ayuda con ruta entrate sip

axeellcevallos

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#1
amigos mios tengo el siguiente problema

nesesito configurar numeros DID de USA en la elastix compre algunos en
didww y configure para que estos sean redirigidos al elastix el problema es que cuando la llamada entra al elastix el elastix no reconose el peer del que le estan mandando la llamada y me da el mensage de no servicio

estos son los dato que me dio el probedor de didww

Outgoing Settings

Trunk Name: didww_us

Incoming Settings
USER Context: from-didww1
USER Details:
dtmf=rfc2833
dtmfmode=rfc2833
host=204.11.194.34
insecure=very
type=peer

context=from-trunk

tambien cree la ruta entrante con el did y la extencion donde quiero que termine la llamada

esta tiene un did zap y funciona bien

los configuro pero la llamada entra a l asterisk y asterisk da el tono de no servicio dice que es un peer desconocido

de antemano gracias por su alluda
 

gamba47

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#2
Pusistes algo en el registrer string ?

gamba47
 

axeellcevallos

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#3
laverdad no no puse nada lo configure segun el provedor de las dids
pero si me dices que devo de poner en el register string para que se registre lo intentare

este es el log de la consola cuando entra la llamada

-- Executing [13055078443@from-sip-external:1] NoOp("SIP/204.11.194.35-08b94e58", "Received incoming SIP connection from unknown peer to 13055078443") in new stack
-- Executing [13055078443@from-sip-external:2] Set("SIP/204.11.194.35-08b94e58", "DID=13055078443") in new stack
-- Executing [13055078443@from-sip-external:3] Goto("SIP/204.11.194.35-08b94e58", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/204.11.194.35-08b94e58", "0?from-trunk|13055078443|1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/204.11.194.35-08b94e58", "TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2009-09-14 21:03:58 UTC.
-- Executing [s@from-sip-external:3] Answer("SIP/204.11.194.35-08b94e58", "") in new stack
-- Executing [s@from-sip-external:4] Wait("SIP/204.11.194.35-08b94e58", "2") in new stack
-- Executing [s@from-sip-external:5] Playback("SIP/204.11.194.35-08b94e58", "ss-noservice") in new stack
-- <SIP/204.11.194.35-08b94e58> Playing 'ss-noservice' (language 'en')
-- Executing [s@from-sip-external:6] PlayTones("SIP/204.11.194.35-08b94e58", "congestion") in new stack
-- Executing [s@from-sip-external:7] Congestion("SIP/204.11.194.35-08b94e58", "5") in new stack
== Spawn extension (from-sip-external, s, 7) exited non-zero on 'SIP/204.11.194.35-08b94e58'
-- Executing [h@from-sip-external:1] NoOp("SIP/204.11.194.35-08b94e58", "Hangup") in new stack
-- Executing [h@from-sip-external:2] Set("SIP/204.11.194.35-08b94e58", "DID=s") in new stack
-- Executing [h@from-sip-external:3] Goto("SIP/204.11.194.35-08b94e58", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("SIP/204.11.194.35-08b94e58", "0?from-trunk|s|1") in new stack
-- Executing [s@from-sip-external:2] Set("SIP/204.11.194.35-08b94e58", "TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2009-09-14 21:04:11 UTC.
-- Executing [s@from-sip-external:3] Answer("SIP/204.11.194.35-08b94e58", "") in new stack
== Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/204.11.194.35-08b94e58'
 

jgutierrez

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#4
Te recomiendo que edites el archivo:
/etc/asterisk/sip_general_custom.conf
y le pongas:
context=from-pstn
Luego ejecutas desde la consola:
asterisk -rx "reload"
 

axeellcevallos

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#5
muchas garcias despues de colocar

context=from-pstn

tambien agregue en el extensions_custom.conf

[from-didww]
exten=> 13055078443,1,Dial(SIP/203)

muchisimas gracias amigos por su ayuda
 

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