Asterisk as VM for Avaya CM - H.323 / called station woes

I’ve got FreePBX running along side an Avaya Definity. Trunking with H.323 and I can make / receive calls no problem.

INB4: SIP is not an option for us on this Definity.

I’m migrating users away from the Avaya / Audix combo to purely FreePBX, and stage one is to get everyone using FreePBX for their voicemail. I’ve already accomplished this same feat with a remote location, but that Avaya has SIP capabilities and everything setup flawlessly

The problem I have with this location is anytime the Avaya Connection Manager sends a call to Asterisk, the Called Station ID doesn’t go with. I’ve tried damn near every combination of trunk group, signaling group, hunt group, vectoring and coverage path settings I can think of. I’ve even tried to rewrite CPN as the called extension’s number using public-unknown-numbering - still no dice.

I prefer to research and work things out on my own, but I’m at my wits end with this one. If anyone has any insight into getting the proper information delivered to FreePBX so voicemail functions…I would be eternally grateful.

That’s going to make this a slow slog no matter what you do. You can try to install openh323, but that’s going to be a hard way too.

I don’t know what to tell you - if we were going the other way, we’d just declare the outbound route as an “intracompany” route and the extension’s caller ID is their extension. Coming back from the other side, I just don’t know how you’d do that.

There are a couple of people here with some Avaya experience. Maybe one of them can chime in.

Openh323 is already up and running. Is there some feature I’m ignorant to that I should be taking advantage of? I’m all ears at this point, been banging my head against the wall…

OK, so you have established an H323 connection between the Avaya and Asterisk? If so, awesome.

I don’t know how to get the Avaya to recognize this link as an “intracompany” route and pass extensions and it sounds like you’ve been working through that as well. If an Avaya calls another Avaya on a “local” H323 connection, does it pass the Calling Machine’s extension?

If there’s no other way to do it, you could always direct the Avaya calls to “*98” instead of trying to route each caller to their specific mailbox on Asterisk.

Not to gripe too much, but why did you decide that voicemail was the pivot you decided to turn on? I would have thought intra switch calling and one-at-a-time conversion of the phones would have been an easier milestone to hit. Regardless, let’s see if we can’t figure this out.

Yes the H323 trunk is established and two way calling works perfectly. I can login to my softphone provided by FreePBX and dial any extension on the Avaya, on other site’s Avayas, or dial public out through our PRI (also lives on Avaya).

The way VM works right now is similar to what you’re suggesting, except instead of *98, Avaya dials 6600 and then Asterisk has rules that deal with inbound calls for 6600 in extensions_override_freepbx.conf:

exten => 6600,1,NoOp(${CALLERID(num)})
exten => 6600,2,NoOp(${CALLERID(rdnis)})
exten => 6600,3,GotoIf($["${CALLERID(rdnis)}" != “”]?4:400)
exten => 6600,4,Playback(silence/1)
exten => 6600,5,Voicemail(${CALLERID(rdnis)}@Avaya-vm)
exten => 6600,6,Hangup
exten => 6600,400,VoicemailMain(${CALLERID(num)}@Avaya-vm)

This same set of rules works perfectly with our remote office Avaya system - but again, that’s SIP and all of the information is properly formatted / sent to Asterisk. It’s technically working with the H323 trunk, too, except the calling station gets their VoicemailMain instead of the called station’s Voicemail greeting because there’s no Called Station number attached.

We have four sites across three states that all use the VM system (Audix) based here, which is tied into our Avaya. It was decided we’d rid ourselves of Audix first, then once that is gone migrate the local users off Avaya.

I am using FreePBX as a voice mail server for Avaya Communication Manager (successor of Definity) after I got rid of Audix.
I also only have H.323 trunks between PBXs and no SIP, which I don’t have licenses for on the Avaya, so I am pretty much in the same boat as you.

Here is what I have done to make this work. I am using AAR table and uniform dial plan to route numbers to FreePBX, as well as the voice mail direct dial prefix on FreePBX.

Let’s say I have an extension on Avaya that’s 3111.
I also create a custom extension 3111 on FreePBX with voice mail account active.
I have a unique coverage path for extension 3111 on Avaya, let’s say it’s coverage path 1. (Different extensions would need different coverage paths).
Coverage path 1 points to r1, which is remote coverage table 1.
Entry in remote coverage table 1 is the AAR feature access code, in my case *9 plus the extension on FreePBX, so *93111 in our example.
In the AAR table I point dialed digits to a route pattern where I insert the FreePBX voice mail direct dial prefix (in my case #2, default is different).

So essentially whenever a call on Avaya goes to coverage path and voice mail, what gets send to FreePBX are the dialed digits prefixed with the voice mail direct dial prefix, accessing voice mail directly on FreePBX.

This sounds complicated, but works pretty well.

Once you add live extensions on FreePBX and still want to keep Avaya for a while, you will need to make a few adjustments on the AAR table, uniform dial plan and remote coverage table, as well as the custom extensions in FreePBX. If you plan on doing that, I can tell you what extras you will need.

Let me know if you have any questions.

Brilliant. Thank you so much for that I have been overthinking it the entire time.

Knew you guys would have an answer!