[Asterisk-Users] OOH323 towards cisco gateway (2691) call setupfails at q931: Mandatory information element is missing (96)

[Dan Austin]

Upgrade if you can.  I remember submitting a report to the ooH323c
developers about this
some months ago and the fixed it right away.
 
Dan


________________________________

	From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of jacobso1
	Sent: Thursday, December 08, 2005 8:21 AM
	To: asterisk-users at lists.digium.com
	Subject: [Asterisk-Users] OOH323 towards cisco gateway (2691)
call setupfails at q931: Mandatory information element is missing (96)
	
	

	 

	Hi,

	 

	I am using ooh323.

	I cannot setup a call towards a cisco gateway.

	The cisco rejects the call right away with : 

	Cause value: Mandatory information element is missing (96)

	            This is in the q931 part.

	 

	Cisco 'explanation'

	Indicates that the equipment that is sending this code has
received a message that

	is missing an information element that must be present in the
message before that

	message can be processed.

	 

	Show version gives :

	Cvs-head-06/21/05-23:51:26

	 

	Someone any clue ?

	 

	 

	H323.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:

	;        H323/name where name is the name of the peer/friend
profile.

	;

	; For unregistered H.323 phones:

	;        H323/ip[:port] OR if gk is used H323/alias where alias
can be any H323

	;                          alias

	;

	; For dialing into another asterisk peer at a specific exten

	;       H323/exten/peer OR H323/exten at 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      ;UPDATE this to proper ip address of your
asterisk box

	 

	;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=TK_BRU_AST1 

	e164=100

	 

	;CallerID to use for calls

	;Default - Same as h323id

	callerid=TK_BRU_AST1

	 

	;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=from-sip2

	 

	;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.

	;Default - ulaw

	; ONLY ulaw, alaw, gsm, g729 and g723 (g723.1) are supported as
of now

	disallow=all     ;Note order of disallow/allow is important.

	allow=g729

	allow=alaw

	allow=ulaw

	 

	; dtmf mode to be used by default for all clients. Only rfc2833
supported as

	; of now.

	;Default - rfc 2833

	dtmfmode=rfc2833

	 

	; User/peer/friend definitions:

	 

	[TK_BRU_GW1]

	type=friend

	context=from-sip2

	ip=195.xxx.yyy.zzz

	port=1720

	disallow=all

	allow=g729

	incominglimit=3

	outgoinglimit=3

	rtptimeout=60

	dtmfmode=rfc2833

	 

	 

	 

	 

	 

	 


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