Tema:
Por si me pueden ayudar tengo instaldo un elastix , y un gateway dlink de 8 puertos fxo , puedo realizar llamadas , pero ni yo oigo ni me oyen y a los 22 segundos se corta la llamada , he revisado los codecs y nada he trabajdo con ulaw y alaw y g723, por si alguien me puede ayudar
Comentarios
Que dice el log mientras
Que dice el log mientras tienes una llamada en curso, que aparece cuando la llamada se corta?
-----
Visita este [url=http://www.sindominio.net/ayuda/preguntas-inteligentes.html]link[/url] :evil:
[img]http://www.bbspot.com/Images/News_Features/2003/01/os_quiz/debian.jpg[/img]
(adsbygoogle = window.adsbygoogle || []).push({});
Gracias por tu respuesta ,
Gracias por tu respuesta , tengo un gateway dlink (192.168.1.70) y primero me da siempre el siguietne error
GOT SIP response 405 "Method Not Allowed" back from 192.168.1.70
Cuando marco desde mi extension 100 a un celular sale esto , veo en el gateway que el da la orden para que marque , suena el celuar y se contesta pero se oye solo un vacio y se corta
AGI Script fixlocalprefix completed, returning 0
-- Executing [s@macro-dialout-trunk:13] Set("SIP/100-09f336e0", "OUTNUM=096312742") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/100-09f336e0", "custom=SIP/300") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/100-09f336e0", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/100-09f336e0", "dialout-trunk-predial-hook|") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/100-09f336e0", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/100-09f336e0", "0?bypass|1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/100-09f336e0", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/100-09f336e0", "SIP/300/096312742|300|") in new stack
-- Called 300/096312742
-- SIP/300-09f39eb0 answered SIP/100-09f336e0
-- Packet2Packet bridging SIP/100-09f336e0 and SIP/300-09f39eb0
-- Got SIP response 405 "Method Not Allowed" back from 192.168.1.70
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/100-09f336e0", "hangupcall|") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/100-09f336e0", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/100-09f336e0", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/100-09f336e0", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/100-09f336e0", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/100-09f336e0", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/100-09f336e0", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/100-09f336e0' in macro 'hangupcall'
== Spawn h extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/100-09f336e0'
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/100-09f336e0' in macro 'dialout-trunk'
== Spawn extension (from-internal, 9096312742, 4) exited non-zero on 'SIP/100-09f336e0'
-- Got SIP response 405 "Method Not Allowed" back from 192.168.1.70
Parece que encontre el error
Parece que encontre el error no se que tengo que hacer , use para debug sip debug y me sale esto :
<------------->
--- (10 headers 0 lines) ---
Sending to 10.0.0.60 : 5060 (no NAT)
elastix*CLI>
<--- Transmitting (no NAT) to 10.0.0.60:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 10.0.0.60:5060;branch=z9hG4bK366a69297cc523e6;received=192.168.1.70
From: ;tag=72c5f156-858539
To: "300" ;tag=as4c6e6e37
Call-ID: 5040d273190d17ad7f064a5629f68770@192.168.1.65
CSeq: 9 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
elastix*CLI>
<--- SIP read from 192.168.1.77:46590 --->
<------------->
elastix*CLI> sip
<--- SIP read from 192.168.1.70:5060 --->
REGISTER sip:192.168.1.65:5060 SIP/2.0
Via:SIP/2.0/UDP 10.0.0.60:5060;branch=z9hG4bK659ff24297db1c43
From:"300" ;tag=d058ead0-858550
To:"300"
Call-ID:1B11-114D-46684816783AE99D6525-001@SipHost
CSeq:8274 REGISTER
Contact:
Expires:600
Max-Forwards:70
User-Agent:dlink 12-36-7924821
Content-Length:0
<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 10.0.0.60 : 5060 (no NAT)
elastix*CLI> sip
<--- Transmitting (no NAT) to 10.0.0.60:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.0.60:5060;branch=z9hG4bK659ff24297db1c43;received=192.168.1.70
From: "300" ;tag=d058ead0-858550
To: "300" ;tag=as07afe1c3
Call-ID: 1B11-114D-46684816783AE99D6525-001@SipHost
CSeq: 8274 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Lo logre
Para informacion de todos , la troncal no hay que registrar en asterisk estoy usando un gateway dlink de 4 puertos fxo , segundo debes usar la interfaz wan del dlink conectado al switch NO LA LAN (error de novato).
De ahi todo ok
Que bien, gracias por
Que bien, gracias por compartir la solución al problema.
-----
Visita este [url=http://www.sindominio.net/ayuda/preguntas-inteligentes.html]link[/url] :evil:
[img]http://www.bbspot.com/Images/News_Features/2003/01/os_quiz/debian.jpg[/img]
(adsbygoogle = window.adsbygoogle || []).push({});