Not sending correct number to UM

[leftyx]

Hello all. I am trying to get the integration between my 8700 and Exchange Unified Messaging 2010.

I have a PRI between the Avaya and AudioCodes Gateway and SIP between the Gateway and UM. The voicemail Pilot number of 2222 rings the UM and I hear the general greeting. If I call an extension I use a Remote Coverage Path to the 2222 ext. This also gives me the general greeting, but it should be the personalized greeting of the user. The PBX is just sending the 2222 from the Remote Coverage Path.

How do I get it to see the ext from which the coverage path came from?

 

[dignoffo]

leftyx,

You need to put the user extension ranges in both public unknown and private numbering tables mapped to the trunk-group number of your Exchange UM.

Ext Ext Trk CPN Total
Len Code Grps CPN
X X XX X

We have a 4 digit dial plan so it looks as follows:

4 4 99 4

You can use standard call coverage paths to your Exchange hunt group.

 

[leftyx]

You can use standard call coverage paths to your Exchange hunt group."

How would I set this up? I am familiar with Coverage Paths but I am confused how to route it to the UM?

I will try the other two suggestions thanks!

 

[leftyx]

basically I cannot get an ext to pass its own number "6702" to the UM it sees "2222" which is the UM hunt number...and gives me the general greeting.

The ext must use a Coverage path correct? So how does it cover to the UM?

 

[dignoffo]

leftyx,

On coverage path form enter hunt-group number as coverage point 1.

On my system Exchange UM is hunt-group 99, enter h99 on coverage path.

 

[leftyx]

OK, but what do I put in the hunt group??

 

[zehava]

Example of my hunt group setup below:

Group Number: 160 ACD? n
Group Name: Microsoft UM01 Queue? n
Group Extension: 75160 Vector? n
Group Type: ucd-mia Coverage Path:
TN: 1 Night Service Destination:
COR: 1 MM Early Answer? n
Security Code: Local Agent Preference? n
ISDN/SIP Caller Display:

Page 2
Message Center: sip-adjunct
Voice Mail Number: 75160
Voice Mail Handle: 75160
Routing Digits: *722

75160 is the pilot number for UM
*722 is the AAR code I used for this

 

[leftyx]

Thanks all! I am getting closer...now when I run my log trace I see this:
10:10:56.230 : 10.1.1.45 : WARNING : ( lgr_call)(203258 ) ?? [WARNING] Call #92 DigitParsing FAILED: Num:6708

And

10:10:56.445 : 10.1.1.45 : NOTICE : SIP/2.0 403 Forbidden
FROM: <sip:6702@10.1.1.45>;tag=1c614079822
TO: <sip:6708@10.1.0.16>;epid=204A55238F;tag=40daaa1dd0


6708 is set up as a mailbox...why would it be forbidden?

 

[dignoffo]

leftyx,

Are you using q-sig on your PRI trunk to the Audio Code gateway?

 

[texeric]

Be sure to update your other thread here (assuming you are the same person that started this thread)

http://www.avayausers.com/showthread.php?t=37242

http://www.theviagroup.com

 

[leftyx]

I am using QSIG, but I think I may not have all what is needed activated on the PBX.
QSIG OPTIONAL FEATURES

Basic Call Setup? y
Basic Supplementary Services? n
Centralized Attendant? n
Interworking with DCS? n
Supplementary Services with Rerouting? n
Transfer into QSIG Voice Mail? n
Value-Added (VALU)? n

 

[texeric]

Yeah I don't think you have enough activated to use QSIG properly. That being true, I think your two options are to activate the features required to use QSIG for voicemail purposes or use another integration method (H323, SIP, DSE, mode code, etc).

http://www.theviagroup.com

 

[leftyx]

Could you possibly post what a correct call to a mailbox looks like?

 

[michael650]

This might be an easy question. I have an Definity G3V8i. It is currently on a supertrunk, is it capable of being a PRI?

 

[zean]

Leftyx, did you ever get this working? I am trying the same thing, although not using QSIG nor a hunt group. Is it not possible to send the ext number through normal PRI and gateway to EUM?

Thx
Zean

 

[leftyx]

You have to use either a PRI with QSIG or a T-1 with a hunt group. I was not able to get the PRI without Qsig to send Originating number. I COULD get the general greeting only.

If there is a better way I would like to hear it myself.

 

[texeric]

With an Avaya PBX, the best option is to use QSIG or SIP. Even SIP requires some QSIG features to work properly. If neither of those are options then you can use these other methods (Assuming certain licenses or features are turned on in the PBX):
- H.323 to a SIP gateway to transcode it to SIP to UM
- DSE, to emulate a 7434ND (best) or 8434D stations
- VMI either using a T1 line or analog ports, but this will not give you ANY external calling party ID when the call is connected to voicemail
- ISDN PRI - this might be a long shot and I have never done it (if it even is supported). But this may only give calling and called party ID, but no redirected information so alot of program will have to be done in the SIP gateway to possible make this work.

http://www.theviagroup.com

 

[leftyx]

Are there any cheap SIP gateway options? Would the Cisco Unified Border Element (CUBE) work as a SIP gateway?

 

[texeric]

Take a look here to see what SIP gateways are supported -

http://technet.microsoft.com/en-us/library/ee364753.aspx

From there you can reach out to get some prices on some of those. Cost of the gateway isn't a big difference but when you add on the integration pieces like T1 or IP to IP transcoding or something else the costs go up.

http://www.theviagroup.com