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/[EMAIL PROTECTED] ; ; 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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