Please correct me if I am wrong. With PJSIP there is no way for Asterisk to stay a OUT of the media path, while with the old SIP channel, using directrtpsetup and directmedia, it just works. The issue I think is that other servers do not accept reinvites or updates to redirect media, so PJSIP will not be able to step out ever. Using the old sip channel, the 200
OK with SDP tells the calling side to talk direcly to the other side. Is there a way to do this with PJSIP?