Ooh323.conf

Next thing was to edit the ooh323.conf file. This file needs to be copied from the samples first:

cp /etc/asterisk-1.2.8-samples/ooh323.conf /etc/asterisk 

Here is our ooh323.conf:

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=no

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


;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
;CHANGED
;gatekeeper = DISABLE
gatekeeper = 129.125.71.171

;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=gsm
allow=ulaw


; 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                             ip
;                                        port
;                                        h323id
;                                        email
;                                        url
;                                        e164
;                                        rtptimeout

;

;Define users here
;Section header is extension
[myuser1]
type=user
context=context1
disallow=all
allow=gsm
allow=ulaw    



[mypeer1]
type=peer
context=context2
ip=a.b.c.d   ; UPDATE with appropriate ip address
port=1720    ; UPDATE with appropriate port
e164=101



[myfriend1]
type=friend
context=default
ip=10.0.0.82   ; UPDATE with appropriate ip address
port=1820    ; UPDATE with appropriate port
disallow=all
allow=ulaw
e164=12345
rtptimeout=60
dtmfmode=rfc2833

Comments on this file:

• All parameters that have been changed by us are preceded with a comment line ;CHANGED. Right now this is only the gatekeeper parameter.
• To make H323 -> SIP calls the Trixbox probably has to act as gateway for the gatekeeper hence gateway = yes might be set later.
• h245tunneling is an option that some people have changed. Right now it does not make a difference but this is a parameter that might make a difference in a later stage.
• We will be using a gatekeeper so gatekeeper = 129.125.71.171 (ip-address of the gatekeeper)