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Exchange 2007 UM integration

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vdn

Programmer
Mar 24, 2008
22
US
Hi,

Does anybody have IP Office talking directly to Exchnage UM or through a SIP gateway or proxy? I have a few customers that would like to integrated directly to their Exchange 2007 server using IP Office. I have our enterprise Avaya switch talking to our Exchnage server in-house through a SIP gateway and was wondering if we can do this with IP Office?

Thanks
 
Thanks Ron,

Would you just use analog lines to the Audio Codes to get this to work? And if so, how do you config the IP Office to send VM to the analog lines?

Thanks!

Dan
 
I haven't done one yet. On the Audiocodes site, they have a doc on how to do it with one of the boxes but its analog and that seem like going backwards to me. They have other docs on avaya product integrations using PRI or T1 but not for IPO and that is the way I want to offer it. I'm sure it can be done this way on IPO but haven't had the time to try.

I'm still not seeing what someone gets out of it. If the client chooses to go all microsoft for messaging and AA OK, but to run run both seems like a waste.

I like VM pro and from what I have read on Exchange 2007, overall I don't think it (Exchange) is as powerful. I might be wrong, but I would like to see a demo of Exchange 2007 UM to decide.

 
We were able to get it to work with another 3rd party SIP Gateway. I think in order for IP Office to connect directly to Microsoft UM, SIP Endpoint must be supported on the IPO.
 
Thanks joe..

Did you just create an analog hunt group and use "group" for voicemail type and point to that hunt group? Is there some cfg setting you could share?

Dan
 
Can you explain how you are doing this?

Thanks,

Dan
 
Sip merge is a software application which sits between a sip trunk on the IP office and exchange 2007. You create a trunk on the IP office, set up forwarding on no answer for the user to a short code you set up to use the trunk and away you go.

The big problem is you cannt do out of exchange dialing, i.e. go into contacts in exchange and ask it to dial the person.
 
I've been reading up on the Microsoft side of UM. Am I correct in thinking that all exchange 2007 UM features need the enterprise server and enterprise CALs? this is a major roadblock to even trying to get UM working with IPO for my site and potential customers.
 
No, you just need to purchase the "enterprise license" or upgrade to it from the standard license you currently have which is about 35 dollars extra and just for the users you would like to UM enable.
 
So RavenAndy,

Thanks for the reply. Maybe you could describe the short code you use. Does the Sipmerge basically convert the protocol from UDP to TCP for Exchange? Is there some doc from Sipmerge on how to set this up? And how much does the proxy cost?

Thanks!!

Dan
 
If you look at the SIP conversation between say an AudioCodes and UM vs the IPO and UM you will notice that the only thing missing in the INVITE is the diversion header (e.g. Diversion: <tel:5000>;reason=unknown;screen=no;privacy=off). The rest of the conversation is identical. Until Avaya adds the ability to provide a diversion header to the INVITE you will not be able to connect directly to UM.

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
 
OK,

I'll get the SIPMerge and test it that way.

Thanks
 
RavenAndy can you share your short code you used with SipMerge. I have it working going to the hunt group just not a users mailbox. Thanks fo any help
 
I am testing Sip merge now and having a few problems. Here is the short code SipMerge recommends:

Code 8N
Feature Dial
Telephone Number N"@10.0.1.19" (use IP of your proxy)
Line Grooup Id 0

 
Not yet... getting some SIP errors from the proxy.
 
This is what I am getting

23/05/2008 13:16:36 SIPStack sending to 10.50.50.26:5060 TCP:
INVITE sip:502@exchange.kyvonstl.local SIP/2.0
Via: SIP/2.0/TCP winsvr05.kyvonstl.local;branch=z9hG4bK6a6226254c7177a41b32c40305e1a66
Via: SIP/2.0/UDP 10.50.55.253:5060;rport;branch=z9hG4bK4916670398f5adbdad1a45794f56faf9
From: mlebowitz <sip:mlebowitz@10.50.55.253>;tag=c5933bd0aa22ca30
To: <sip:502@10.50.50.24>
Call-ID: 69c4c87b3a269a262fa408d7b0f92233@10.50.55.253
CSeq: 482680923 INVITE
Contact: mlebowitz <sip:winsvr05.kyvonstl.local>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Type: application/sdp
Content-Length: 277
Supported: replaces
Diversion: <tel:502> ;reason=no-answer
Record-Route: <sip:winsvr05.kyvonstl.local:5060;lr>

The User in Exchange is 502. When I forward my calls to it exchange will not answer. I can call exchange piolet number 700 and get with AA to answer. Thanks For any help this is driving me crazy.
 
What advantage is there for using Exchange as a voicemail system. Seems a long way around the houses to me!!!

I can't see it working as slick as IPO VM anyway.

Jamie Green

ACA:Implement - IP Office
ACS:Implement - IP Office


Football is not a matter of life and death-It is far more important!!!!
 
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