[Serusers] SIP to SIP calls & Asterisk

[Patrick Baker]

Hello all,

I've ran into a dilemma regarding on the call structure is setup for
my system right now.  As of current everything goes through asterisk
ie

sip user -> ser -> asterisk -> sip user

what I want to try and accomplish is sip user -> ser -> sip user.

I believe this would remove unnecessary load on asterisk servers and
just connect the call directly.

I'm having a hard time understanding how I will do this thought.  as
of right now I have a forward statement

if (uri=~"^sip:[0-9]*@.*") {
               forward( 10.0.18.3, 5060 );
               break;
       };

Say I have multiple companies, how would I setup extensions to call
sip devices and if I wanted to dial into a sales queue how would it
forward to asterisk.  Another thing would be voice mail...how would
the extension know to goto voicemail after a certain amount of seconds
and play a custom greeting that they assigned for their box.

and how would they be billed... sip to sip would be billed thru ser,
all zaptel channels thru asterisk??


Best regards,

Patrick