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

[jacobso1]

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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