[Asterisk-Users] SIP DID routing

[snacktime]

How do you get the called number on incoming SIP calls?  I've never
had multiple DID's via SIP from one provider before and somehow I
never realized that with IAX it just works, and SIP is different.

If I don't set an extension in the register command the incoming
invite has <sip:s at me.com> in the To field.  Now if I have multiple
DID's that I want routed to different extensions, what's the solution?
 Is there a SIP header that is normally used to pass the called number
in?

Hope that makes sense..

Chris