Exchange 2007 UM Integration

[internalit]

Hi

Please note my background is IT, not PBXs

We have an Avaya IP office (406?), Avaya VP Pro, and IP phones as well as digitial.

Just migrating to Exchange 2007 and have installed the Unified Messaging component.

How easy is it to link the two up. Any documentation around? do I need to buy some hardware interface/gateway - or can I simply give Exchange the IP address of the Avaya PBX and it just works! (LOL!!!!)

Thanks for any tips

 

[kwing112000]

I dont think it is supported on 2007 yet. You can check out

marketingtools.avaya.com/knowledgebase for the docs on how to set it up.

 

[mdekruif]

Hello,

please check this tread:

http://www.tek-tips.com/viewthread.cfm?qid=1367220

I've been told in the future avaya IPO will connect direct to the UM.

mathijs

 

[RavenAndy]

We have it running in house, we use a basic SIP trunk and a third party piece of software that handles the conversion. Works very well and no complex set up as with the asterix, just give it the IP address of both the IP office and exchange server and off you go.

If anybody is interested I will post more details.

 

[mdekruif]

It is very interesting.. It's a nice sollution for IT minded people

 

[johnnybrian]

Hi RavenAndy!

id like some more info on how to get UM to work with IPO. Can you post something, please?

/JB

 

[kholladay]

I'd like more info on your solution. We have been using AudioCodes gateways to accomplish this.

Kyle Holladay
Certified: ACACN, ACSCI, ACSCM, TIA-CTP
ACE Implement: IP Office

"If it worked the way it should you wouldn't need me

 

[ronromano]

What 3rd party piece of software?

 

[Shaun E]

We got it working on ours (running 4.1(9) ) in a roundabout way.

We have a H323 trunk between IPO and Asterisk (it's the only way to get it to send through the caller ID; I can't find a way to do it with SIP trunks).

Asterisk sets the appropriate headers (mainly the Diversion header) and then passes the call on a UDP SIP trunk between Asterisk and SipX.

From there the call is passed from SipX via a TCP SIP trunk to Exchange.

This works for Voicemail (although we're not using voicemail, as all of our users have VM Pro mailboxes). It also works for UM access to the mailbox and auto attendants (we mainly use it for mailbox access).

-Shaun

 

[kristiandg]

RavenAndy.

I'd love details on both the PBX's side and Exchange side. We're currently using 8-port ATA's (SIP Capable of course) and I'd love to make those SIP TIEs instead, to get rid of that nasty 2-ring delay.....

If you could either make a FAQ or email me at "kristian_guntzelman" at "realworldtime.com" that would be great....

Thx....

 

[kholladay]

Did anyone get a response to this? If so would you forward a copy of the information over to me at
kholladay at yahoo dot com?

Kyle Holladay
ACA-I, ACA Call Center, ACS-I, ACS-M, TIA-CTP, MCP/MCTS Exchange 2007
ACE Implement: IP Office

"If it worked the way it should you wouldn't need me

 

[msmithCPIP]

RavenAndy, we too would like to know what the 3rd party software solution is, what are you using?

 

[Gregos]

RavenAndy, we too would like to know what the 3rd party software solution is if you dont mind?

 

[kristiandg]

ShaunEwing,

Can you provide details on your setup? It appears that the method RavenAndy is using will be hidden forever.

Thx.

Kris G.

 

[kholladay]

RavenAndy is using SIPMerge.

Kyle Holladay
ACA-I, ACA Call Center, ACS-I, ACS-M, TIA-CTP, MCP/MCTS Exchange 2007
ACE Implement: IP Office

"If it worked the way it should you wouldn't need me

 

[puregold]

Crikey Andy, they are loving you!!!!!

ACE - Avaya Certified Expert
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