H323 from callmanager to asterisk

We are trying to get a H323 gateway configured between a callmanager box and a asterisk box. The gateway is in place on callmanager and the config is in place on asterisk using the sample ooh323.cnf as a foundation (actual config file below). We are running asterisk version 1.2.13 and whenever we place a call from a callmanager extension to an asterisk extension it crashes amp and we recieve a busy signal. We have enabled the 323 log but it is not recieving any entries, however, here is a snippet of what the asterisk full_log shows with the verbosity and debug each set to 10+…

/usr/sbin/safe_asterisk: line 51:  5174 Segmentation fault      (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS}
Asterisk exited on signal 11.
Automatically restarting Asterisk.
Set to realtime thread
Running as group 'asterisk'
Running as user 'asterisk'

Immediately after recieving this entry it completely restarts asterisk. On the callmanager side we are requiring MTP and we are not using faststart, everything else is pretty much the default settings. Here is the ooh323.conf we are using.

;Define the asetrisk server h323 endpoint

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

;The dotted IP address asterisk should listen on for incoming H323
;Default - tries to find out local ip address on it's own

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

;Whether asterisk should use fast-start and tunneling for H323 connections.
;Default - yes

;H323-ID to be used for asterisk server
;Default - Asterisk PBX

;CallerID to use for calls
;Default - Same as h323id

;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

;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

;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)

;amaflags = default

;The account code used by default for all clients.

;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.

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

; 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

;ip=a.b.c.d   ; UPDATE with appropriate ip address
;port=1720    ; UPDATE with appropriate port

ip=   ; UPDATE with appropriate ip address
port=1720    ; UPDATE with appropriate port

If anyone has any ideas or suggestions on some things to try it will be greatly appreciated.

Switch to Asterisk 1.4 or up - its chan_h323 works flawlessy with Cisco Call Manager and provides support for some Cisco-related extensions.