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este es el archivo de configuracion de el ooh323 de el trixbox
Objective System's H323 Configuration example for Asterisk
; ooh323c driver configuration
;
; [general] section defines global parameters
;
; This is followed by profiles which can be of three types - user/peer/friend
; Name of the user profile should match with the h323id of the user device.
; For peer/friend profiles, host ip address must be provided as "dynamic" is
; not supported as of now.
;
; Syntax for specifying a H323 device in extensions.conf is
; For Registered peers/friends profiles:
; OOH323/name where name is the name of the peer/friend profile.
;
; For unregistered H.323 phones:
; OOH323/ip[:port] OR if gk is used OOH323/alias where alias can be any H323
; alias
;
; For dialing into another asterisk peer at a specific exten
; OOH323/exten/peer OR OOH323/exten@ip
;
; Domain name resolution is not yet supported.
;
; When a H.323 user calls into asterisk, his H323ID is matched with the profile
; name and context is determined to route the call
;
; The channel driver will register all global aliases and aliases defined in
; peer profiles with the gatekeeper, if one exists. So, that when someone
; outside our pbx (non-user) calls an extension, gatekeeper will route that
; call to our asterisk box, from where it will be routed as per dial plan.
[general]
;Define the asetrisk server h323 endpoint
;The port asterisk should listen for incoming H323 connections.
;Default - 1720
port=1720
;The dotted IP address asterisk should listen on for incoming H323
;connections
;Default - tries to find out local ip address on it's own
bindaddr=0.0.0.0
;This parameter indicates whether channel driver should register with
;gatekeeper as a gateway or an endpoint.
;Default - no
gateway= yes
;Whether asterisk should use fast-start and tunneling for H323 connections.
;Default - yes
faststart=yes
h245tunneling=yes
;H323-ID to be used for asterisk server
;Default - Asterisk PBX
;h323id=ObjSysAsterisk
;e164=100
;CallerID to use for calls
;Default - Same as h323id
callerid=asterisk
;Whether this asterisk server will use gatekeeper.
;Default - DISABLE
;gatekeeper = DISCOVER
;gatekeeper = a.b.c.d
gatekeeper = DISABLE
;Location for H323 log file
;Default - /var/log/asterisk/h323_log
;logfile=/var/log/asterisk/h323_log
;Following values apply to all users/peers/friends defined below, unless
;overridden within their client definition
;Sets default context all clients will be placed in.
;Default - default
context=default
;Sets rtptimeout for all clients, unless overridden
;Default - 60 seconds
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
; when we're not on hold
;Type of Service
;Default - none (lowdelay, thoughput, reliability, mincost, none)
;tos=lowdelay
amaflags = default
;The account code used by default for all clients.
accountcode=h3230101
;The codecs to be used for all clients.Only ulaw and gsm supported as of now.
;Default - ulaw
; ONLY ulaw, gsm, g729 and g7231 supported as of now
disallow=all ;Note order of disallow/allow is important.
allow=g711
allow=gsm
allow=ulaw
allow=g729
allow=g723
; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad
; h245alphanumeric, h245signal.
;Default - rfc 2833
dtmfmode=rfc2833
[entrada-uno]
type=user
host=66.28.190.144
context=custom-entrantes
allow=g711
; User/peer/friend definitions:
; User config options Peer config options
; ------------------ -------------------
; context
; disallow disallow
; allow allow
; accountcode accountcode
; amaflags amaflags
; dtmfmode dtmfmode
; rtptimeout ip
; port
; h323id
; email
; url
; e164
; rtptimeout
;
este es el archiov de extension-custom.conf
[custom-entrantes]
exten => _8888XXXXXXXXXXX,1,Dial(Zap/g1/0${EXTEN:7})
el cisco me envia la llamada pero no la puedo terminar
me pueden ayudar para que las llamadas se completen por favor ayuda.......
Que te dice el log ?tail -f
Que te dice el log ?
tail -f /var/log/asterisk/full
Juega con los siguientes parametros:
fastStart=no
h245Tunnelling=no
h245inSetup=no
Yo tengo un gateway h.323 que funciona si y solo si coloco faststart, h245tunneling y h245insetup en NO.
[quote]
[entrada-uno]
type=user
host=66.28.190.144
context=custom-entrantes
allow=g711
[/quote]
Otro detalle es utilizar como codec ulaw en tus pruebas y luego ir cambiando a g726, g729, etc. cambia allow=g711 por allow=ulaw
[quote]context=default[/quote]
Trata utilizando from-internal como contexto.
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respuesta trixbox
al ejecutar este comando tail -f /var/log/asterisk/full el resulatod es esto:
Aug 1 09:59:03 DEBUG[14463] chan_sip.c: Stopping retransmission on '5703cb99027ea50a55b7c28f49b61a7c@200.124.232.2' of Request 102: Match Found
Aug 1 10:00:03 DEBUG[14463] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
Aug 1 10:00:03 DEBUG[14463] chan_sip.c: Stopping retransmission on '1268e2fa4a13fafa3c5bffad7c1ca54d@200.124.232.2' of Request 102: Match Found
Aug 1 10:01:03 DEBUG[14463] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
Aug 1 10:01:04 DEBUG[14463] chan_sip.c: Stopping retransmission on '3593569a1bad251e43b3ea2150238cc3@200.124.232.2' of Request 102: Match Found
Aug 1 10:01:46 DEBUG[14463] chan_sip.c: Allocating new SIP dialog for 445cd2397bb207488430013f6e939ad2@10.1.255.253 - OPTIONS (No RTP)
Aug 1 10:02:04 DEBUG[14463] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
Aug 1 10:02:04 DEBUG[14463] chan_sip.c: Stopping retransmission on '0e5c8c45147c8ff110923243541e352b@200.124.232.2' of Request 102: Match Found
Aug 1 10:03:04 DEBUG[14463] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)
Aug 1 10:03:04 DEBUG[14463] chan_sip.c: Stopping retransmission on '5964c6f62fda00c50b9456b457c9f0a3@200.124.232.2' of Request 102: Match Found
ya le cambie el g711 por el ulaw y confirme que los parametros fastStart=yes h245Tunnelling=yes h245inSetup=yes sean igulaes en ambos equipos
el cdr que se genera en el *
es 1. 2006-08-01 10:54:17 Zap/1-1... s NO ANSWER 00:00
Me referia a que nos
Me referia a que nos muestres el log en el momento preciso que haces una llamada desde el gateway H.323.
[quote]es 1. 2006-08-01 10:54:17 Zap/1-1... s NO ANSWER 00:00[/quote]
Esto esta bastante raro... porque sale la llamada por un canal ZAP cuando se supone que deberia salir por el canal OOH323?
Posteanos tu extensions.conf .
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