitored: 0 online, 0 offline Unmonitored: 35 online, 3 offline]
    Con lo cual observo que la extensión a la que deseo llamar esta en otro puerto diferente y creo que ese es el problema, pero al yo llamar a la 1105 que tampoco tiene el puerto 5060 si existe un enlace en ambas partes, lo cual se me hace muy extraño :s

    Ya abrí el puerto 5060 en todos mis routers pero aun sigo presentando el problema.

    Espero me puedan ayudar en mi extensa pregunta jejeje.

    Saludos.
    Etiquetas: Ninguno
  • #2
    Re: Problema en escuchar algunas llamadas

    Hola, lo del puerto en que se conecta la extensión es normal pueden tomar puertos diferentes, lo que me llama la atención es que si ya le dijiste que las VPN las tome como LocalNet el peer este tomando la IP "nateada"(aunque no me queda claro si esta extensión este usando una VPN o se conecte por fuera), suponiendo que usa una vpn esa extensión en su peer usa nat=no, si no usa una vpn y esta tomando la IP externa correctamente verifica que en efecto los puertos de RTP también los tengas abiertos y deja el nat=yes.

    También provee el sip debug de una llamada con ese problema.

    Saludos.

    Comentario

    • #3
      Re: Problema en escuchar algunas llamadas

      Hola que tal!

      Anexo el debug de mi extensión y una llamada con el problema:

      Código:
      srvTelIP:/etc/asterisk# asterisk -r
Asterisk 1.8.7.2, Copyright (C) 1999 - 2011 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/extconfig.conf':   == Found
  == Binding sipusers to mysql/asterisk/rt_sipfriends
  == Binding sippeers to mysql/asterisk/rt_sipfriends
  == Binding voicemail to mysql/asterisk/rt_voicemail
  == Binding extensions to mysql/asterisk/rt_extensions
Connected to Asterisk 1.8.7.2 currently running on srvTelIP (pid = 2920)
Verbosity is at least 3

<--- SIP read from UDP:192.168.0.162:5060 --->
BYE sip:1105@192.168.0.160:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.162:5060;branch=z9hG4bK-412ebc3c
From: <sip:1012@192.168.0.162>;tag=e3f6a80239c87422i0
To: "Cesar" <sip:1105@192.168.0.160>;tag=as56c75b50
Call-ID: 5d8b539a0237e8af0d1094c01b9f0233@192.168.0.160:5060
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.0.162:5060 (NAT)
Scheduling destruction of SIP dialog '5d8b539a0237e8af0d1094c01b9f0233@192.168.0.160:5060' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 192.168.0.162:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.162:5060;branch=z9hG4bK-412ebc3c;received=192.168.0.162;rport=5060
From: <sip:1012@192.168.0.162>;tag=e3f6a80239c87422i0
To: "Cesar" <sip:1105@192.168.0.160>;tag=as56c75b50
Call-ID: 5d8b539a0237e8af0d1094c01b9f0233@192.168.0.160:5060
CSeq: 101 BYE
Server: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (macro-llamada, s, 2) exited non-zero on 'SIP/1105-00000f63' in macro 'llamada'
  == Spawn extension (digitalife, 1012, 1) exited non-zero on 'SIP/1105-00000f63'
  == Using SIP RTP CoS mark 5
    -- Executing [1012@digitalife:1] Macro("SIP/1105-00000f65", "llamada,SIP/1012")
    -- Executing [s@macro-llamada:1] NoOp("SIP/1105-00000f65", "Llamada a SIP/1012 en proceso") in new stack
    -- Executing [s@macro-llamada:2] Dial("SIP/1105-00000f65", "SIP/1012") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.162:5060:
INVITE sip:1012@192.168.0.162:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK0e828219;rport
Max-Forwards: 70
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
To: <sip:1012@192.168.0.162:5060>
Contact: <sip:1105@192.168.0.160:5060>
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 102 INVITE
User-Agent: Digitalife PBX
Date: Mon, 18 Jun 2012 23:22:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 399

v=0
o=root 1390864722 1390864722 IN IP4 192.168.0.160
s=Asterisk PBX 1.8.7.2
c=IN IP4 192.168.0.160
t=0 0
m=audio 15812 RTP/AVP 18 0 8 112 111 3 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/1012

<--- SIP read from UDP:192.168.0.162:5060 --->
SIP/2.0 100 Trying
To: <sip:1012@192.168.0.162:5060>
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK0e828219
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.162:5060 --->
SIP/2.0 180 Ringing
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK0e828219
Contact: "Benjamin" <sip:1012@192.168.0.162:5060>
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- SIP/1012-00000f66 is ringing

<--- SIP read from UDP:192.168.0.162:5060 --->
SIP/2.0 200 OK
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK0e828219
Contact: "Benjamin" <sip:1012@192.168.0.162:5060>
Server: Cisco/SPA504G-7.4.9c
Content-Length: 238
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 112369523 112369523 IN IP4 192.168.0.162
s=-
c=IN IP4 192.168.0.162
t=0 0
m=audio 16460 RTP/AVP 18 101
a=rtpmap:18 G729a/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (12 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xd1e (gsm|ulaw|alaw|g726|g729|ilbc|g726aal2), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.162:16460
list_route: hop: <sip:1012@192.168.0.162:5060>
set_destination: Parsing <sip:1012@192.168.0.162:5060> for address/port to send to
set_destination: set destination to 192.168.0.162:5060
Transmitting (NAT) to 192.168.0.162:5060:
ACK sip:1012@192.168.0.162:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK0eb50de9;rport
Max-Forwards: 70
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
Contact: <sip:1105@192.168.0.160:5060>
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 102 ACK
User-Agent: Digitalife PBX
Content-Length: 0


---
    -- SIP/1012-00000f66 answered SIP/1105-00000f65
    -- Remotely bridging SIP/1105-00000f65 and SIP/1012-00000f66
set_destination: Parsing <sip:1012@192.168.0.162:5060> for address/port to send to
set_destination: set destination to 192.168.0.162:5060
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.162:5060:
INVITE sip:1012@192.168.0.162:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7b979ec6;rport
Max-Forwards: 70
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
Contact: <sip:1105@192.168.0.160:5060>
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 103 INVITE
User-Agent: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1390864722 1390864723 IN IP4 192.168.1.123
s=Asterisk PBX 1.8.7.2
c=IN IP4 192.168.1.123
t=0 0
m=audio 16466 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.0.162:5060 --->
SIP/2.0 200 OK
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7b979ec6
Contact: "Benjamin" <sip:1012@192.168.0.162:5060>
Server: Cisco/SPA504G-7.4.9c
Content-Length: 238
Content-Type: application/sdp

v=0
o=- 112369523 112369524 IN IP4 192.168.0.162
s=-
c=IN IP4 192.168.0.162
t=0 0
m=audio 16460 RTP/AVP 18 101
a=rtpmap:18 G729a/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (10 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xd1e (gsm|ulaw|alaw|g726|g729|ilbc|g726aal2), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.162:16460
set_destination: Parsing <sip:1012@192.168.0.162:5060> for address/port to send to
set_destination: set destination to 192.168.0.162:5060
Transmitting (NAT) to 192.168.0.162:5060:
ACK sip:1012@192.168.0.162:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK63621c54;rport
Max-Forwards: 70
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
Contact: <sip:1105@192.168.0.160:5060>
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 103 ACK
User-Agent: Digitalife PBX
Content-Length: 0


---
set_destination: Parsing <sip:1012@192.168.0.162:5060> for address/port to send to
set_destination: set destination to 192.168.0.162:5060
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.162:5060:
INVITE sip:1012@192.168.0.162:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK459f1aa4;rport
Max-Forwards: 70
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
Contact: <sip:1105@192.168.0.160:5060>
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 104 INVITE
User-Agent: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1390864722 1390864724 IN IP4 187.163.25.92
s=Asterisk PBX 1.8.7.2
c=IN IP4 187.163.25.92
t=0 0
m=audio 16466 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.0.162:5060 --->
SIP/2.0 200 OK
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 104 INVITE
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK459f1aa4
Contact: "Benjamin" <sip:1012@192.168.0.162:5060>
Server: Cisco/SPA504G-7.4.9c
Content-Length: 238
Content-Type: application/sdp

v=0
o=- 112369523 112369525 IN IP4 192.168.0.162
s=-
c=IN IP4 192.168.0.162
t=0 0
m=audio 16460 RTP/AVP 18 101
a=rtpmap:18 G729a/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (10 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xd1e (gsm|ulaw|alaw|g726|g729|ilbc|g726aal2), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.162:16460
set_destination: Parsing <sip:1012@192.168.0.162:5060> for address/port to send to
set_destination: set destination to 192.168.0.162:5060
Transmitting (NAT) to 192.168.0.162:5060:
ACK sip:1012@192.168.0.162:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK4bd04667;rport
Max-Forwards: 70
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
Contact: <sip:1105@192.168.0.160:5060>
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 104 ACK
User-Agent: Digitalife PBX
Content-Length: 0


---
set_destination: Parsing <sip:1012@192.168.0.162:5060> for address/port to send to
set_destination: set destination to 192.168.0.162:5060
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.162:5060:
INVITE sip:1012@192.168.0.162:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK21a10331;rport
Max-Forwards: 70
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
Contact: <sip:1105@192.168.0.160:5060>
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 105 INVITE
User-Agent: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1390864722 1390864725 IN IP4 192.168.1.123
s=Asterisk PBX 1.8.7.2
c=IN IP4 192.168.1.123
t=0 0
m=audio 16466 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.0.162:5060 --->
SIP/2.0 200 OK
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 105 INVITE
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK21a10331
Contact: "Benjamin" <sip:1012@192.168.0.162:5060>
Server: Cisco/SPA504G-7.4.9c
Content-Length: 238
Content-Type: application/sdp

v=0
o=- 112369523 112369526 IN IP4 192.168.0.162
s=-
c=IN IP4 192.168.0.162
t=0 0
m=audio 16460 RTP/AVP 18 101
a=rtpmap:18 G729a/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (10 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xd1e (gsm|ulaw|alaw|g726|g729|ilbc|g726aal2), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.162:16460
set_destination: Parsing <sip:1012@192.168.0.162:5060> for address/port to send to
set_destination: set destination to 192.168.0.162:5060
Transmitting (NAT) to 192.168.0.162:5060:
ACK sip:1012@192.168.0.162:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK0d21cbaf;rport
Max-Forwards: 70
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
Contact: <sip:1105@192.168.0.160:5060>
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 105 ACK
User-Agent: Digitalife PBX
Content-Length: 0


---
set_destination: Parsing <sip:1012@192.168.0.162:5060> for address/port to send to
set_destination: set destination to 192.168.0.162:5060
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.0.162:5060:
INVITE sip:1012@192.168.0.162:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK1d4cba98;rport
Max-Forwards: 70
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
Contact: <sip:1105@192.168.0.160:5060>
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 106 INVITE
User-Agent: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1390864722 1390864726 IN IP4 192.168.0.160
s=Asterisk PBX 1.8.7.2
c=IN IP4 192.168.0.160
t=0 0
m=audio 15812 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.0.162:5060 --->
SIP/2.0 200 OK
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 106 INVITE
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK1d4cba98
Contact: "Benjamin" <sip:1012@192.168.0.162:5060>
Server: Cisco/SPA504G-7.4.9c
Content-Length: 238
Content-Type: application/sdp

v=0
o=- 112369523 112369527 IN IP4 192.168.0.162
s=-
c=IN IP4 192.168.0.162
t=0 0
m=audio 16460 RTP/AVP 18 101
a=rtpmap:18 G729a/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (10 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xd1e (gsm|ulaw|alaw|g726|g729|ilbc|g726aal2), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.162:16460
set_destination: Parsing <sip:1012@192.168.0.162:5060> for address/port to send to
set_destination: set destination to 192.168.0.162:5060
Transmitting (NAT) to 192.168.0.162:5060:
ACK sip:1012@192.168.0.162:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK02941bea;rport
Max-Forwards: 70
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
Contact: <sip:1105@192.168.0.160:5060>
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 106 ACK
User-Agent: Digitalife PBX
Content-Length: 0


---
Scheduling destruction of SIP dialog '35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:1012@192.168.0.162:5060> for address/port to send to
set_destination: set destination to 192.168.0.162:5060
Reliably Transmitting (NAT) to 192.168.0.162:5060:
BYE sip:1012@192.168.0.162:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK241b5408;rport
Max-Forwards: 70
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 107 BYE
User-Agent: Digitalife PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (macro-llamada, s, 2) exited non-zero on 'SIP/1105-00000f65' in macro 'llamada'
  == Spawn extension (digitalife, 1012, 1) exited non-zero on 'SIP/1105-00000f65'

<--- SIP read from UDP:192.168.0.162:5060 --->
SIP/2.0 200 OK
To: <sip:1012@192.168.0.162:5060>;tag=faafe2e6fa3dc44i0
From: "Cesar" <sip:1105@192.168.0.160>;tag=as32bc4455
Call-ID: 35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060
CSeq: 107 BYE
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK241b5408
Server: Cisco/SPA504G-7.4.9c
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '35e696eb03d360b108ee2713271df8a7@192.168.0.160:5060' Method: INVITE
Really destroying SIP dialog '43a2fed7-b0dd7c95@192.168.0.162' Method: BYE
srvTelIP*CLI> exit
      En cuanto a lo del rtp, se tiene que hacer en el archivo rtp.conf? como podría hacer esa modificacion?

      Saludos.

      Comentario

      • #4
        Re: Problema en escuchar algunas llamadas

        La negociación SDP se repite mucho al final. ¿Si tienes licencias de G729? Sobre lo del RTP sí, en el rtp.conf tu defines el rango de los puertos que vas a usar y ese mismo es el que debes abrir en tu firewall si usas extensiones remotas, esos puertos es por donde viaja la voz.

        Comentario

        • #5
          Re: Problema en escuchar algunas llamadas

          Hola!

          Pues en sí tenía entendido que si eran telefonos CISCO no requería de comprar codec alguno por que esos teléfonos ya lo incluyen no se si estoy en lo correcto.

          Saludos.

          Comentario

          • #6
            Re: Problema en escuchar algunas llamadas

            Así es, la mayoría de los teléfonos(si no es que todos) ya incluyen el codec g729, pero me refería a la parte de tu servidor.

            Comentario

            • #7
              Re: Problema en escuchar algunas llamadas

              Pues instalé un codec precompilado

              Comentario

              • #8
                Re: Problema en escuchar algunas llamadas

                El comando "core show translation" te muestra que SI tienes G729 disponible?
                dCAP Christian Cabrera R.
                Para aprender a usar Asterisk, asiste a uno de mis cursos Asterisk
                Si deseas asesoría pagada, por favor contáctame

                Comentario

                • #9
                  Re: Problema en escuchar algunas llamadas

                  pregunta estas agregando la red de la vpn dentro del parámetro localnet en el sip.conf ?

                  sldss

                  Comentario

                  • #10
                    Re: Problema en escuchar algunas llamadas

                    Hola que tal!

                    El core show translation me muestra:

                    Código:
                        g723   gsm  ulaw  alaw g726aal2 adpcm  slin lpc10  g729 speex  ilbc  g726  g722 siren7 siren14 slin16  g719 speex16 testlaw
     g723     -  4002     2     2     4002     2     1     2  4001  8001     -     2     2      -       -   4003     -    8004    4001
      gsm  8001     -  4001  4001     8001  4001  4000  4001  8000 12000     -  4001  4001      -       -   8002     -   12003    8000
     ulaw  4002  4002     -     1     4002     2     1     2  4001  8001     -     2     2      -       -   4003     -    8004    4001
     alaw  4002  4002     1     -     4002     2     1     2  4001  8001     -     2     2      -       -   4003     -    8004    4001
 g726aal2  4002  4002     2     2        -     2     1     2  4001  8001     -     2     2      -       -   4003     -    8004    4001
    adpcm  4002  4002     2     2     4002     -     1     2  4001  8001     -     2     2      -       -   4003     -    8004    4001
     slin  4001  4001     1     1     4001     1     -     1  4000  8000     -     1     1      -       -   4002     -    8003    4000
    lpc10  8001  8001  4001  4001     8001  4001  4000     -  8000 12000     -  4001  4001      -       -   8002     -   12003    8000
     g729  4002  4002     2     2     4002     2     1     2     -  8001     -     2     2      -       -   4003     -    8004    4001
    speex  4002  4002     2     2     4002     2     1     2  4001     -     -     2     2      -       -   4003     -    8004    4001
     ilbc     -     -     -     -        -     -     -     -     -     -     -     -     -      -       -      -     -       -       -
     g726  8001  8001  4001  4001     8001  4001  4000  4001  8000 12000     -     -  4001      -       -   8002     -   12003    8000
     g722  8001  8001  4001  4001     8001  4001  4000  4001  8000 12000     -  4001     -      -       -   4001     -    8002    8000
   siren7     -     -     -     -        -     -     -     -     -     -     -     -     -      -       -      -     -       -       -
  siren14     -     -     -     -        -     -     -     -     -     -     -     -     -      -       -      -     -       -       -
   slin16  8002  8002  4002  4002     8002  4002  4001  4002  8001 12001     -  4002     1      -       -      -     -    4001    8001
     g719     -     -     -     -        -     -     -     -     -     -     -     -     -      -       -      -     -       -       -
  speex16 12002 12002  8002  8002    12002  8002  8001  8002 12001 16001     -  8002  4001      -       -   4000     -       -   12001
  testlaw  4002  4002     2     2     4002     2     1     2  4001  8001     -     2     2      -       -   4003     -    8004       -
                    Parece ser que si esta instalado.

                    En cuanto al problema solucioné una parte forzando el puerto en el sip.conf y conectando los telefonos a la dirección IP local del servidor, en lugar del dyndns. El problema que presento ahora es con un solo grupo de teléfonos, ese grupo requiero que se conecte desde cualquier otro sitio que no este dentro de la VPN (aeropuertos, hoteles, etc.) El problema es que al tratar de jalar con la IP fija que tenemos o con el dyndns, de un lado se escucha y del otro lado nada. Anexo log de uno de los telefonos con el problema.

                    Código:
                    srvTelIP*CLI> sip set debug peer 1105
SIP Debugging Enabled for IP: 187.163.25.92

<--- SIP read from UDP:187.163.25.92:1125 --->
INVITE sip:1002@187.177.181.224 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK-a7d08bc3
From: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
To: "Edith" <sip:1002@187.177.181.224>
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Cesar" <sip:1105@192.168.1.123:5060>
Expires: 240
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 395
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 15153 15153 IN IP4 192.168.1.123
s=-
c=IN IP4 192.168.1.123
t=0 0
m=audio 16450 RTP/AVP 18 0 2 8 9 96 97 98 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (14 headers 18 lines) ---
Sending to 187.163.25.92:1125 (NAT)
Using INVITE request as basis request - da10c4a9-cd52c91b@192.168.1.123
Found peer '1105' for '1105' from 187.163.25.92:1125

<--- Reliably Transmitting (NAT) to 187.163.25.92:1125 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK-a7d08bc3;received=187.163.25.92;rport=1125
From: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
To: "Edith" <sip:1002@187.177.181.224>;tag=as74af5f97
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 101 INVITE
Server: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2596f17f"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'da10c4a9-cd52c91b@192.168.1.123' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:187.163.25.92:1125 --->
ACK sip:1002@187.177.181.224 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK-a7d08bc3
From: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
To: "Edith" <sip:1002@187.177.181.224>;tag=as74af5f97
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Cesar" <sip:1105@192.168.1.123:5060>
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 0

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

<--- SIP read from UDP:187.163.25.92:1125 --->
INVITE sip:1002@187.177.181.224 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK-6067f63
From: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
To: "Edith" <sip:1002@187.177.181.224>
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="1105",realm="asterisk",nonce="2596f17f",uri="sip:1002@187.177.181.224",algorithm=MD5,response="e3ad20d87ec50a2d9087d1ee9849fa8c"
Contact: "Cesar" <sip:1105@192.168.1.123:5060>
Expires: 240
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 395
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces
Content-Type: application/sdp

v=0
o=- 15153 15153 IN IP4 192.168.1.123
s=-
c=IN IP4 192.168.1.123
t=0 0
m=audio 16450 RTP/AVP 18 0 2 8 9 96 97 98 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (15 headers 18 lines) ---
Sending to 187.163.25.92:1125 (NAT)
Using INVITE request as basis request - da10c4a9-cd52c91b@192.168.1.123
Found peer '1105' for '1105' from 187.163.25.92:1125
  == Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found unknown media description format G726-40 for ID 96
Found unknown media description format G726-24 for ID 97
Found unknown media description format G726-16 for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - 0xd1e (gsm|ulaw|alaw|g726|g729|ilbc|g726aal2), peer - audio=0x190c (ulaw|alaw|g726|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x90c (ulaw|alaw|g726|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.123:16450
Looking for 1002 in digitalife (domain 187.177.181.224)
list_route: hop: <sip:1105@192.168.1.123:5060>

<--- Transmitting (NAT) to 187.163.25.92:1125 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK-6067f63;received=187.163.25.92;rport=1125
From: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
To: "Edith" <sip:1002@187.177.181.224>
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 102 INVITE
Server: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1002@187.177.181.224:5060>
Content-Length: 0


<------------>
    -- Executing [1002@digitalife:1] Macro("SIP/1105-00000677", "llamada,SIP/1002")
    -- Executing [s@macro-llamada:1] NoOp("SIP/1105-00000677", "Llamada a SIP/1002 en proceso") in new stack
    -- Executing [s@macro-llamada:2] Dial("SIP/1105-00000677", "SIP/1002") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/1002
    -- SIP/1002-00000678 is ringing

<--- Transmitting (NAT) to 187.163.25.92:1125 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK-6067f63;received=187.163.25.92;rport=1125
From: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
To: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 102 INVITE
Server: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1002@187.177.181.224:5060>
Content-Length: 0


<------------>
    -- SIP/1002-00000678 answered SIP/1105-00000677
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 187.163.25.92:1125 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK-6067f63;received=187.163.25.92;rport=1125
From: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
To: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 102 INVITE
Server: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:1002@187.177.181.224:5060>
Content-Type: application/sdp
Content-Length: 340

v=0
o=root 1819468860 1819468860 IN IP4 187.177.181.224
s=Asterisk PBX 1.8.7.2
c=IN IP4 187.177.181.224
t=0 0
m=audio 13556 RTP/AVP 18 0 8 2 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
    -- Remotely bridging SIP/1105-00000677 and SIP/1002-00000678

<--- SIP read from UDP:187.163.25.92:1125 --->
ACK sip:1002@187.177.181.224:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.123:5060;branch=z9hG4bK-60f4c054
From: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
To: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="1105",realm="asterisk",nonce="2596f17f",uri="sip:1002@187.177.181.224",algorithm=MD5,response="e3ad20d87ec50a2d9087d1ee9849fa8c"
Contact: "Cesar" <sip:1105@192.168.1.123:5060>
User-Agent: Cisco/SPA303-7.4.9a
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:1105@192.168.1.123:5060> for address/port to send to
set_destination: set destination to 192.168.1.123:5060
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 187.163.25.92:1125:
INVITE sip:1105@192.168.1.123:5060 SIP/2.0
Via: SIP/2.0/UDP 187.177.181.224:5060;branch=z9hG4bK4b0da378;rport
Max-Forwards: 70
From: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
To: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
Contact: <sip:1002@187.177.181.224:5060>
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 102 INVITE
User-Agent: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1819468860 1819468861 IN IP4 192.168.0.164
s=Asterisk PBX 1.8.7.2
c=IN IP4 192.168.0.164
t=0 0
m=audio 16526 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:187.163.25.92:1125 --->
SIP/2.0 200 OK
To: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
From: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 102 INVITE
Via: SIP/2.0/UDP 187.177.181.224:5060;branch=z9hG4bK4b0da378
Contact: "Cesar" <sip:1105@192.168.1.123:5060>
Server: Cisco/SPA303-7.4.9a
Content-Length: 230
Content-Type: application/sdp

v=0
o=- 15153 15154 IN IP4 192.168.1.123
s=-
c=IN IP4 192.168.1.123
t=0 0
m=audio 16450 RTP/AVP 18 101
a=rtpmap:18 G729a/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (10 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xd1e (gsm|ulaw|alaw|g726|g729|ilbc|g726aal2), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.123:16450
set_destination: Parsing <sip:1105@192.168.1.123:5060> for address/port to send to
set_destination: set destination to 192.168.1.123:5060
Transmitting (NAT) to 187.163.25.92:1125:
ACK sip:1105@192.168.1.123:5060 SIP/2.0
Via: SIP/2.0/UDP 187.177.181.224:5060;branch=z9hG4bK3d11be85;rport
Max-Forwards: 70
From: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
To: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
Contact: <sip:1002@187.177.181.224:5060>
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 102 ACK
User-Agent: Digitalife PBX
Content-Length: 0


---
set_destination: Parsing <sip:1105@192.168.1.123:5060> for address/port to send to
set_destination: set destination to 192.168.1.123:5060
Audio is at 5060
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 187.163.25.92:1125:
INVITE sip:1105@192.168.1.123:5060 SIP/2.0
Via: SIP/2.0/UDP 187.177.181.224:5060;branch=z9hG4bK5a557150;rport
Max-Forwards: 70
From: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
To: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
Contact: <sip:1002@187.177.181.224:5060>
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 103 INVITE
User-Agent: Digitalife PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1819468860 1819468862 IN IP4 187.177.181.224
s=Asterisk PBX 1.8.7.2
c=IN IP4 187.177.181.224
t=0 0
m=audio 13556 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:187.163.25.92:1125 --->
SIP/2.0 200 OK
To: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
From: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 103 INVITE
Via: SIP/2.0/UDP 187.177.181.224:5060;branch=z9hG4bK5a557150
Contact: "Cesar" <sip:1105@192.168.1.123:5060>
Server: Cisco/SPA303-7.4.9a
Content-Length: 230
Content-Type: application/sdp

v=0
o=- 15153 15155 IN IP4 192.168.1.123
s=-
c=IN IP4 192.168.1.123
t=0 0
m=audio 16450 RTP/AVP 18 101
a=rtpmap:18 G729a/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (10 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729a for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xd1e (gsm|ulaw|alaw|g726|g729|ilbc|g726aal2), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.123:16450
set_destination: Parsing <sip:1105@192.168.1.123:5060> for address/port to send to
set_destination: set destination to 192.168.1.123:5060
Transmitting (NAT) to 187.163.25.92:1125:
ACK sip:1105@192.168.1.123:5060 SIP/2.0
Via: SIP/2.0/UDP 187.177.181.224:5060;branch=z9hG4bK7672fb2b;rport
Max-Forwards: 70
From: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
To: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
Contact: <sip:1002@187.177.181.224:5060>
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 103 ACK
User-Agent: Digitalife PBX
Content-Length: 0


---
  == Spawn extension (macro-llamada, s, 2) exited non-zero on 'SIP/1105-00000677' in macro 'llamada'
  == Spawn extension (digitalife, 1002, 1) exited non-zero on 'SIP/1105-00000677'
Scheduling destruction of SIP dialog 'da10c4a9-cd52c91b@192.168.1.123' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:1105@192.168.1.123:5060> for address/port to send to
set_destination: set destination to 192.168.1.123:5060
Reliably Transmitting (NAT) to 187.163.25.92:1125:
BYE sip:1105@192.168.1.123:5060 SIP/2.0
Via: SIP/2.0/UDP 187.177.181.224:5060;branch=z9hG4bK79b11927;rport
Max-Forwards: 70
From: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
To: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 104 BYE
User-Agent: Digitalife PBX
Proxy-Authorization: Digest username="1105", realm="asterisk", algorithm=MD5, uri="sip:187.177.181.224", nonce="", response="b002c53fdd7a2ba4780d2268a542432c"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:187.163.25.92:1125 --->
SIP/2.0 200 OK
To: "Cesar" <sip:1105@187.177.181.224>;tag=f3c23f51c65b4e93o0
From: "Edith" <sip:1002@187.177.181.224>;tag=as614836f4
Call-ID: da10c4a9-cd52c91b@192.168.1.123
CSeq: 104 BYE
Via: SIP/2.0/UDP 187.177.181.224:5060;branch=z9hG4bK79b11927
Server: Cisco/SPA303-7.4.9a
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'da10c4a9-cd52c91b@192.168.1.123' Method: ACK
                    No se como se podría agregar la red VPN? No sirve unicamente agregar las subredes?

                    Saludos y gracias .

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