[Serusers] SIP to SIP calls & Asterisk

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