[Asterisk-Dev] sorry about this: sip header/sip_pvt questions:
nwong at kancharla.com
Tue Jun 14 08:31:01 MST 2005
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
What variables does transmit_invite use in p to generate the request To:
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?
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