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

Thu Dec 8 14:37:29 MST 2005

Hi,

I upgraded my chan-ooh323

Same problem

I was running 0.2, now 0.3 (that was the latest I did found)

Do I need to upgrade asterisk too ?

Up to 1.2.1 ?

   _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Austin
Sent: jeudi 8 décembre 2005 18:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] OOH323 towards cisco gateway (2691)
callsetupfails at q931: Mandatory information element is missing (96)

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

--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.13.12/193 - Release Date: 6/12/2005

--
No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.13.12/193 - Release Date: 6/12/2005

-- 
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.13.12/193 - Release Date: 6/12/2005

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