Send incoming (PRI) call to SIP Trunk

[StarDestroyer]

A little background... I have a Mitel 5000 CP with about 90 extensions. We have a block of 200 DID numbers, most of which go to an extension. I'm trying really hard to conenct my phone system up to a Microsoft Lync server. Offcially, this is not possible. I didn't like that answer  .

I purchased a couple of SIP trunk licenses and installed AsteriskNOW. I got everything working where I can load a softphone connected to Asterisk and call an internal extension. From my desk phone, I can also dial '8' then dial an Asterisk extension.

On the Lync side, I set up another SIP trunk which allows me to dial an Enterprise voice enabled test user from my Asterisk softphone. I'm also able to dial the Asterisk softphone from my Lync client.

I modfied my dial rule a little bit and am even able to dial '8' then the Lync user extension from my desk phone and establish a call. And vice versa: I can dial my phone extension from Lync and it rings my desk phone.

So far so good... but the next steps are where I seem to be having a problem.
I'd like to have it so when a specific DID is called, it sends the call directly to the SIP trunk and dials the Lync conference extension. At this point in time, I can't even get it to send the call to the SIP trunk. The call routing table will accept the PRI trunk as the destination, but not the SIP trunk... of course, even this wouldn't be ideal as the remote user would still have to dial the conference attendant's ext.

On the flip side, I'm also not able to call the outside world from the other side of my SIP trunk (my Asterisk softphone).

Any/all help with either one of these issues would be most appreciated. I feel like I'm so close, yet so far away!

[NTEDave]

Got a spare SLT port?

Point a DDI at the SLT port.

Set the associated outgoing extn on the SLT to your SIP trunk group.

Set a call forward on the SLT to the extn number of your conference assistant on your Lync setup.

Dial the DDI and theoretically the SLT port will perform a call forward to the nominated conference number via the SIP trunk group.

This might work 

[StarDestroyer]

Sorry, I'm still a little new with phone systems... what is an "SLT Port"?

[StarDestroyer]

Is it Single Line Telephone? If so, then I do currently have 2 ports free... though I don't see anywhere that I can set forwarding to an "external" number on that.

I just looked and it appears that the company that set this up created a phantom extension that forwards to a cell phone. It seems to me that I should be able to duplicate that setup for what I want to do here. Only trouble is, I don't see anything in the configuration of that "extension" that actually sends the call to the cell phone. I'm thinking it may be one of the features the installer hid from me *grumble* *grumble*.

[NTEDave]

There's two ways to set up forwarding on a Phantom or a port that doesn't have a phone plugged in.

Using the programming tool Click View then Online Monitor, this gives you lots of extra settings that you don't normally need to touch 

Go to the Phantom or SLT and you should be able to see an extra option called forward information, you can set forwards from this page.

The other way is to use the remote program feature code of 359, enter the phantom/slt extn number, password, forward codes etc.

[StarDestroyer]

That's awesome.... thank-you! (I don't have access to the Online Monitor so I used the feature code programming.)

Now it looks like the only problem I have to solve is being able to call an outside number from Asterisk/Lync.

[NTEDave]

How do you currently dial a Mitel extn from your Asterisk/Lync extns?

[StarDestroyer]

Currently there's a rule in Lync that basically sends all calls to Asterisk. Then there's a rule in Asterisk that takes any numbers in our extension range and makes them 7 digit numbers that match our DID numbers. It then sends them over the SIP trunk to Mitel. The SIP Trunk Group on Mitel is configured to go to Call Routing Table 10. The first rule in that routing table looks for numbers in our DID range and sends them to call routing table 1. Table 1 contains all of our DID destinations.

I have a second rule in call routing table 10 that catches any other 7 digit number and sends it to an application we use for unused DID's just so I can make sure I'm capturing that pattern correctly... but I can't figure out how to send that call someplace useful (like to the PSTN with the passed number dialed).

[NTEDave]

Create a DDI that routes to DISA, this should give you dial tone from the Mitel, you can then dial the trunk group access code and dial out over the ISDN30.

Unfortunately you can't just send a massive string of digits to the Mitel to activate DISA then dial out, there's got to be pauses in there.

[StarDestroyer]

I've gotta be doing something wrong, but I'm not sure what ... Everytime I set the destination to be DISA in my call routing table, that extension just rings the receptionist.

Though unless I can find some way for Asterisk to put the pauses in to automatically dial the number, it may not be much use anyway . The only other option I can think of is a seperate SIP trunk through a third party provider for outgoing calls. I was really hoping to avoid that, however, being that I already pay for a T1 PRI and have the channels to spare.

[NTEDave]

Do you have an E and and a + at the bottom of your table that catches calls and routes them to the receptionist?

If so your DDI for DISA received from Asterisk must not match the digits entered into the table.

[StarDestroyer]

I find myself as perplexed as ever by this...

This is a fairly black Call Routing Table that I just started using for this. I had a '+' entry that sent the call to voicemail (just so I'd recognize it if it came up). I tried removing it with no change in behaviour.

I've been mostly testing by trying to call my cell phone using its 7 digit number from Asterisk. I've put in various patterns in the Call Routing Table in order to match what I know to be coming from Asterisk. If I set the destination on a patern that matches my cell phone to "Single" and my phone extension, then dial my cell phone from Asterisk, the phone on my desk rings as expected. If I change nothing other than the destination to DISA, then the receptionist's phone rings. This goes for pretty much any "Single" destination.