H.323 for trixbox
Submitted by andrew on Wed, 02/14/2007 - 10:05am.
The trixbox project does not have a lot of experience with H.323 nor do we have any way to test H.323 on trixbox. We do include the new ooH323 module for Asterisk.
There is a basic config file for ooH323 in the /etc/asterisk-1.2.7.1_samples directory
If you do this command from Linux
cp /etc/asterisk-1.2.7.1_samples/ooh323.conf /etc/asterisk
amportal stop
amportal start
you will have basic H.323 functionality. You can edit the ooh323.conf file using the config editor built into the web GUI. Follow the comments in the file to get it working. Incoming calls should go to context from-pstn. You will need to create a custom trunk in FreePBX for outbound calls. I think something like this will work for the custom dial string.
OOH323/$OUTNUM$@XX.XX.XX.XX:XXXX
(x is the IP address and port of your H.323provider)
Hope this helps get you started. If you come up with more complete documentation on how to use OOH323 with a H.323 provider please post it to this forum.
Re: H.323 for trixbox
Wed, 06/07/2006 - 12:18am
OOH323 works (sort of) when following your instructions. Dialing H.323<=>SIP and SIP<=>H.323 rings and can be answered, and has audio, but it dies after 15-30 seconds with the message:
Attempting native bridge of OOH323/EXT.7003-7493 and SIP/5000-e10d
This is the same failure that I got with AAH, which leads me to believe that there is a problem with OOH323 not signalling properly. (It also prevents H.323 phones from calling other H.323 phones.)
Does anyone know of a "fix" for this, or, if the version of OOH323 packaged with Trixbox is not the most recent version, would it be possible to "easily" update it?
-30-
Re: H.323 for trixbox
Fri, 06/09/2006 - 5:15pm
sorry guys, yesterday i made a little miracle, i used your "suggestions" and now i'm able to make calls between sip and h323 phones. follow all the lines u find in the tutorial and u'll do the same
please keep alive h323 protocol
u were fantastic
many tnx
grazios72
Re: H.323 for trixbox
Mon, 06/12/2006 - 9:28am
Just one quick thing guys: I wanted to install GnuGK as an h323 gatekeeper. I trried the pwlib /openh323 / gnugk compile but got a zillion errors! I downloaded the old asterisk@home addon and, after a few restarts all is fine now. Did I do anything wrong? I just hope that installing the asteriskathome-h323-1.0.zip didn't messup trixbox's existing h323 implementation!
:-o
Re: H.323 for trixbox
Mon, 06/12/2006 - 12:12pm
Re: H.323 for trixbox
Fri, 06/23/2006 - 6:22am
Hi grazios72,
Would you please give me a hand?
When you have a SIP <--> H323 call, are the RTP packets (audio) are going through Asterisk or not? Type "rtp debug ip x.x.x.x"
(where x.x.x.x is your ip address of SIP phone or H323 phone) at the asterisk console should enable the debug message. If you see a lot of messages, it means the RTP packets are going through Asterisk.
SIP has the reinvite feature and RTP packets for the SIP <--> SIP call go directly between 2 phones.
I want to know for the SIP <--> H323 call, do the RTP packets go through Asterisk?
I tried to use "h323 channel driver" provided by Asterisk (i.e. asterisk/channels/chan_h323.c) but the native bridge was not working.
I tried to use "asterisk-oh323 channel driver", same thing.
Please give it a quick try and if it works, I may change the channel driver.
Your help is greatly appreciate.
Thanks
ken
ck000@mailcity.com
Re: H.323 for trixbox
Thu, 07/13/2006 - 1:34pm
Hi,
Could you tell me what should I do to use some H.323 terminal (ex. NetMeetiug) as an external extension number. I mean how to configure the chanel to register H.323 phone as for example 4000 extension and to have a possibility to call that ext form some SIP ext.
Do I need some Gatekeeper? Or where should I use appropriate configuration?
Currently I'm using Asterisk@Home v.1.2.4 and ooh323 channel.
I succeded to setup H323 -> SIP call.
Thanks in advance.
My H323 trixbox experience
Fri, 07/21/2006 - 1:47am
I have been trying to get H323 to work with trixbox for some time now. First I tried to get the included ooh323 channel driver to work by setting up 2 trixboxes and creating trunks to each other. My first experiences were not that good. Everytime when i tried to make a call to my other trixbox over the oh323 trunk one of the asterisk would crash and restart. I have tried a lot of configuration options but to no aveal. Because it crashes i was presuming something had to be wrong with the channel driver so i compiled it from source and tried again. Again this would not work and would still crash my asterisk when making a call (the call does not hang up because the other side will crash). Because i've seen posts/mails about setups where this channel driver does seem to work i tried rebuilding everything from source (I even created a little script, if somebody needs it let me know). This means asterisk zaptel sounds spandsp libpri addons. After I build them and started asterisk all seems to be working well. I can make calls between 2 asterisk servers hang up and call again without crashing or problems. I dont know what caused these problems but i do know the steps I made helped me getting a working ooh323.
My real intention was actually to connect my asterisk to our company's h323 gatekeeper so i would be able to use asterisk in our VoIP network. But unfortunately I could not get this to work so I switched back again to the old oh323 channel driver and compiled it from source. I also installed the binary version of gnugk on my asterisk and now i let our gateways register to the gnugk and i can make calls to the gateway. Now i just need to find out how i can map h323 aliases to my sip/iax extensions so i can call from the h323 gateways to my trixbox extensions.
I really took me several weeks to get to this point. There is not much information regarding h323 (or i am looking in the wrong direction) thats why i post my experiences here.
-carlo
Re: My H323 trixbox experience
Mon, 07/24/2006 - 9:38pm
I too have been struggling to get H323 to work. We have a few older systems out there (most around AAH 1.5 from memory) where you could download and install the addon pack no problems.
We need to connect to a provider via H323 for out-going toll calls, however I can't get ooh323 configured correctly, though I can at least install it. The other route I took was trying to compile new pwlib / oh323 and get the old version that worked with AAH going.
Nothing seems to work and there is very little if any information floating around about setting up H323 as trunks to providers.
Re: My H323 trixbox experience
Thu, 07/27/2006 - 10:24pm
Tisconz, I'm in the same situation as you. It's very frustrating when you cannot find anything after countless hours of googling. I cannot believe why no one has ever encountered what we are going thru when H.323 was so popular and so many providers still use this protocol? Please let me know if you make any headway. I'll do the same.
Re: H.323 for trixbox
Thu, 08/10/2006 - 11:50pm
Re: My H323 trixbox experience
Fri, 08/25/2006 - 1:18am
Re: My H323 trixbox experience
Tue, 08/29/2006 - 2:24am
Hello,
I have same your problems on port 1720 and my configuration 00h323.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:
; OOH323/name where name is the name of the peer/friend profile.
;
; For unregistered H.323 phones:
; OOH323/ip[:port] OR if gk is used OOH323/alias where alias can be any H323
; alias
;
; For dialing into another asterisk peer at a specific exten
; OOH323/exten/peer OR OOH323/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
;port=3729
;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=10.0.0.115
;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=ObjSysAsterisk
e164=100
;CallerID to use for calls
;Default - Same as h323id
callerid=asterisk
;Whether this asterisk server will use gatekeeper.
;Default - DISABLE
;gatekeeper = DISCOVER
;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=default
;Sets rtptimeout for all clients, unless overridden
;Default - default
context=default
;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.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=g729
allow=gsm
allow=ulaw
allow=g723
; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad
; h245alphanumeric, h245signal.
;Default - rfc 2833
dtmfmode=rfc2833
; User/peer/friend definitions:
; User config options Peer config options
; ------------------ -------------------
; context
; disallow disallow
; allow allow
; accountcode accountcode
; amaflags amaflags
; dtmfmode dtmfmode
; rtptimeout ip
; port
; h323id
; email
; url
; e164
; rtptimeout
;
;Define users here
;Section header is extension
;[myuser1]
;type=user
;context=default
;disallow=all
;allow=gsm
;allow=ulaw
;allow=g729
[505]
type=peer
context=default
ip=10.0.0.112 ; UPDATE with appropriate ip address
port=1720 ; UPDATE with appropriate port
e164=100
OH323/10.0.0.112:1720
gatekeeper=ENABLE
OOH323/505
;[myfriend1]
;type=friend
;context=default
;ip=10.0.0.82 ; UPDATE with appropriate ip address
;port=1820 ; UPDATE with appropriate port
;disallow=all
;allow=ulaw
;e164=12345
;rtptimeout=60
;dtmfmode=rfc2833
Re: My H323 trixbox experience
Fri, 10/13/2006 - 9:08pm
Dear All,
I want to ask about trixbox setting. I have try installed at server . The Trixbox was connecting with ITG Plenet VIP H323. But I have some problem with those H323, it was softphone(SIP) can be normally to ITG (H323) , but from ITG (H323) only one time can calling to Softphone (SIP) and calling again i should be reload to asterisk. Please help me for solving those problem.
Re: My H323 trixbox experience
Mon, 10/23/2006 - 6:15am
I have exactly the same problem, sip to h323 no problem, only able to make one call for h323 to sip and then server hangs, I'm still able to make sip to sip calls but the ooh323 seems to have crashed. I have to restart trixbox to make another call? Has anybosy else had similar problems?
Thanks in Advance
Re: H.323 for trixbox
Mon, 12/11/2006 - 11:45am
Hi
How do i see if i have the file
ooH323 in the /etc/asterisk-1.2.7.1_samples
i tried to enter this after root@asterisk1 ....
but file is not found, is there anyway to browse for the file?
As you see i am quite new :lol:
Morten
Quote:
andrew wrote:
The trixbox project does not have a lot of experience with H.323 nor do we have any way to test H.323 on trixbox. We do include the new ooH323 module for Asterisk.
There is a basic config file for ooH323 in the /etc/asterisk-1.2.7.1_samples directory
If you do this command from Linux
cp /etc/asterisk-1.2.7.1_samples/ooh323.conf /etc/asterisk
amportal stop
amportal start
you will have basic H.323 functionality. You can edit the ooh323.conf file using the config editor built into the web GUI. Follow the comments in the file to get it working. Incoming calls should go to context from-pstn. You will need to create a custom trunk in FreePBX for outbound calls. I think something like this will work for the custom dial string.
OOH323/$OUTNUM$@XX.XX.XX.XX:XXXX
(x is the IP address and port of your H.323provider)
Hope this helps get you started. If you come up with more complete documentation on how to use OOH323 with a H.323 provider please post it to this forum.
Re: My H323 trixbox experience
Mon, 01/29/2007 - 12:20am
Hi
please send me the steps you did to get this working please, please email me on adi@weblk.com. please send me the scripts as well.
Re: My H323 trixbox experience
Wed, 02/14/2007 - 10:05am
Hi Carlo.
Can you send me the scrip that you make to fix the problem using ooh323. I have the same problem. My asterisk crash when I make a call using h323.
My email is dmonzontsd@hotmail.com
thank you
Danil
H323 trixbox experience
Thu, 06/21/2007 - 1:06pm
h.323 Asterisk <-> Avaya CM 3.1
Mon, 06/25/2007 - 9:32am
I have basic call functionality between TB 2.2 and Avaya including dtmf, using H.323.
** the only items I've not completed yet is turning MWI (red light) on Avaya extensions and using Avaya as a tandem switch to reach the pstn.
If anyone is interested in Asterisk to Avaya setup, respond and I'll post general settings...
or
email = matt@tycal.com
aim= matt mitchell
Well that was relatively easy
Tue, 07/10/2007 - 12:42am
Once I installed the G729 codec, got ooh323 (the one in asterisk addons) to compile without errors, and fiddled a little with the IP Office I now have a working connection. I did have a few small challenges. Calls would ring and be disconnected on answer. Turned out to be codec related. The link could not negotiate a common codec.
For the record, in the IP Office on the Line VoIP setting select g729 as the codec. Disable H450 and check fast start. Don't forget to setup the short code in both the Line and the Short Codes section. In asterisk add the asterisk IP Address to ooh323.conf and create a custom trunk and dial string (OOH323/$OUTNUM$@XX.XX.XX.XX:1720).
That's it!! If you want my configs let me know.
great
Tue, 07/10/2007 - 6:52am
I have g3si's, g3r's and the new s87xx platforms working, but been trying to get ahold of an ip office to test this out. Thanks for the info, I might not be able to test for another few weeks, i'll PM you if you dont mind if i run into any trouble... I do have 2 questions though..
1 - did u use 729 because of bandwidth limits? why not use the 711...?
2 - were u able to initiate a call from TB -> IP office -> PSTN... for example, make a call to your cell phone from a sip phone on TB thru ip office and out its co or pri line? This has been extremely flaky for me using h.323... just curious.
thanks again...
H323 & IP Office
Sat, 07/14/2007 - 4:29pm
Hi mmitchel,
I used g729 because TB and the IPO couldn't or wouldn't negotiate a codec when set to auto negotiate on the IPO. Initially the call didn't succeed, but after setting both ends to g729 TB -> IPO worked but IPO -> TB didn't work because of short codes (see my previous post). Once I fixed that the call cleared on answer.
I can call from IPO -> TB and use the IPO as a PSTN gateway, ie TB -> IPO -> PSTN. I have only tested call duration TB -> IPO and the call was not torn down until I hung up one end. I had the call up for 20 minutes.
I have only use an IAX softphone on TB but expect to be able to test SIP soon. (Currently only run IAX softphones.)
Cheers
http://www.trixbox.org/forums
Wed, 02/13/2008 - 6:27pm
Member Since:
2006-05-30