[Asterisk-Dev] sorry about this: sip header/sip_pvt questions:

[Nicholas Wong]

Buncha quick questions that I hope someone bothers answering:

I'm trying to forward a call to a new number based on a response.
handle_request grabs response and handles the changes to p: then...

What header routes the call? Contact: or To:? Both?

What mode does sip_pvt need to be in to transmit an invite, or should I
just call transmit_invite? AST_CONTROL mode need to be set to something
special?

What variables does transmit_invite use in p to generate the request To:
field?

I keep ending up with a To field that looks like this
To: <sip:number<number at xxx.xxx.xxx.xx>>

so obviously I'm putting too much into p->tohost...


Any quick answers? Or am I doomed?

Nick Wong