[asterisk-dev] SIP question

[Mr. Jones]

Hi Folks,

I'm trying to roll out Asterisk as a hosted PBX for a client. They've
selected a inbound origination provider (Provider B) for DID service.
They also have an inbound provider (Provider B) for toll-free
originiation.

Asterisk seems able to determine the dialied number (DID) for Provider
A automatically, but not for Provider B.

In digging into the SIP headers between two providers and  I've
noticed the following.

The first sample is from Provider "A" and works:

To: <sip:8005551212 at 66.123.123.6:5060>;tag=as6519d118

I also noticed the "Contact" header seems to be filled in:

Contact: <sip:8005551212 at 66.123.123.6>

>From provider B:

To: <sip:3125551212 at 66.123.123.6;user=phone>;tag=as7154d7ed

Contact: <sip:s at 66.123.123.6>

So I guess I have 3 questiosn:
1. What sets the "Contact" field - Asterisk, or the sending machine?
2. Would the ;user=phone and the lack of a port number be causing
Asterisk not to set the exten properly?
3. If the Contact is not being sent by the sending end is there anyway
Asterisk can use the SIP To: field to determine the extension?

Currently I have a macro that parses the To: field and passes the
dialed number into another macro which rings the appropriate
extension.

This isn't quite as clean as I'd like.

TIA!