Exchange 2007/2010 connected to IPO - FINALLY...

[kristiandg]

That isn't a question folks - I'm very excited about this. FINALLY got Exchange 2007 (and now 2010) to integrate with IPO. I'll admit, there's an intermediate piece to it all, but in the end, you can call over SIP to check messages (and it identifies you as YOU), you can leave messages for users, and you can transfer out of Exchange from a mailbox or speech-enabled Auto Attendant.

We've been testing the final deployment at one of our customer sites for a little while now and so far ZERO issues.

I'm curious how many other people out there have been trying to get this working, and if anyone else has been successful or not. I know there have been some people who have gotten 1/2 of it working (you can call via SIP or you can deliver messages via SIP), but didn't think anyone had gotten the whole deal working.

I know this sounds like a teaser, but I'm not quite sure how to present this info, as I'm not really sure who all has been looking to do this. I know I was for several years. If its just a few people, so be it. If a ton of people are looking to do this, then I need to figure out the best way to get the info out there.

Have a great evening everybody.

Kris Guntzelman
G&C Interconnects

Kris

 

[hairlessupportmonkey]

sounds good

write up an FAQ and you will be a hero!

 

[eddiebullman]

Hi I have tried to get this to work using SIPX and Trixbox in between IP Office and Exchange server but could only get Voicemail working with SIPX I could not get the Voice AA to work it would answer the call but it could not transfer the call. Also tried adding an NEC SV8100 into the mix but with no joy

I have got VMpro UMS working with exchange 2007 so have left it at that at the mo
I did not want to go the analogue to sip gateway route that a lot of people have gone to get this working

I would like to integrate the IP Office with exchange and office communications server as well

So I am glad that you have got it working and would like to know how

 

[jamie77]

We have got UMS working a treat but unable to get "play On Phone" out of Outlook to work. I can see it is SIP and it tried to make a call through the IPO, but unsure if it after a SIP endpoint or trunk. Tried both with no success.

Not sure I would ever want to use Exchange as v/m and AA but if you write a FAQ then people can take out of it what they need.

Good work

Jamie Green

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

 

[kholladay]

We do it quite often using a Dialogic or Audiocodes gateway. IP Office now supports the SIP diversion header which would allow Exchange to accept the VM request but we still don't seem to support the port shuffle that Exchange requests.

Are you saying you have this working direct via SIP or are you using a box in the middle?

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

"Thinking is the hardest work there is, which is the probable reason why so few engage in it." - Henry Ford

 

[kristiandg]

Kyle,

Yes, this is all SIP. Both inbound and outbound. Those SIP adapters/converters sound like crap. We used to deploy the Audiocodes 118 and it was OK, but always sounded like an analog line (unclear).

This is all-SIP.

Looking forward to seeing you at TM again this year.

Kris

 

[kholladay]

Would be excited to hear how you got the IP Office to respond to the port change request. If you have any rough notes on this I'd love to look at them.

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

"Thinking is the hardest work there is, which is the probable reason why so few engage in it." - Henry Ford

 

[kholladay]

And while you're at it you really should add a picture to your peerport profile

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

"Thinking is the hardest work there is, which is the probable reason why so few engage in it." - Henry Ford

 

[kristiandg]

Aw man. Forgot about that. Haven't logged in there in a while.

 

[Ekster]

Hi,

I have been fighting to get this working too and would love to know what the "intermediate" piece is. As far as I can tell the IP Office SIP trunk won't support "302 Redirects" and this is where the it all goes wrong. The UMS service listens on port 5060, when it receives an invite request it redirects it to another service that is listening on port 5065, the IP Office doesn't recognise this redirect and just hangs!

Kris, can you just give us an overview of how you achieved this? Don't need details just what your basic methodology is.

Cheers,
Eric

+++ Divide By Cucumber Error. Please Reinstall Universe And Reboot +++

 

[qlafferty]

I am interested in this solution too. I ran into the same issue as the other posts with the problem of the 302 redirect. On a whim I changed the IP Office SIP trunk to point to port 5065 so that it doesn't need to redirect. Exchange actually seems to be OK with this and answers the call. You can see my response about this here:

http://tek-tips.com/viewthread.cfm?qid=1583698.

I sort of gave up for a while after I posted that thread and haven't put much time into trying again on 6.0. Please share the details of your solution. I would be happy hearing a general description without all of the details if that is all you have time for.

 

[MarkToo]

bump... interested as well.

 

[jamie77]

Somebody doesn't want to share.

I suspect the intermediate step will be a software SIP gateway and this is not direct SIP from the IPO to the exchange server.

We are looking at using 3CX or our in house Swyx to achieve this. Still not ideal to sell to a customer and probably won't do it. Just playing in house.

Jamie Green

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

 

[eddiebullman]

Hi

It would be great to know how you got this working as I can get exchange answer the call but it will not transfer the call to the extension.
I have got UMS working with vmpro and when you click on the play on the phone in the email it rings the phone but then the DTMF will not work
If I dial into exchange the system will connect me to my mailbox once I enter my pin
So once again if you have got this working let us know how you did it KRISTIANDG

 

[kristiandg]

Sorry, guys. Been slammed - not that I don't wanna share jamie..

OK, so there's some config that has to happen and it is a throwback to an older era where you'd build voicemail boxes "separately" in audix.....

SIP trunk from IPO to SIPxECS (or avaya/nortel SCS-same thing/different skin). From there, you build stations in that system that match but with a prefix. Link exchange to SIPx. Now, on an IPO station, Call Forward on NA to, say, 111and your extension (if you're 1234 you cfwd to 1111234).

In exchange you still make their primary e164 address the main extension, but you put a second one in for 1111234 (example) so it can identify both as that box (I also do my cell phone number so I can call straight in and not need to enter my extension).

The SIPx server is what ends up providing the redirect notification, since IPO can't do it properly yet.

Transfers out go via exchange, to SIPx, to the TIE trunk back to IPO. In IPO, your SIP data is just a "*" so it basically passes everything it gets to your Incoming Call Route tables. In there, build a single entry (any number/blank), targets "." This will send ALL CALLS to the system's SHORTCODE table.

See, without screenshots, its a ton of info and sounds harder than it actually ended up being.

Yes, its an intermediate step, but its sound quality is far superior to those stupid tdm gateways, and its lighting fast. The call quality comes into play when you're dealing with 2010's transcription feature.

Hope this helps until I can get some time to whip something up.

Kris

 

[eddiebullman]

Hi Kristiandg

Have you got the auto attendant working in exchange in this setup as i could not get it working if the call came from the ip office to sipx and then passed onto exchange

if i dialed from an extension on the SIPX box to the exchange AA it would transfer the call to the avaya

could do with some help if you have it working thanks

 

[kristiandg]

I have (both speech-enabled or dtmf). But it wasn't anything odd - not like I did anything special, it just worked.

I am a little bummed that exchange doesn't support DTMF transmission via INFO packets...

 

[eddiebullman]

Hi Kristiandg

What version of SIPX are you using?

when i was playing about with this last year the way i got the leave message to work was if you go into SIPX voicemail set it to use exchange as voicemail
And on the AVAYA set up a call forward to 8ext no (81207) and send this to SIPX this would the remove the 8 and send the call to the right mailbox in exchange e.g. 1207 to match the AVAYA.

the AA side of things is what i really want to get working

the way i had it set was a sort code on the avaya of 1401 which was sent to the SIPX box
the SIPX then passed it onto the Exchange to the AA i had set with number 1401 this would answer the call and you could say the persons name and it would try to transfer the call and say sorry i could not put the call through
But if i dialed from an extension on the SIPX box it would work.
i have not got a clue where i am going wrong with this one

 

[RavenAndy]

Hi kristiandg

I would really like to see this working, do you have any more detailed instructions, how did you set the Sip trunk up? I have tried but the SIPxECS doesnt seem to talk to the IPOFFice

 

[kristiandg]

Damn, I'm sorry. Got busy and forgot to get this out to here. I'll get screenshots of how we pulled this off. I'll try to do over this weekend.