jacobso1
2005-Dec-08 09:20 UTC
[Asterisk-Users] OOH323 towards cisco gateway (2691) call setup fails 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@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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