Llamada x LAN (sip.conf / extensions.conf) sms=> No matching peer found

[andrewtwo]

Hola ahora tengo otro problemita estoy tratando de hacer una llamada de prueba en un ambiente wmware, estoy usando como cliente el linphone, ajunto captura de la configuración del cliente, ambos maquinas estan en red y responden a los ping en ambos sentidos aunque los teléfonos se registran correctamente en el asterisk , no logro conectarme del tel1(ext 1001) al tel2(ext 1001) o del tel2 al tel1
Name/username Host Dyn Forcerport ACL Port Status
tel1/tel1 192.168.74.135 D N 5060 OK (29 ms)
tel2/tel2 192.168.74.137 D N 5060 OK (24 ms)

el mensaje es el siguiente:
[Jan 2 18:47:12] NOTICE[2310]: chan_sip.c:25494 handle_request_register: Registration from '<sip:1002@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found


configuración del SIP.CONF
[general]
context=default ;contexto por defecto para las llamadas entrantes
bindport=5060 ;puerto para conexiones SIP
bindaddr=0.0.0.0
srvlookp=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
language=es

;Datos para registro

[tel1]
secret=123
context=users
host=dynamic
type=friend
qualify=yes
nat=yes
dtmfmode=rfc2833

[tel2]
secret=123
context=users
host=dynamic
type=friend
qualify=yes
nat=yes
dtmfmode=rfc2833

configuración EXTENSIONS.CONF
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]

[users]

extem => 1001,1,Dial(SIP/tel1,20)
extem => 1002,1,Dial(SIP/tel2,20)

De antemano te agradezco cualquier ayuda

[Christian Cabrera]

El error que te aparece es de registro. Sin embargo, tus extensiones están registradas, así que ese error es por otra cosa.

Si la configuración que pones está copiada tal cual, el error es que usaste extem en lugar de exten dentro del extensions.conf. Haz el cambio, recarga el plan de llamadas, y péganos la salida del CLI al momento en que marcas para ver si te sigue ocasionando algún problema.

Saludos,

[andrewtwo]

Hola ya corregie l extem por exten pero aun asi sigue el problema, cuando marco desde la linea tel1 el softphone hace el proceso de llamado a la linea tel2 y genera el mensaje calling ... hasta que la llamada se cuelga porque nadie responde. Ahora cuando marco desde la linea tel2 a la tel1 me genera de forma inmediata call ended no hace el proceso de llamado como como el tel2. y si juego cambiando los parametros el softphone como colando la extensión y no el usuario no funciona tampoco

Estos fueron los últimos intentos que hice, lo que me pareció raro es que a veces genera registros con configuración como <sip:te2@192.168.74.136> donde anteriormente me habia equivocado pero que en los archivos figura correctamente configurado.
ya no se me ocurre que mas probar, yo lo estoy usando dentro de un wmware con nat=yes aunque no creo que eso interfiera en una llamad x lan

gracias por la ayuda

[Jan 4 00:38:15] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<sip:1002@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found
[Jan 4 00:38:21] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<sip:te2@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found
[Jan 4 00:40:16] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<sip:1002@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found
[Jan 4 00:40:22] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<sip:te2@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found
[Jan 4 00:42:17] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<sip:1002@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found
[Jan 4 00:42:23] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<sip:te2@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found
-- Unregistered SIP 'tel1'
[Jan 4 00:43:33] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<SIP:1001@192.168.74.136>' failed for '192.168.74.135:5060' - No matching peer found
[Jan 4 00:43:33] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<SIP:1001@192.168.74.136>' failed for '192.168.74.135:5060' - No matching peer found
[Jan 4 00:43:40] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<SIP:1001@192.168.74.136>' failed for '192.168.74.135:5060' - No matching peer found
[Jan 4 00:43:45] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<SIP:1001@192.168.74.136>' failed for '192.168.74.135:5060' - No matching peer found
[Jan 4 00:43:50] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<SIP:1001@192.168.74.136>' failed for '192.168.74.135:5060' - No matching peer found
-- Registered SIP 'tel1' at 192.168.74.135:5060
> Saved useragent "Linphone/3.5.2 (eXosip2/3.6.0)" for peer tel1
[Jan 4 00:43:57] NOTICE[3067]: chan_sip.c:21329 handle_response_peerpoke: Peer 'tel1' is now Reachable. (30ms / 2000ms)
[Jan 4 00:44:18] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<sip:1002@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found
[Jan 4 00:44:24] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<sip:te2@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found
-- Registered SIP 'tel2' at 192.168.74.137:5060
> Saved useragent "Linphone/3.3.2 (eXosip2/3.3.0)" for peer tel2
[Jan 4 00:44:46] NOTICE[3067]: chan_sip.c:21329 handle_response_peerpoke: Peer 'tel2' is now Reachable. (29ms / 2000ms)
[Jan 4 00:46:19] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<sip:1002@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found
[Jan 4 00:46:25] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<sip:te2@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found
[Jan 4 00:48:20] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<sip:1002@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found
[Jan 4 00:48:26] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<sip:te2@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found
[Jan 4 00:50:21] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<sip:1002@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found
[Jan 4 00:50:27] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<sip:te2@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found
-- Unregistered SIP 'tel1'
-- Unregistered SIP 'tel2'
-- Registered SIP 'tel1' at 192.168.74.135:5060
[Jan 4 00:51:08] NOTICE[3067]: chan_sip.c:21329 handle_response_peerpoke: Peer 'tel1' is now Reachable. (29ms / 2000ms)
-- Registered SIP 'tel2' at 192.168.74.137:5060
[Jan 4 00:52:06] NOTICE[3067]: chan_sip.c:21329 handle_response_peerpoke: Peer 'tel2' is now Reachable. (35ms / 2000ms)

[Christian Cabrera]

No tienes el verbose activado, por lo que solo se ven los mensajes de ERROR/WARNING/NOTICE pero no ves los de verbose. Ejecuta core set verbose 3 en el *CLI para que aparezca el detalle de los comandos y ponnos esa salida.

Sobre los errores que pegas:

[Jan 4 00:38:15] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<sip:1002@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found

Significa que el teléfono con la IP 192.168.74.137 tiene configurado en su campo de username la cadena 1002, en vez de decir tel2.

El error de:

[Jan 4 00:38:21] NOTICE[3067]: chan_sip.c:25494 handle_request_register: Registration from '<sip:te2@192.168.74.136>' failed for '192.168.74.137:5060' - No matching peer found

Significa que desde la IP 192.168.74.137 tienes configurado te2 como username, en lugar de tel2.

Dado que ambos aparecen provenir de la misma IP, es de mi parecer que tienes 2 teléfonos configurados en el mismo equipo, y cada uno con sus propios problemas. Debes corregirlos para que te dejen de aparecer pero de igual manera, no estás viendo el verbose así que no sé si esa sea o no la causa de todo el problema que comentaste inicialmente.

Saludos,

[andrewtwo]

Hola el verbose no se me activa con el comando core set verbose 3 ni con core set verbose 50 no muestra nada y me asegure de que estuviera activo con core set verbose 1
Sobre los errores que pegas hice las pruebas de varias formas con la extension y con el username pero pese a que lo tenia ya corregido me continuaba imprimiendo el verbose con datos viejos a pesar de estar actualizados
al final me genero algo de verbose solo estas lineas

[Jan 10 14:12:50] WARNING[2841]: acl.c:719 ast_ouraddrfor: Cannot connect
[Jan 10 14:12:50] WARNING[2841]: chan_sip.c:3681 __sip_xmit: sip_xmit of 0xa584150 (len 494) to 169.254.5.138:5060 returned -2: Network is unreachable
> Saved useragent "Linphone/3.3.2 (eXosip2/3.3.0)" for peer tel2
> Saved useragent "Linphone/3.5.2 (eXosip2/3.6.0)" for peer tel1

le active el debug: core set debug on y me genera lo siguiente:
gracias por la ayuda

<--- SIP read from UDP:192.168.74.137:5060 --->
jaK
<------------->
Reliably Transmitting (NAT) to 192.168.74.137:5060:
OPTIONS sip:tel2@192.168.74.137:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.74.136:5060;branch=z9hG4bK5d5cb4a5;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.74.136>;tag=as2057f9be
To: <sip:tel2@192.168.74.137:5060>
Contact: <sip:asterisk@192.168.74.136:5060>
Call-ID: 761cebb21aa9bf695781647f7ba870c2@192.168.74.136:50 60
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.19.0
Date: Thu, 10 Jan 2013 19:36:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.74.137:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.74.136:5060;branch=z9hG4bK5d5cb4a5;rport=5 060
From: "asterisk" <sip:asterisk@192.168.74.136>;tag=as2057f9be
To: <sip:tel2@192.168.74.137:5060>;tag=2114475383
Call-ID: 761cebb21aa9bf695781647f7ba870c2@192.168.74.136:50 60
CSeq: 102 OPTIONS
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '761cebb21aa9bf695781647f7ba870c2@192.168.74.136:5 060' Method: OPTIONS

<--- SIP read from UDP:192.168.74.135:5060 --->
jaK
<------------->
Reliably Transmitting (NAT) to 192.168.74.135:5060:
OPTIONS SIP:tel1@192.168.74.135;line=196cd3895a8d45c SIP/2.0
Via: SIP/2.0/UDP 192.168.74.136:5060;branch=z9hG4bK0407adff;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.74.136>;tag=as0409805a
To: <SIP:tel1@192.168.74.135;line=196cd3895a8d45c>
Contact: <sip:asterisk@192.168.74.136:5060>
Call-ID: 68a3ac2b014887e554ebc2425857cd53@192.168.74.136:50 60
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.19.0
Date: Thu, 10 Jan 2013 19:37:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.74.135:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.74.136:5060;branch=z9hG4bK0407adff;rport=5 060
From: "asterisk" <sip:asterisk@192.168.74.136>;tag=as0409805a
To: <SIP:tel1@192.168.74.135;line=196cd3895a8d45c>;tag =1505841845
Call-ID: 68a3ac2b014887e554ebc2425857cd53@192.168.74.136:50 60
CSeq: 102 OPTIONS
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '68a3ac2b014887e554ebc2425857cd53@192.168.74.136:5 060' Method: OPTIONS

<--- SIP read from UDP:192.168.74.137:5060 --->
jaK
<------------->

<--- SIP read from UDP:192.168.74.135:5060 --->
jaK
<------------->

<--- SIP read from UDP:192.168.74.137:5060 --->
jaK
<------------->

<--- SIP read from UDP:192.168.74.135:5060 --->
jaK
<------------->

<--- SIP read from UDP:192.168.74.137:5060 --->
jaK
<------------->

<--- SIP read from UDP:192.168.74.135:5060 --->
jaK
<------------->

<--- SIP read from UDP:192.168.74.137:5060 --->
jaK
<------------->

<--- SIP read from UDP:192.168.74.135:5060 --->
jaK
<------------->

<--- SIP read from UDP:192.168.74.137:5060 --->
jaK
<------------->
Reliably Transmitting (NAT) to 192.168.74.137:5060:
OPTIONS sip:tel2@192.168.74.137:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.74.136:5060;branch=z9hG4bK50e7c435;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.74.136>;tag=as63dad097
To: <sip:tel2@192.168.74.137:5060>
Contact: <sip:asterisk@192.168.74.136:5060>
Call-ID: 68dc499f6845d3d36591e00a3a92a068@192.168.74.136:50 60
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.19.0
Date: Thu, 10 Jan 2013 19:37:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.74.137:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.74.136:5060;branch=z9hG4bK50e7c435;rport=5 060
From: "asterisk" <sip:asterisk@192.168.74.136>;tag=as63dad097
To: <sip:tel2@192.168.74.137:5060>;tag=631179295
Call-ID: 68dc499f6845d3d36591e00a3a92a068@192.168.74.136:50 60
CSeq: 102 OPTIONS
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '68dc499f6845d3d36591e00a3a92a068@192.168.74.136:5 060' Method: OPTIONS

<--- SIP read from UDP:192.168.74.135:5060 --->
jaK
<------------->
Reliably Transmitting (NAT) to 192.168.74.135:5060:
OPTIONS SIP:tel1@192.168.74.135;line=196cd3895a8d45c SIP/2.0
Via: SIP/2.0/UDP 192.168.74.136:5060;branch=z9hG4bK2d6917dd;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.74.136>;tag=as2ad8fa0e
To: <SIP:tel1@192.168.74.135;line=196cd3895a8d45c>
Contact: <sip:asterisk@192.168.74.136:5060>
Call-ID: 0e90c1b849403e1f5b0ba2860e3ccd28@192.168.74.136:50 60
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.19.0
Date: Thu, 10 Jan 2013 19:38:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.74.135:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.74.136:5060;branch=z9hG4bK2d6917dd;rport=5 060
From: "asterisk" <sip:asterisk@192.168.74.136>;tag=as2ad8fa0e
To: <SIP:tel1@192.168.74.135;line=196cd3895a8d45c>;tag =768459130
Call-ID: 0e90c1b849403e1f5b0ba2860e3ccd28@192.168.74.136:50 60
CSeq: 102 OPTIONS
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, SUBSCRIBE, NOTIFY, INFO
Accept: application/sdp
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '0e90c1b849403e1f5b0ba2860e3ccd28@192.168.74.136:5 060' Method: OPTIONS

<--- SIP read from UDP:192.168.74.137:5060 --->
jaK
<------------->

<--- SIP read from UDP:192.168.74.135:5060 --->
jaK
<------------->

<--- SIP read from UDP:192.168.74.137:5060 --->
jaK
<------------->

<--- SIP read from UDP:192.168.74.135:5060 --->
jaK
<------------->

<--- SIP read from UDP:192.168.74.137:5060 --->
jaK
<------------->