I have an Avaya IP Office 406 and I want that it is able to comunicate with a trixbox, particularly that the phone connected to the Avaya could call the Trixbox extensions number and viceversa.

I've added a new line in the Avaya pointing to the Trixbox IP address with codec g711u 64k and a funcion code 67 to route the call starting with this two number vs the trixbox.

With Freepbx in the trixbox I've created a custom trunk with a custom dial string
OOH323/$OUTNUM$@XXX.XXX.XXX.XXX:1720
where XXX.XXX.XXX.XXX is the ip address of Avaya
and Maximum channels = 5

and in the outbound routes I've created one named
to-avaya
Dial Patterns= 67|.
Trunk Sequence= AMP:OOH323/$OUTNUM$@XXX.XXX.XXX.XXX:1720

Now the Trixbox extensions can call with the Avaya extensions putting 67 before the extension number to call, but on the contrary when I try to call a trixbox extension from an Avaya ip-phone the trixbox extensions ring once and hangs up.

Is there anyone can help me to solve this problem?

My ooh323.conf is :

ooh323.conf
[general]
;Define the asterisk server h323 endpoint
language=it

;The port asterisk should listen for incoming H323 connections.
;Default - 1720
;port=1720

;The IP address, asterisk should listen on for incoming H323
;connections
;Default - 0.0.0.0: tries to find out local ip address on it's own
;bindaddr=0.0.0.0
bindaddr=YYY.YYY.YYY.YYY

;H.225 channel port range
;Default range is 12030 to 12230, Accepts port range from 1025 to 65500
;h225portrange=12030,12230

;Alias address for for asterisk server
;Default - "Asterisk PBX"
h323id=ObjSysAsterisk
e164=100

;CallerID for the asterisk originated calls
;Default - Same as h323id
callerid=asterisk

;This parameter indicates whether channel driver should register with
;gatekeeper as a gateway or an endpoint.
;Default - no
;gateway=no

;Whether this asterisk server will use gatekeeper.
;Default - DISABLE
;gatekeeper = DISCOVER
;gatekeeper = a.b.c.d
gatekeeper = DISABLE

;Whether asterisk should use fast-start and tunneling for H323 connections.
;Default - yes
;faststart=yes
;h245tunneling=yes
;faststart=no
;h245tunneling=no

;Whether media wait for connect for fast start call
;Default - no
;mediawaitforconnect=no

;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=from-internal

;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=ulaw
musiconhold=default

; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad
; h245alphanumeric, h245signal.
;Default - rfc 2833
dtmfmode=rfc2833

; User/peer/friend definitions:
; User config options Peer config options
; ------------------ -------------------
; context
; disallow disallow
; allow allow
; accountcode accountcode
; amaflags amaflags
; dtmfmode dtmfmode
; rtptimeout rtptimeout
; ip
; port
; h323id
; email
; url
; e164
;
;