Hola amigos del foro:

He compilado /usr/src/asterisk/channels/h323 correctamente, el modulo es cargado por asterisk y tengo un h323.conf en /etc/asterisk

Tengo varias dudas respecto a h323, estoy obligado a usarlo porque tengo varios equipos (Elesign ESC1700) que utilizan exclusivamente este protocolo :roll: . Basicamente no encuentro donde hacer la configuracion para que los equipos SIP que tengo se comuniquen con los h323 y viceversa.

Este es mi h323.conf


; The NuFone Network's
; Open H.323 driver configuration
;
[general]
port = 1720
bindaddr = 0.0.0.0
tos=lowdelay
;
; You may specify a global default AMA flag for iaxtel calls.  It must be
; one of 'default', 'omit', 'billing', or 'documentation'.  These flags
; are used in the generation of call detail records.
;
amaflags = billing
;
; You may specify a default account for Call Detail Records in addition
; to specifying on a per-user basis
;
accountcode=h323-001
;
; You can fine tune codecs here using "allow" and "disallow" clauses
; with specific codecs.  Use "all" to represent all formats.
;
disallow=all
;allow=all              ; turns on all installed codecs
;disallow=g723.1        ; Hm...  Proprietary, don't use it...
allow=g723.1
;allow=gsm              ; Always allow GSM, it's cool :)
allow=ilbc
;allow=speex
allow=ulaw
allow=alaw
;allow=lpc10
;
; User-Input Mode (DTMF)
;
; valid entries are:   rfc2833, inband
; default is rfc2833
;dtmfmode=rfc2833
;dtmfmode=inband
;
; Set the gatekeeper
; DISCOVER                      - Find the Gk address using multicast
; DISABLE                       - Disable the use of a GK
; <IP address> or <Host name>   - The acutal IP address or hostname of your GK
;gatekeeper = h323.manta.telconet.net
gatekeeper = DISABLE
;
;
; Tell Asterisk whether or not to accept Gatekeeper
; routed calls or not. Normally this should always
; be set to yes, unless you want to have finer control
; over which users are allowed access to Asterisk.
; Default: YES
;
AllowGKRouted = yes
;
; Default context gets used in siutations where you are using
; the GK routed model or no type=user was found. This gives you
; the ability to either play an invalid message or to simply not
; use user authentication at all.
;
context=default
;
; H.323 Alias definitions
;
; Type 'h323' will register aliases to the endpoint
; and Gatekeeper, if there is one.
;
; Example: if someone calls [email]time@your.asterisk.box.com[/email]
; Asterisk will send the call to the extension 'time'
; in the context default
;
;   [default]
;   exten => time,1,Answer
;   exten => time,2,Playback,current-time
;
; Keyword's 'prefix' and 'e164' are only make sense when
; used with a gatekeeper. You can specify either a prefix
; or E.164 this endpoint is responsible for terminating.
;
; Example: The H.323 alias 'det-gw' will tell the gatekeeper
; to route any call with the prefix 1248 to this alias. Keyword
; e164 is used when you want to specifiy a full telephone
; number. So a call to the number 18102341212 would be
; routed to the H.323 alias 'time'.
;
;[time]
;type=h323
;e164=18102341212
;context=default
;
;[det-gw]
;type=h323
;prefix=1248,1313
;context=detroit
;
;
; Inbound H.323 calls from BillyBob would land in the incoming
; context with a maximum of 4 concurrent incoming calls
;
;
; Note: If keyword 'incominglimit' are omitted Asterisk will not
; enforce any maximum number of concurrent calls.
;
;[BillyBob]
;type=user
;host=192.168.1.1
;context=incoming
;incominglimit=4
[asterisk]
type=h323
context=default
;ie164=188020             ; Internal
;e164=17007772505        ; IAXtel
;e164=17009267595,267595 ; FWD
;e164=8829900003301      ; Freenetworks
;prefix=1880             ; Internal
;prefix=1800,1700,1888,1866,1877,4,88299
                                                                                
[h323-entrantes]
type=h323
                                                                                
[h323-salientes]
type=h323
context=h323-salientes
                                                                                
[300]
type=h323
username=300
callerid=300
context=h323-entrantes

Si a un cliente h323 le configuro como gateway la pc que corre *, entonces puedo llamar de h323 a SIP. Lo que no puedo hacer es lo contrario. Llamar de SIP a H323.

Estoy haciendo algo mal ? cuales son los archivos que debo editar?

Gracias de antemano, saludos desde Ecuador !
:twisted: