Unlimitel's setup is quite stable, just that you have to get the right parameters set up.

The sip Proxy , and the RTP feed originate from two different IP addresses, making the use of Unlimitel with a NAT a little challenging.

1. If you are using Nat, just make sure that you have your RTP ports (/etc/asterisk/rtp.conf) forwarded from your Gateway to your Linux Box.

2. In sip.conf, create yourself a registration entity.
register => XXXXXXXXXX:myBigSecret@sip.unlimitel.ca/XXXXXXXXXX

3. In sip.conf, create yourself a peer definition for your calls
unlimitelXXXXXXXXXX
type=peer
auth=md5
username=XXXXXXXXXX
fromuser=XXXXXXXXXX
fromdomain=unlimitel.ca
secret=myBigSecret
host=sip.unlimitel.ca
port=5060
nat=no
canreinvite=no
qualify=no
disallow=all
allow=ulaw
allow=g729
dtmfmode=rfc2833
insecure=very

4.In your extensions.conf
Create yourself an extension for XXXXXXXXXX to handle the reception based on your default context, and voilà, it works!

I've been using them for about a month now, and, when using uLaw, the sound is simply amazing. Obviously, uLaw is bandwidth intensive, so, if you can support G729, my suggestion is to use it.

They do not support RE-INVITES, so, therefor you cannot redirect the media stream. So, if you want asterisk out of the media path, you can only do it within your local extensions. This is a known bug, and there is a software revision in the works for the solid-state switch, that should hit production in the next month or so.

Cheers

To have calls come in a specific context, make sure you a section like this in your sip.conf — it has to be the _last_ section where host=address.of.sip.server matches the address.of.sip.server used in the register statement (i.e. below all your outbound sections to the same sip server, if you have any).

{unlimitel-in}
type=peer
host=address.of.sip.server
context=unlimitel-inbound

Then, make sure you have a context like this in your extensions.conf

{unlimitel-inbound}
; Calls from Unlimitel land here
exten => 6130000000,1,NoOp(Whatever you want to do here)
exten => 4160000000,1,NoOp(Call from Toronto DID)
exten => _NXXNXXXXXX,1,NoOp(No Service at this number...)

Note: replace curly with square brackets

...one more thing.
If their SIP server calling you is not the same as the one you register to and create an entry for in SIP.conf, * won't match it to the right entry in SIP.conf, hence won't be able to figure out what context to send the call to.
Just enter a default context in the general settings, and use that context for all incoming SIP calls.

This looks like a classic NAT problem.
Try these settings:

canreinvite=no
nat=yes