Dan Austin
2005-Dec-08 10:13 UTC
[Asterisk-Users] OOH323 towards cisco gateway (2691) call setupfails 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@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of jacobso1 Sent: Thursday, December 08, 2005 8:21 AM To: asterisk-users@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@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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051208/08fd204a/attachment.htm
jacobso1
2005-Dec-08 14:37 UTC
[Asterisk-Users] OOH323 towards cisco gateway (2691) callsetupfails at q931: Mandatory information element is missing (96)
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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dan Austin
Sent: jeudi 8 d?cembre 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@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of jacobso1
Sent: Thursday, December 08, 2005 8:21 AM
To: asterisk-users@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@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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