Trixbox 2.0 to Alcatel A4400 over h323
Submitted by tlombardi on Wed, 09/12/2007 - 4:58am.
hello,
I am trying to setup trixbox 2.0 to call the A4400 extensions from sip extensions. The alcatel is running R4.2-d2.304-4-ai-mt-c6s2 . I dont think its coming from the alcatel side since I tried other H323 gateways instead of trixbox which worked fine.
I'm running Trixbox 2.0 with asterisk asterisk-addons-1.2.7_1.2.21.1-4 and g729 and g723 codecs installed. calls from asterisk towards the 4400 are established with voice success but disconnected after three seconds...
Has anyone any idea why this is happening? Ive been trying to get around this problem for months!!! grrrr. Found several posts on this issue with no plausible explanation or solution for it. Has anyone ever got to getting around this problem?
Thanks
Tonio
asterisk-addons.i386
Fri, 09/14/2007 - 6:50am
asterisk-addons.i386 1.2.7_1.2.23-4 (do I need it for h323 ??)
[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=a.b.c.d
;This parameter indicates whether channel driver should register with
;gatekeeper as a gateway or an endpoint.
;Default - no
;gateway=no
;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=myID
e164=1234567890
callerid=TrixBox
;Whether this asterisk server will use gatekeeper.
;Default - DISABLE
context=from-trunk
;The codecs to be used for all clients.Only ulaw and gsm supported as of now.
;Default - ulaw
; ONLY ulaw, gsm, g729 and g7231 supported as of now
disallow=all ;Note order of disallow/allow is important.
allow=ulaw
;allow=gsm
canreinvite=no
dtmfmode=h245signal
asterisk-addons contains the
Mon, 09/17/2007 - 12:02am
asterisk-addons contains the ooh323 package too. It's installed along with trixbox if you installed it from the trixbox ISO. I noticed some differences in your conf file (mainly the dtmfmode) which I tried but the problem still persists. Calls are connected for 3 to 4 seconds before abruptly hanging up. When I switch on ooh323 debug in asterisk i get the following trace. Have you ever encountered this problem?
Thanks for your help
--- ooh323_request - data 3342@10.16.248.7:1720 format 0x100 (g729)
--- find_peer "10.16.248.7:1720"
comparing with "10.16.248.10"
comparing with "10.16.248.7"
+++ find_peer "10.16.248.7:1720"
+++ ooh323_request
--- ooh323_call- 3342@10.16.248.7:1720
--- onNewCallCreated ooh323c_o_2
--- find_call
+++ find_call
setting callid number 6003
Outgoing call 10.16.248.7:1720(ooh323c_o_2) - Codec prefs - (g723)
Adding capabilities to call(outgoing, ooh323c_o_2)
Adding g7231 capability to call (outgoing, ooh323c_o_2)
--- configure_local_rtp
+++ configure_local_rtp
+++ onNewCallCreated ooh323c_o_2
+++ ooh323_call
-- Called 3342@10.16.248.7:1720
--- onAlerting ooh323c_o_2
--- find_call
+++ find_call
+++ onAlerting ooh323c_o_2
-- OOH323/10.16.248.7:1720-c19e is ringing
--- setup_rtp_connection
--- find_call
+++ find_call
+++ setup_rtp_connection
--- onCallEstablished ooh323c_o_2
--- find_call
+++ find_call
+++ onCallEstablished ooh323c_o_2
-- OOH323/10.16.248.7:1720-c19e answered SIP/6003-09af3e00
--- close_rtp_connection
--- find_call
+++ find_call
+++ close_rtp_connection
--- onCallCleared ooh323c_o_2
--- find_call
+++ find_call
--- ooh323_hangup
hanging 10.16.248.7:1720
+++ ooh323_hangup
Here's my log when dialing
Mon, 09/17/2007 - 5:13am
Here's my log when dialing sip/602 to avaya/204
Mine is quite different: for instance, it never says "ringing"
I got G729 disabled for this trunk, but ooh323 still tries to use it ??
asterisk -r
ooh323 debug
--- ooh323_request - data 204@AvayaIP:1720 format 0x100 (g729)
--- find_peer "AvayaIP:1720"
+++ find_peer "AvayaIP:1720"
+++ ooh323_request
--- ooh323_call- 204@AvayaIP:1720
+++ ooh323_call
--- onNewCallCreated ooh323c_o_32
--- find_call
+++ find_call
setting callid number 602
Outgoing call AvayaIP:1720(ooh323c_o_32) - Codec prefs - (ulaw)
Adding capabilities to call(outgoing, ooh323c_o_32)
Adding g711 ulaw capability to call(outgoing, ooh323c_o_32)
--- configure_local_rtp
+++ configure_local_rtp
+++ onNewCallCreated ooh323c_o_32
--- setup_rtp_connection
--- find_call
+++ find_call
+++ setup_rtp_connection
--- onAlerting ooh323c_o_32
--- find_call
+++ find_call
+++ onAlerting ooh323c_o_32
Picking up 204 on avaya .....
--- onCallEstablished ooh323c_o_32
--- find_call
+++ find_call
+++ onCallEstablished ooh323c_o_32
Hanging up 204......
--- close_rtp_connection
--- find_call
+++ find_call
+++ close_rtp_connection
--- onCallCleared ooh323c_o_32
--- find_call
+++ find_call
--- ooh323_hangup
hanging AvayaIP:1720
+++ ooh323_hangup
--- ooh323_destroy
Destroying AvayaIP:1720
+++ ooh323_destroy
thanks for sending your
Wed, 09/19/2007 - 1:37am
thanks for sending your channel driver file. It did not work. I guess its not compiled for my system. I tried to update using yum as you said using:
yum -y install tbm-GUIcore
yum -y update
but the problem persists. What do you mean by
Trixbox backup
reinstall (=format)
are they built in commands?
backup and restore are just
Wed, 09/19/2007 - 2:12pm
backup and restore are just freepbx modules.
I'm on a P3@1GHz
Worth trying: Disable dualcore/hyperthreading in CPU and don't use the smp kernel
Can you dial a not registered H323 client? Just add custom extension using dial like this
OOH323/1234567890@10.a.b.c:1720
Hi all!
I need to link my
Thu, 05/08/2008 - 12:24am
Hi all!
I need to link my Alcatel OmniPCX Office to an Asterisk PBX, and I am in the search for options. So far I have been looking for SIP or PRI/QSIG, but our alcatel has an old software release, and if I want SIP support we'll need to spend lots of money on software, licenses... you know.
As a "cheap" alternative I would like to try H323 before the other alternatives. I know this is an old thread, but what do you think about this? Should I at least try? My config will be the following:
OXO <--> Trixbox <-SIP-> 25 phones.
Thank you very much. Regards,
Ignacio.
Member Since:
2006-08-23