[Asterisk-Dev] SIP INVITE vs TO URI

[William Lloyd]

I'm trying to integrate Asterisk to work with a SIP provider.   
Outbound calls are not an issue.

It's inbound with DID's I'm looking at.

In sip.conf when I register the provider without an extensions like
register => 1XXXYYYZZZZ:password at sip.provider.com:5060

Asterisk registers at the provider with the 's' extension.

the problem comes with the provider and the way they setup the user  
accounts.  They consider you to only have 1 account, not 2.  So the  
pilot number of 15145551212 is the only one that it accepts  
registrations from.

For example say you have 2 DID's with this provider
15145551212 and
15145551213

you would register with
register => 15145551212:password at sip.provider.com:5060

but when there is an incoming call from the provider for 15145551213  
the provider passes the destination in the TO field of the SIP header  
not in INVITE.

The provider sends the destination as part of the TO field.  In the  
example the asterisk host is behind NAT.

For example
INVITE sip:s at 192.168.100.20
TO: <sip:15145551213 at 69.23.45.200>

or

INVITE sip:s at 192.168.100.20
TO: <sip:15145551212 at 69.23.45.200>

Both get handled by Asterisk s destination.

I've been looking at the get_destination and handle_request_invite  
code in chan_sip.c to modify this behavior when the INVITE URI and TO  
URI do not match.

In a quick read of the SIP RFC's this providers implementation seems  
to be valid.

How would be the best way to handle this addition in the asterisk code?

Ideally I'd like to approach it in a way that would be accepted back  
into the base asterisk code.

thought?  suggestions?

-bill
wlloyd at slap.net