How do I set up Asterisk as Voicemail for Avaya 8710

Asterisk Asterisk Support
1

Good Day Everyone,
I am slowly configuring an Asterisk base install to be a voicemail box for an Avaya 8710 platform. I am using the latest 1.4.x with the latest Dahdi and a single port FXO card as a proof of concept.

Can anyone give me some pointers on how I might configure this? Do I need Mode Codes turned on in the Avaya or can I do it without? (I currently don’t have it turned on and it would be a hassle to PAY them to turn on a switch for a feature that should be included… <Don’t get me started>)

Right now I have the Avaya talking to Asterisk through the single FXO port. I can see the call come in but I am not sure what to look for coming from the Avaya. Should I get familiar with NoOp()?

Google has usually been my friend on these sorts of things but so far I am not feeling the love on this search…

Thanks for any words of Wisdom…
Perry

2

After some digging around and some trial and error (Mostly Error…) I have made some progress. However, to my surprise, the progress is via SIP and not an FXO channel. So here’s what I have done so far…

I created a SIP signaling and trunk group in Avaya. I then told the Avaya SES server how to route the voicemail extension (In this case it was x2017 to get to Asterisk Voicemail). I then added 2017 to the UDP table as an AAR extension. I also added it to my AAR and ARS tables. All of this can be googled pretty easily. The trick came in when I was setting up my voicemail hunt group. I noticed something that I hadn’t seen before so I started messing with it and researching it.

In my Avaya hunt group form here is what I set:
Page 1:
Group Name
Group Extension
Group Type - UDP/MIA
TN and COR
ACD, Queue, and Vector are all set to ‘N’
Everything else is either set to ‘N’ or left empty.

Page 2:
Message Center - ‘sip-adjunct’ (Here is where I had the ah-ha moment)
Voicemail Number - 2017
Voicemail Handle - 2017
Routing Digits AAR/ARS Access Code - Mine is set at 100 but you will need to do a ‘display feature-access-codes’ and look on the first page to see what your AAR Access Code is.

Now at this point you should be able to either dial your voicemail extension directly or the group extension of the hunt group and you should see it come in on the Asterisk Side… Even if nothing is programmed in Asterisk yet, you should at least see a reject of some sort. At least you know the traffic is passing from Avaya to Asterisk.

I then created a coverage path (This one was 500) to route my voicemail coverage. I pointed the coverage path to h336 (That is my hunt group for voicemail).

Finally I went in to the programming of my test station in the Avaya and changed the coverage path to 500 on the first page. On the second page I changed the MWI Served User Type to be ‘sip-adjunct’.

Now… Back in the Asterisk world, I created / plagiarized the following code that should see the original called number being passed in. This is where I am stuck because it isn’t working…

exten=2017,1,GotoIf($["${RDNIS}"=""]?400)
exten=2017,2,MailboxExists(${RDNIS}@default)
exten=2017,3,Congestion
exten=2017,103,Voicemail(su${RDNIS})
exten=2017,104,Playback(vm-goodbye)
exten=2017,105,Hangup
exten=2017,400,VoicemailMain

This will look at the RDNIS to see if the Avaya SIP server has passed in an originally called extension. If not then it will assume that someone is calling in to check their voicemail. If an RDNIS value is found, it will then route the call to the voicemail of the appropriate person based on the originally called extension dialed.

From this SIP debug I can see the ‘history-info’ with the originally called extension but it is not the most recent ‘history-info’ in the list so I don’t think RDNIS is picking it up.

So my Question, after all of that explanation, is how do I set my RDNIS value to be a particular extension found in the ‘history-info’?

I really hope that someone gets a good tip from the ‘SIP-Adjunct’ usage in Avaya. I am just pulling my hair out trying to figure this out. I am SOOOO close but this last little bit I just can’t figure out…
Thanks for any suggestions…

Here is the SIP Debug from a sample call…

3

It looks like I figured this one out myself. It’s probably not super clean and I certainly hope someone out there can add to this but here is the base of it…

I just learned how to read SIP_HEADER and put it in to a variable. I then took the variable and stripped out the first 5 characters which are “<sip:” and then took the next 4 characters which are the original extension.

I know there are ways of cleaning this up to work with more than just a 4 digit extension and I will be making my scenario more pretty but for now my concept is working and I can move on to the next steps… Here is my code for those who care.

exten=2017,1,Answer
exten=2017,2,Set(orig_exten=${SIP_HEADER(History-Info)})
exten=2017,3,Set(orig_exten=${orig_exten:5:4})
exten=2017,4,Voicemail(${orig_exten})
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I’m also interested in using Asterisk as Voicemail for Avaya legacy G3Si PBX. My only caveat is that I will not be able to do any SIP integration since SES does not integrate with Avaya legacy CM1 PBX.
I will be using TDM integration with Digium TE122 board. I will try the same approach as you Perry and I hope to get lucky and make it work.

5

Salazarw,

Were you able to accomplish what you wanted?
I am using an ISDN to SIP gateway to connect to the Asterisk and wanting to use it as a voice mail server.
How does Avaya send information of the called party, i.e. the station that was originally dialed and went to coverage? Maybe over DTMF tones?
Cause that is apparently information, which my Asterisk is not detecting.