Mitel UM exchange 2010

[headbite]

I have the doc for setting up UM on exhcange 2010 and when I test from exchange it is all good but when I try to dial exchange I get dead air? If I set my SIP Peer Status to Auto Detect/Normal I get a SIP Link State of "OUT OF SERVICE" If I set it to Always Active I get dead air but my Sip Link State is O.K.

Just looking for a place to start looking.

 

[fcummins]

I'm about to start working on the integration between our Ex2k10 server and our Mitel 3300 phone system. What document are you referring to and where did you get it?

TIA.

 

[headbite]

The document is SIP CoE 10-4940-00117 I am also not sure if I can post it as an attachment. If there is an administrator out there that can answer this maybe we can help each other out during this process. I am new to both Mitel and Exchange 2010 and I am still not sure which end my problem is on. I have this thread started in a couple forums and no one seems to be able to help.

 

[headbite]

I am going to build a small website about this and post the document there. Hopefully we can share info about this and leave the site up in order to help others. I will post back to this when it is done. I hope to have it by the end of the day today.

 

[fcummins]

Thanks to our phone vendor I found the document you referenced. Below is a document I got from an Exchange guru from Microsoft:

http://mikestacy.typepad.com/files/miteltechnoteforexchangeum.doc

 

[headbite]

That is pretty much the same document for older versions. We still cant figure it out. If we test from exchange we can dial a phone and answer and the exchange is able to gather digits entered on the phone. If we call exchange from the mitel phone we get dead air. I think it is a signaling issue. Digits need to be added somewhere or the exchange needs configured to accept the call. Just a guess.

 

[boombox56]

I'm pretty sure it's a gateway issue on the Microsoft side of things. Basically the server isn't recognising the "Gateway" extension it's being sent and doesn't know what to do with the call i.e. Voicemail or AA.

 

[LoopyLou]

The like is to Exchange 2007. There is a newer one for SIp to Exchange 2010 out there. Maybe someone has that one.

There may be no i in team but there are three f's in fudge off.

 

[headbite]

thanks boombox56 I will run this by the exchange person. We were looking for a place to start looking and you gave it to us.

LoopyLou. I have the newer document and it shows the Mitel configuration very nicely but it does not go into the exchange configuration in the same detail. It does show how to test from exchange which has been very usefull. I listed the document number above and if there is a document storage area on this site I would be happy to upload it.

fcummins started a thread in the Mitel forums

http://www.mitelforums.com/index.php?topic=814.0

I will be following that thread also.

 

[quinalt]

Headbite, did you ever figure out this issue? I just upgraded to Exchange 2010 and have all the 3300 integration working except for MWI. I'm not a phone guy, so my terminology might be wrong, but here's a very quick summary of what was done:

To be able to call the exchange server we had to build a trunk i.e. trunk service assignment, assign a cos, assign the trunk service number to the sip peer profile. Since we couldn't forward directly to the sip trunk, we created a speed dial (in my case 6100) that pointed to the Exchange sip trunk. Also, don't forget to setup your dialplan. Now when 6100 is called, they are dropped directly into Outlook Voice Access.

Outbound from exchange was a little trickier. The autoattendant seems to work find with the above configuration i.e. having the AA connect you to an extension, because the call is originating from the pilot number (6100). However things like 'play on phone' didn't work until I added our extensions to the sip peer profile assignment form and email addresses/extensions to the 'URI/Number translation' table. It took many wireshark traces to figure that out, but all seems good now.

As far as MWI, apparently Exchange is supposed to send a sip Notify message to the controller when a voicemail is waiting...however, I don't see that message in the wireshark captures. If anyone has an idea why, please let me know

Hope that helps you or someone out there!

 

[headbite]

We still have not gotten this to work. I setup wirshark to mirror the port on the 3300 and I also setup a Linksys 2102 sip device. I get sip and rtp traffic when I call the 2102 device but I get nothing when I dial exchange. I am going to look at your post and repeat your steps.

Mark

 

[headbite]

quinalt what do you mean when you wrote. "Since we couldn't forward directly to the sip trunk, we created a speed dial (in my case 6100) that pointed to the Exchange sip trunk."

I have 6000 setup as our ARS digits Dialed and we are dialing it from my phone.

 

[quinalt]

Headbite,

Since I know just enough to be dangerous on the Mitel, I asked one of our phone guys about this. He said:

First create a ARS route to the SiP trunk that will connect to the Exchange server. Then create a speed dial that will follow the ARS path. Then reroute to The speed dial.

That's the way we have things setup which is not to say it's the best or only way.

 

[quinalt]

Also, I have discovered that if the UM DialPlan URI type is 'SIP URI' the MWI will not work. I changed mine to 'telephone extension' and now everything is working fine.

 

[headbite]

In my ARS Digits dialed I have.
Digits Dialed - 2910
# of Digits to Follow - 0
Termination Type - Route
Termination Number - 3

Route 3 is my Sip Exchange Route.

I have to believe that it is my routing out of the Mitel. The wire shark displays nothing. Where is the link between the sip trunks and the network interface?

 

[headbite]

I have two extensions X2910 and X6000 the both use Route 3.

Route 3 is a Routing medium SIP Trunk to SIP Peer Profile Exchange.

SIP Peer Profile Exchange goes to the Network Element Exchange. (I should not have used the same label for both the SIP Peer Profile and Network Element.)

The Network Element Exchange IP address is the exchange server.

 

[quinalt]

As far as I can tell the ARS looks right.

Try this: In the system speedcall form, create a speedcall number (maybe 2911) with the actual number being 2910 - no toll - type S/C. Dial that number and see what happens. Hopefully I'm not sending you down the garden path here...

As far as the need fore a speedcall: it seems you need that speedcall for call rerouting i.e. to voicemail because you cannot enter an ARS route in the call rerouting form.

 

[headbite]

Tried it and still nothing. I don't get any sort of SIP or RTP signaling from the MITEL system when the extension is dialed. I think I should see something on the port.