SIP Trunk between Avaya IP Office and Asterisks / TrixBox

I figured I'd post this HowTo on how to setup a SIP trunk between a IP Office and a Asterisks Phone system for intra extension dialing because I just spent the last 3 days trying to figure this out, and there seams to be plenty of articles on how to do this via H323, but there are very limited docs/HowTo's on doing this via SIP.

At the end of the day, this was a lot simpler that I expected, but I learned alot in the process.

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Avaya IP Office Side:
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1) Create a new SIP Trunk (SIP Licenses are required for this)
2) The only thing you fill in here is the IP address of Asterisks/TrixBox. Please note that this config is done anonymously, so I assume the two machines are either on the same LAN or connected securely via a VPN, I would not recommenced this setup if you are doing this over the internet.

ITSP Domain: <blank>
IPSP IP : Asterisks IP
Prim Auth / Pass: <blank>
Eveything Else: <blank>

Check "in Service" box

3) Setup SIP URI - ADD a channel and set this to <use user Data> - Make sure to set the groups to something unique. I used group 420 for incoming and outgoing.

4) The other tabs can be left default settings

5) Create a short code for calling the Asterisk Box. The extension on my asterisks are 4xx. So my SC looks like this:

Code: 4XX ( Change this with your extension format )
Feature: Dial 3k1
Tel Number: 4N"@x.x.x.x" (Replace with IP of asterisks)
Line Group : 420 (Group Id Set in URI)

6) Create an Incoming Call Route Like this:

Bearer: Any Voice
Line Group : 420
<the rest you leave default>

Under Destination
Default Value is . (just a period)

7) Under Each User, make sure to set their Sip Name to their extension number under there SIP Tabs.

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ASTERISK SIDE VIA FreePBX GUI
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1) Create a SIP Trunk that looks like this:

Trunk Name: IPO
Peer Details:

host=x.x.x.x (IP of IP Office)
type=friend


2) Create an Outbound Route
Route name: IPOffice

Intra Company Route <checked>

Dial Patterns : 2XX ( Replace with the format of your IP Office extension )

Trunk Sequence: SIP\IPO


3) Under General Settings
Set "Allow Anonymous Inbound Sip Calls" to yes

That should be it, you should now be able to call back and forth between the 2 systems as if they are one.