Hi,
I’ve got an Avaya Communication Manager R3 PBX successfully making calls TO the Asterisk but I can’t seem to get the settings right to have a SIP softphone registered to the Asterisk call the Avaya system. Everything is great if the Avaya calls the Asterisk, but not the other way around.
From what I’ve read you should not edit the conf files ‘owned’ by FreePBX otherwise those manual changes you make will be overwritten later when you make changes in the web GUI. But I’ve also read that the h323.conf file is NOT owned by the FreePBX so I went ahead and put this in there to tell it about my Avaya system…
This is in the H323.conf file, NOT the OOH323.conf file just so we’re clear:
[general]
port = 1720
bindaddr = XXX.XXX.XXX.XXX ;IP of the Asterisk
disallow=all
allow=ulaw
dtmfmode=inband
gatekeeper = DISABLE
context=default
progress_setup = 8
progress_alert = 8
progress_audio = yes
fastStart=yes
h245tunneling=yes
srvlookup=no
[Avaya]
type=friend
context=default
host=XXX.XXX.XXX.XXX ;IP of the Avaya C-LAN card
port=1720
Like I said, the Avaya can now easily call the Asterisk registered SIP softphone and life is good. But for the life of me I can’t seem to find the right settings to get the Asterisk to call the Avaya.
What I’d LIKE to do is use the Web GUI of FreePBX to add the entries to allow calls TO the Avaya system. And of course have it work properly 
I’m willing to make manual entries in the ‘appropriate’ (I.E. not owned by freepbx) files if I have to, but I’d much rather get this working correctly using the web GUI, because that’s the whole point of having the GUI right?..ease of use?
I’m running the latest install (fresh install) of Trixbox: trixbox-2.8.0.1
Seems like this should be a simple/quick fix since I’ve already got the Avaya able to call the Asterisk, now I need it the other way around
Thanks!
I am not sure how this is working since the context is set to default.
Did you make a custom trunk in the Avaya to send calls to the h.323 peer?
Have you looked at my Avaya threads on IP office and FreePBX integration.
I could also swear it’s supposed to be in the ooh323.conf. Have you checked your peer status in the CLI?
Thanks for the reply. I figured it out. Still a lot more to learn though. Anyway, I setup the ooh323.conf file with the appropriate settings, added a custom trunk in trixbox that contains nothing more than: OOH323/$OUTNUM$@IP_ADDRESS_OF_AVAYA_PBX:1720 and finally, created a an Outbound route in the trixbox that contains an extension I want to dial on the Avaya as a test and then of course selected which trunk I want this to use. That’s basically it.
Here’s something else I need to learn how to do. Any help with this one is appreciated too. I need to bring a call in on the h323 trunk (got that working already) and send it back out of the trixbox to another server over SIP, like a Microsoft OCS server for example :). Is there a quick and easy way to do this? In a straight up Asterisk server where you’re editing the files, is it simply adding the OCS server in the sip.conf file and then defining what numbers will go there in the extensions.conf file? So once the call comes in on the h323 trunk and it tells the asterisk server it’s looking for ext xxxx the asterisk server will say yes ok that ext xxxx exists over there on that other server and the way you get there is via the sip trunks and so it sends the call out over that sip trunk. Does that sound right? I have a working example like that setup, but not in a trixbox/freepbx…it’s on a simple asterisk server where you enter everything via the command prompt.
Update already. I’m on a roll. I figured this one out too. I simply created a new SIP trunk with the following:
type=peer
host=IP_ADDRESS_OF_OUR_MEDIATION_SERVER
qualify=yes
qualifyfreq=20
transport=tcp,udp
Then I added an Outbound Route as a test. I dialed it and got the welcome message from our server
Sounds like you are getting it under control.
You can add the same custom extensions you would do in Asterisk in extensions_custom.conf
Make sure you read the docs in the file and look at the examples of how to hook the FreePBX dial plan.
Glad you’re getting along with things.
The doc says:
; Extensions in AMP have access to the ‘from-internal’ context.
; The context ‘from-internal-custom’ is included in ‘from-internal’ by default
So am I to assume that I should put [from-internal-custom] above any of the info I want to put in the extensions_custom.conf file? I played around with this a little and couldn’t seem to get it to work. I can play with it more but if you have a few good tips on how to be sure to get the system to use the data I put in the custom ext file I’d appreciate it
Well so far I have SIP calls working that I’ve defined in the extensions_custom.conf file. But for some reason I can’t seem to get OOH323 calls I’ve defined to there to work. Any ideas? I have OOH323 working if I define the parameters in the web GUI. I can’t figure out why this part doesn’t work.
Did some more testing and got some ooh323 working in the custom file, but it’s sort of flaky. Can’t nail it down yet but I can’t make it work as easily as I’d expect if I was just editing the normal extensions.conf file in a non-FreePBX Asterisk. Guess I’ll have to keep testing. Any tips about using the custom ext file and how it plays with freepbx is appreciated.
There are no tips that I can think of. I always start with a simple app that plays a recording or something similar. That way I know I have hooked into the right part of the dialplan.
From that point I can build my code to suit. I often call subroutines out of from-internal-custom. Don’t forget to make sure you have the variables set to pass or are using globals.
The variables is one thing I need to learn more about. I see my co-worker (he’s out for awhile due to an injury so I’m having to learn this on the fly) setup some things in Asterisk (not using asterisknow or trixbox) and I was trying to apply what he wrote in the 2 existing boxes that don’t run freepbx over to the new system i built that IS running freepbx. There’s a lot I don’t know and need to learn. For now I have the basics working. I can pass calls from an avaya system via h323 over to the trixbox and then on via sip over to our OCS mediation server. But now I need to figure out the details like caller-id, etc. Maybe he’ll be back in time to show me what he did and what the reasoning is for what he did. Right now I’m sort of in the dark not knowing if I should try to use trixbox or a standard asterisk box that was mostly built before he left. I’m leaning towards the trixbox, but at the same time I’d hate to introduce something different that doesn’t work the way we need it to and later find out we need to do it over again with a standard asterisk box. I think trixbox can do what we need, but I just don’t know enough about it to properly configure all the minor details - caller-id, etc, etc…and anything else he has setup in the asterisk boxes he built.
To help we need to know what you have done. What does the h323.conf file look like? Have you used the h323 debug commands in Asterisk?
If the ooh323 module is not loading check the Asterisk logs for an error when you try and load the module.
Remember if you don’t know the syntax in Asterisk CLI just use the question mark.
Some commands
module show like h323 - see if you have the module
core show channeltypes - verify that module is a registered channeltype
ooh323 ? - Shows h.323 commands including debug and peer status
If you have the module but it is not registering as a channel type you have an error in your config file that is keeping it from loading.
Remember to use the proper from-internal or from-trunk context depending on what part of the FreePBX dial plan you want the h.323 peer calls to enter.
I have scoured the forums and I am having no luck at all figuring out how to set up an H.323 trunk between a 1.8 asterisk server and an Avaya G3r 1.3. I think it might be my setup on the G3r, but I’m just not sure. Any help would be appreciated. Even if I could just figure out how to troubleshoot it, that would be awesome. I would be happy to share my Avaya setup once I get it working…
Hi ,
I am trying to accomplish the same thing as you . I would like to connect Avaya phone system with my asterisk box . I only started yesterday to try to configure it but I was unsuccessful. I was wondering if you had a write up or a step by step on how you configured the trunk on the asterisk side. And if I need to install special module in order to make it work.
On my Avaya I created 2 trunks , one SIP and one H323 since I didn’t know which one will work better. On my asterisk I tried with SIP it didn’t work and now I am trying to configure H323 but I am not sure if I need to install a module in order to have H323 available , that is what I am working on right now .
Thanks in advance.
What version of FreePBX are you running? How was your system installed(distro, if so what one if not what OX etc.) Asterisk version?
Do you have the ooh323 module in asterisk? ‘module show like 322’ will let you know.
Have you looked at the sample chan_ooh323.conf file with the Asterisk documentation?
Essentially once you have the trunk up you just use the FreePBX from-trunk or from-internal as the context to get the call into the dial plan where you want it.