Configuracion llamadas entrantes /salientes SIP

pedrer

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#1
Hola chicos, despues de tanto batallar con elastix a decididio reinstalarle y ahora no consigo configurarle para que me funcione correctamente. Tengo un proveedor SIP (voztelecom). Elastix se me registra bien

Tambien en la opcion TRUNK SIp tengo configurado el peer de salientes [voztelecom] y el user context para las entrantes [from-voztelecom] de manera que si hago sip show peers me salen los dos peers establecidos con el operador.

Por otro lado creo las rutas entranets y salientews normales y creo uan extension...y resulta que no puedo hacer ninguna llamada, ni entrante nio saliente.


por favor necesito ayuda urgente.

gracias
pedrer
 

pedrer

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#2
Esta es la traza para una llamada entrante

Scheduling destruction of SIP dialog 'eb422da7-89e6aba1-0d287@193.22.119.20' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog 'eb422da7-45b6aba1-2b287@193.22.119.20' Method: OPTIONS
elastix*CLI>
<--- SIP read from 193.22.119.20:5060 --->
INVITE sip:34911820519@192.168.0.29 SIP/2.0
Record-Route: <sip:193.22.119.20;lr=on;ftag=9BE80B60-2446>
Record-Route: <sip:193.22.119.20;lr=on;ftag=9BE80B60-2446>
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bKa442.b640d507.0
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bKa442.a640d507.0
Via: SIP/2.0/UDP 193.22.119.35:5060;branch=z9hG4bK94880B379
From: <sip:685868409@193.22.119.35>;tag=9BE80B60-2446
To: <sip:911898238@193.22.119.20>
Call-ID: FA40DBF7-DDD011DE-BC80898E-F04A96@193.22.119.35
Min-SE: 1800
CSeq: 101 INVITE
Max-Forwards: 31
Remote-Party-ID: <sip:685868409@193.22.119.35>;party=calling;screen=yes;privacy=off
Contact: <sip:685868409@193.22.119.35:5060;nat=yes>
Expires: 180
Content-Type: application/sdp
Content-Length: 582
P-RTP-Proxy: YES

v=0
o=CiscoSystemsSIP-GW-UserAgent 775 7426 IN IP4 193.22.119.35
s=SIP Call
c=IN IP4 193.22.119.14
t=0 0
m=audio 28956 RTP/AVP 18 4 8 0 100 101
c=IN IP4 193.22.119.14
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3;annexa=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194,200-202
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 192-194,200-202
a=X-cap: 2 image udptl t38
a=nortpproxy:yes

<------------->
--- (18 headers 23 lines) ---
Sending to 193.22.119.20 : 5060 (no NAT)
Using INVITE request as basis request - FA40DBF7-DDD011DE-BC80898E-F04A96@193.22.119.35
Found peer 'voztelecom'
elastix*CLI>
<--- Reliably Transmitting (no NAT) to 193.22.119.20:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bKa442.b640d507.0;received=193.22.119.20
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bKa442.a640d507.0
Via: SIP/2.0/UDP 193.22.119.35:5060;branch=z9hG4bK94880B379
From: <sip:685868409@193.22.119.35>;tag=9BE80B60-2446
To: <sip:911898238@193.22.119.20>;tag=as07c9c066
Call-ID: FA40DBF7-DDD011DE-BC80898E-F04A96@193.22.119.35
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4f074535"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'FA40DBF7-DDD011DE-BC80898E-F04A96@193.22.119.35' in 32000 ms (Method: INVITE)
elastix*CLI>
<--- SIP read from 193.22.119.20:5060 --->
ACK sip:34911820519@192.168.0.29 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bKa442.b640d507.0
From: <sip:685868409@193.22.119.35>;tag=9BE80B60-2446
Call-ID: FA40DBF7-DDD011DE-BC80898E-F04A96@193.22.119.35
To: <sip:911898238@193.22.119.20>;tag=as07c9c066
CSeq: 101 ACK
User-Agent: OpenSER (1.2.1-notls (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
elastix*CLI>
<--- SIP read from 192.168.0.10:42372 --->



<------------->
Really destroying SIP dialog '5d06ed741caccdbe018142fb2dfbf6e3@127.0.0.1' Method: REGISTER
elastix*CLI> exit
[root@elastix log]# Sending to 193.22.119.20 : 5060 (no NAT)
-bash: syntax error near unexpected token `('
[root@elastix log]# Using INVITE request as basis request - FA40DBF7-DDD011DE-BC80898E-F04A96@193.22.119.35
-bash: Using: command not found
Via: SIP/2.0/UDP 193.22.119.35:5060;branch=z9hG4bK94880B379
[root@elastix log]# Found peer 'voztelecom'
-bash: Found: command not found
To: <sip:911898238@193.22.119.20>;tag=as07c9c066
[root@elastix log]# elastix*CLI>
Call-ID: FA40DBF7-DDD011DE-BC80898E-F04A96@193.22.119.35
-bash: syntax error near unexpected token `newline'
[root@elastix log]# <--- Reliably Transmitting (no NAT) to 193.22.119.20:5060 --->
-bash: syntax error near unexpected token `('
[root@elastix log]# SIP/2.0 407 Proxy Authentication Required
-bash: SIP/2.0: No existe el fichero o el directorio
[root@elastix log]# Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bKa442.b640d507.0;received=193.22.119.20
-bash: Via:: command not found
[root@elastix log]# Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bKa442.a640d507.0
-bash: Via:: command not found
[root@elastix log]# Via: SIP/2.0/UDP 193.22.119.35:5060;branch=z9hG4bK94880B379
-bash: Via:: command not found
[root@elastix log]# From: <sip:685868409@193.22.119.35>;tag=9BE80B60-2446
-bash: syntax error near unexpected token `;'
[root@elastix log]# To: <sip:911898238@193.22.119.20>;tag=as07c9c066
-bash: syntax error near unexpected token `;'
[root@elastix log]# Call-ID: FA40DBF7-DDD011DE-BC80898E-F04A96@193.22.119.35


-bash: Call-ID:: command not found
[root@elastix log]# CSeq: 101 INVITE
-bash: CSeq:: command not found
[root@elastix log]# User-Agent: Asterisk PBX
-bash: User-Agent:: command not found
[root@elastix log]# Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
-bash: Allow:: command not found
[root@elastix log]# Supported: replaces
-bash: Supported:: command not found
[root@elastix log]# Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4f074535"
-bash: Proxy-Authenticate:: command not found
[root@elastix log]# Content-Length: 0
-bash: Content-Length:: command not found
[root@elastix log]#
[root@elastix log]#
[root@elastix log]# <------------>
-bash: syntax error near unexpected token `newline'
[root@elastix log]# Scheduling destruction of SIP dialog 'FA40DBF7-DDD011DE-BC80898E-F04A96@193.22.119.35' in 32000 ms (Method: INVITE)
-bash: syntax error near unexpected token `('
[root@elastix log]# elastix*CLI>
-bash: syntax error near unexpected token `newline'
[root@elastix log]# <--- SIP read from 193.22.119.20:5060 --->
-bash: syntax error near unexpected token `newline'
[root@elastix log]# ACK sip:34911820519@192.168.0.29 SIP/2.0
-bash: ACK: command not found
[root@elastix log]# Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bKa442.b640d507.0
-bash: Via:: command not found
[root@elastix log]# From: <sip:685868409@193.22.119.35>;tag=9BE80B60-2446
-bash: syntax error near unexpected token `;'
[root@elastix log]# Call-ID: FA40DBF7-DDD011DE-BC80898E-F04A96@193.22.119.35
-bash: Call-ID:: command not found
[root@elastix log]# To: <sip:911898238@193.22.119.20>;tag=as07c9c066
-bash: syntax error near unexpected token `;'
[root@elastix log]# CSeq: 101 ACK
-bash: CSeq:: command not found
[root@elastix log]# User-Agent: OpenSER (1.2.1-notls (i386/linux))
-bash: syntax error near unexpected token `('
[root@elastix log]# Content-Length: 0
-bash: Content-Length:: command not found
 

pedrer

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#3
Hola a todos... despues de tanto buscar.. al final en la configuracion del trunk solamente configure el principal y el users lo deje en blanco y me funcionan las entrantes pero las salientes no me funcionan cdo intento llamar me dice una locucion que todas las lineas estan ocupadas.....


Some helpç'
Thanks

Ariel
 

arusnet

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Sep 11, 2008
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#4
Normalmente cuando suelta ese mesaje es que esta mal configurado el dialplan en las llamadas salientes, es decir, no escuentra la trocal para sacar la llamada.
Para hacer las pruebas te recomiendo que pongas en el dial plan un "." (sin las comillas, claro) con esto te acepta todo lo que marques en el telefono.
Un saludo
Arusnet
 

jlsistemas

Joined
Oct 30, 2010
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#5
Re: Re:Configuracion llamadas entrantes /salientes SIP

hola te saluda jorge la verdad q soy nuevo y telefonia ip....y t estaria agradecido si me puedes ayudar...te comento trabajo en una municipalidad aca en peru, tengo un radio enlace el cual trabaja muy bien bajo esa plataforma envio voz tengo un server de voz en linux y elastix tengo los sigueintes problemas:
no entran ni salen llamadas de afuera y reiniciando el server se solucionan el problema.
cuando marco para afuera me sale un mensaje en ingles...
se me mueren los telefonos tengo q apagralos y volver a prenderlos son unos granteams se que son malos...y tengo otos linsys pero igual.
la verdad que ya no se q hacer. si m puedes ayudar t lo agradeceria...
 

Victor

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#6
Re: Re:Configuracion llamadas entrantes /salientes SIP

Aquí dejo la configuración correcta para conexión SIP de Voztele.com y Elastix 2.0.0:

PEER Details:

canreinvite=no
dtmfmode=rfc2833
type=peer
defaultexpirey=300
host=voztele.com
insecure=very
qualify=yes
fromuser=
fromusername=
fromdomain=voztele.com
username=
secret=

USERS Details:

type=friend
host=voztele.com
context=from-trunk
 

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