Need help with simple H.323 trunk calls to Avaya

[phoneguy01]

Hi,

I've got an Avaya Communication Manager R3 PBX successfully making calls TO the Asterisk (Trixbox/FreePBX install) 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!

 

[engjohn]

you will need to setup an outbound trunk and the dial strings

 

[phoneguy01]

I need specific info I.E. EXACTLY what to type in (based on my above h323 config) on the custom trunk and/or custom extension fields on the FreePBX interface. I'm a novice as you might guess when it comes to Asterisk and even more so on the FreePBX/AsteriskNow/Trixbox solutions.

 

[engjohn]

you will have to try something like this.

Make a new CUSTOM trunk:
Outbound CallerID = whatever
Dial Rules = x.
custom dial string = oh323/$OUTNUM$@ip.of.avaya.systemortnumber

Then create an outbound route to use the trunk...
Name the Trunk something like ToAvaya
dial patterns = 3xxx (for extensions in the 3000 to 3999 range)
In the trunk sequence, select the Avaya Custom trunk

 

[phoneguy01]

So the outbound dial rules would cover the ext I'd be dialing on the Avaya right?...so I don't need to create a new ext in the Asterisk for this? Just create the trunk in the GUI, create the route so that someone dialing the exact ext or the range of ext's would be forced to this route and the route would then use the h323 trunk I define? Just making sure I got the thinking right

 

[engjohn]

Sounds correct. The dial patterns on the outbound route will be matched to send the calls to the Avaya Trunk. The Dial Rule on the Avaya Trunk "x." means that anything goes.

 

[phoneguy01]

Well I guess this is better than before. I get 'all circuits are busy now, please try your call again later'....a message from the asterisk server. I'm actually watching for a specific ext in the route so I'm sure it's matching that one entry. If I remove the entry in the route so it doesn't know what to do, I get a 'call can't be completed' sort of message. I know the trunks are in service..I can call from the avaya over to the asterisk still, just can't call from asterisk to the avaya. I see there was someone else on here in the past that had the exact same problem but never resolved it - he stopped using trixbox and went with a basic install of asterisk using the command line to type everything in the conf files.

Any ideas on why it says circuits are busy? Why does it normally give that recording?

I see a lot of this when I do a asterisk -vvvvvr to watch the call activity...

-- Executing [s@macro-dialout-trunk:27] Goto("SIP/63785-0a0137f8", "s-CHANUNAVAIL,1") in new stack

63785 is the ext I'm calling from in asterisk (sip softphone)

 

[phoneguy01]

Hey it's working now! I've been playing around with the h323 part of the dial string. I'm sure I tried it both ways, OOH323 and simply H323 in the dial string for the trunk....but just now switching it back to OOH323 it works even though I'm using the h323.conf file and not the ooh323.conf file.

Thanks

 

[sergejf]

We need to set up an H.323/G.729 to SIP/G.711 protocol/codec converter between an Avaya CM IP PBX and some SIP-based Media Server. We are considering Asterisk with G.729 codecs to handle up to 200 concurrent calls on 4 CPU cores.

Could anyone provide us with some outline of their use of Asterisk interfacing with Avaya CM, such as number of concurrent calls handled, Asterisk server configuration (CPUs), time the solution has been in service, reliability and quality of the solution, etc.

Thanks a lot!

Serge

 

[busster]

We have had a connection to our Definity / CM II now for several years. Started trying to perfect the H.323, but gave up and are using a PRI connection. Please realize this was end of 2006, start of 2007 so less information was available.

 

[sergejf]

Thanks Busster. I had a similar experience around that time. I just wonder if more recently anyone was successful in interfacing between an Avaya CM and Asterisk using H.323 and (ideally) G.729, as well as some characteristics of that connection: number of concurrent calls, reliability, quality, etc.

 

[leftyx]

phoneguy01...

Could you post your sig group and trunk group config's on the Avaya side?

I am trying to find out what "Supplementary Service Protocol" to use.

I have read that it should be "b" but my QSIQ Basic Supplementary Services feature is not assigned. Any help would be great.