I am from cisco voip background (still a novice :-)). In cisco, Callmanager is used for signalling only, the actual RTP media flows between end points and not through callmanager. When i was reading Future of telephony edition 2, author said that media will flow through asterisk1.4. However there is parameter, when configured will allow end points to talk directly. He said it was experimental. What is the current situation in Freepbx ? can someone kindly highlight it ?