H323 trunks on TB 2.6.0.5 ?
The trix distribution does
Sun, 03/23/2008 - 10:23am
The trix distribution does no include the h.323 module. If you rebuilt Asterisk with the module you would not be able to update Asterisk via trix anymore.
If you have a server with a feature set that is acceptable to you this may be an acceptable compromise to rolling your own setup.
Scott
--
Scott
aka "Skyking"
Suggestion that will
Mon, 03/24/2008 - 6:35am
This works on 2.4
Tue, 04/01/2008 - 1:52pm
I used this for H.323 support on TB 2.4.
I'm not sure why it wouldn't work on 2.6, but I haven't tried it.
http://www.trixbox.org/forums/trixbox-forums/h-323/installing-h-3...
Josh
Hi all,
Mon, 05/12/2008 - 11:32pm
Hi all,
I've testing H323 for a week, because I'll need support for my next project. Using trixbox CE 2.6.0.7 I had only to download the asterisk-addons package (using freePBX package management module) and then create a ooh323.config file under /etc/asterisk.
I did not have to recompile asterisk...
Basic tests had worked, calling into a TB extension from a H323 Softphone. Hope this helps.
Let me get this straight,
Tue, 05/13/2008 - 9:13am
No, I was not able to do
Tue, 05/13/2008 - 12:02pm
No, I was not able to do that. Sorry if I did not explain myself better. I was able to install TB, download the asterisk-addons package from Digium, and copy the ooh323.conf file included in that package to /etc/asterisk. Then a "core show channeltypes" showed the OOH323 channel as available. Then I defined a custom trunk under FreePBX to dial through the OH323 channel and I was able to route calls...
So as long as you make sure
Fri, 05/16/2008 - 8:32am
So as long as you make sure that you have the asterisk addons for the same built of Asterisk you can add the module in without compiling?
When you install add-ons you have to make clean and make install. This requires the source to be loaded on the server.
In a default 2.6 install the h.323 module is not in Asterisk
maitrix*CLI> core show channeltypes Type Description Devicestate Indications Transfer ---------- ----------- ----------- ----------- -------- Zap Zapata Telephony Driver w/PRI no yes no MGCP Media Gateway Control Protocol (MGCP) yes yes no Local Local Proxy Channel Driver yes yes no Phone Standard Linux Telephony API Driver no yes no SIP Session Initiation Protocol (SIP) yes yes yes IAX2 Inter Asterisk eXchange Driver (Ver 2) yes yes yes Agent Call Agent Proxy Channel yes yes no ----------
However it does look like the module is in /var/lib/asterisk/module
So you added it in /etc/asterisk/modules.conf and then reloaded and it worked fine?
Scott
--
Scott
aka "Skyking"
Here is what I did:
Fri, 05/16/2008 - 9:34am
Here is what I did:
- Fresh TB 2.6.0.7 install
- Download the asterisk-addons package from http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.di...
- Copy the configs/ooh323.conf.sample to /etc/asterisk
- Restart asterisk (amportal restart)
And... voila! A "core show channeltypes" on asterisk console shows the OOH323 channel as available, and I'm able to call EXTENSION_NUMBER@ASTERISK_IP from my H323 Softphone.
Probably, next week I'll be testing H323 communication with our Alcatel PBX, I'll let you know the news.
Thank you iverona for the
Fri, 05/16/2008 - 2:03pm
Thank you iverona for the procedure.
I have download the file asterisk-1.4-current.tar.gz but I do not know in which repertory to copy it in the tribox server.
After the copy, what are the commands which I have to make?
By opening the file to download, I saw a file h323.conf and not ooh323.conf, are it the same thing?
Please could you tell me in which repertory to copy the file and which are the commands which I have to make then?
Thank you.
Hi josepha,
Sorry, my fail!
Fri, 05/16/2008 - 2:17pm
Hi josepha,
Sorry, my fail! You have to download the asterisk-addons-1.4-current.tar.gz not the asterisk-1.4-current.tar.gz. It seems that the forum does not allow to edit a post, so I can not fix it. But download the asterisk-addons tarball and you will find an ooh323.conf.sample in the configs directory. Then, as I said, just copy this file to /etc/asterisk/ooh323.conf and restart asterisk. You should be able to make h323 calls from and to your asterisk extensions.
hi,
I made as you have say
Sat, 05/17/2008 - 4:55pm
hi,
I made as you have say and module OOH323 is now active. I configured the trunk and the outbound routes and there I can call since my softphone sip the h323 phone but I have not the sound of the sip towards it h323. While the sound of h323 towards sip works. It is the codec g729 that I use on the phone h323. The following codec is activated in my file ooh323.conf
disallow=all
allow=gsm
allow=ulaw
allow=g729
allow=g7231
Please Why we not hear the sound of the sip towards it h323?
Best Regards,
Hi,
Sun, 05/18/2008 - 3:23am
Hi,
In the sample ooh323.conf file included in the package, I see this lines:
;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=gsm
allow=ulaw
As I said, I did not configure anything in TB. Only fresh install, added a SIP extension (without changing the allow/disallow fields) using FreePBX and restarted the whole thing.
Using XMeeting (I'm on a Mac), I configured it for using only ulaw/alaw. Also, check if you H323 client lets you activate H.245 Tunnel & Fast Start, may be that helps. The other end is a DECT IP phone, Siemens C450IP, using SIP. I get full bidirectional sound.
Let me know if you are able to get it working!
Regards.
Hi,
I evolved a little in
Mon, 05/19/2008 - 1:40pm
Hi,
I evolved a little in the implementation of the configuration. But I have a problem with the parameter " faststart".
When I put it in "no" I arrive to call phones connected on the gatekepeer and the communication is ok in both directions. But the calls towards the international which are to route by the gatekepeer do not work. As soon as the destination picks up, the connection cuts itself.
When I puts it in "yes" the calls towards the international is ok. But the communications towards phones connected on the gatekepeer are only made in 1 direction. We not hear the sound of the softphone SIP towards endpoint H323 while the sound of h323 endpoint towards the softphone SIP is OK.
THAT are what could cause that?
HOW I could activate the faststart in the config of the gnugk?
Best regards
Hi there Iverona
Was
Thu, 06/05/2008 - 3:15pm
Hi there Iverona
Was searching and I found your post. I too which to enable H323 support on my trixBox 2.6. I manage to download the file you provided in the URL. However not to sure as how to install it. Is it possible for you to provide some guidance?
Thanks in advance
Pablo
I have an external line
Wed, 07/02/2008 - 2:16pm
Member Since:
2008-03-20