Avaya calls covering to Asterisk/Trixbox Voicemail 2

[thefunksterdeluxe]

I've seen this question asked elsewhere, but I haven't found a definitive answer.

I'm trying to use ooh323 or SIP trunking (or even analog TDMs in a hunt group if I must) to cover calls from an Avaya S87XX to a TB2.8 Comedian Mail. The trunks are working perfectly for delivering calls both ways.

When receiving a call at the TB, I'm not seeing RDNIS being sent in the debug and I don't know if it's because Avaya isn't sending it or TB isn't set up to receive it. My dahdi config has switchtype=qsig.

Here's what I've tried in the coverage path:

-(Audix-like)Hunt Group and I've tried various message centers (qsig-mwi seems to be the intuitive choice)

If I add a VM lead number that is an AAR point, then that number gets sent instead of RDNIS.

-VDN & Vector with the AAR point. No love here either. Vector variables can identify ANI, but not RDNIS

-Queue hunt group with no message center, calls are recognized as "s@from-pstn".

Of course this could work if everyone had a unique coverage path, but that's an administrative nightmare.

Thanks

 

[engjohn]

Here is some info from Voip-info on asterisk and RGN/RDNID

http://www.voip-info.org/wiki/view/RDNIS

 

[thefunksterdeluxe]

I set my extensions_custom.conf thusly:
[from-trunk]
;retrieve RDNIS
exten => _XXXX,1,Macro(stdexten,${CALLERID(rdnis)},${CALLERID(rdnis)})
exten => _XXXXX,1,Macro(stdexten,${CALLERID(rdnis)},${CALLERID(rdnis)})
exten => _12599,1,VoiceMailMain()
exten => _12599,2,NoOp(${Mailbox})
exten => _12599,3,Hangup()

And the trace still shows:

Connected to Asterisk 1.6.0.10-FONCORE-r40 currently running on trixbox1 (pid = 20550)
Verbosity was 3 and is now 6
-- Executing [12599@from-trunk:1] NoOp("OOH323/avaya-99c9", "Catch-All DID Match - Found 12599 - You probably want a DID for this.") in new stack
-- Executing [12599@from-trunk:2] Goto("OOH323/avaya-99c9", "ext-did,s,1") in new stack
-- Goto (ext-did,s,1)

Is it safe to assume Avaya isn't sending a redirecting number?

 

[busster]

It may not be what you want....but if you have mode code enabled on your switch, you should be able to connect analog ports to an Asterisk.
Set them up as VMI ports, check your system mode-code settings.
Have not seen this done, but in theory the VMI ports should send all the information necessary. Not sure how you are going to get Asterisk to use LWC to turn these lights on.

Using H323, have you tweaked the Redirect Notification on second page of stations?
How about COS, change a station to a Client Room to see if information is different.

 

[thefunksterdeluxe]

Thanks engjohn & busster for the help.

I ran across another post on Tek-Tips where they linked this Avaya Modular Messaging Doc:

http://www.avaya.com/uk/emea/en-us/resource/assets/applicationnotes/dcsmm.pdf

One of the pages showed isdn QSIG settings that I haven't set up. Qsig has got to be the answer. I'm hopeful that this will at least perform the redirect. Yes, the MWI may need to go out through analogs, but that's fine.

Regards.

 

[busster]

Let us know how this goes. Will be interesting to see if this works.

 

[thefunksterdeluxe]

Found it!

We read the SIP packet and saw the RDNIS (original called party #) was present in the "History-Info" segment. I googled and some super-genius figured out how to have that read as a variable and wrote a context to route accordingly.

My coworker and I have spent well over a month trying to figure this out and just this morning we got it working.

Here is the thread if anyone is interested:

http://forums.digium.com/viewtopic.php?p=132128&sid=8b4b22c38c756f55f3f49d140e495b7a