There was some discussion on the IRC channel[1] about SIP and the re-invite option recently, and at first I thought if you did a re-invite on a call that was originally being handled through asterisk that you would loose the capabilities that asterisk provided once the call was moved to the phone accepting the call. But last night I asked Jim why he thought they'd separated the control and media channels in SIP -- and it dawned on me that I already knew why, I'd just forgotten the obvious. It's to support concepts like re-invite, where Asterisk holds on to the control channel so that you can still perform all your asterisk functions, and just transfers the media channel to the phone accepting the call.
I guess the one drawback (or is it an advantage?) of re-invites is that you can't monitor calls ;-) Hmm... but what if the DTMF is in band. Is supporting out of band DTMF a prerequisite for SIP re-invite? [1] http://irc.uc.org -- * Simon P. Ditner / ON-Asterisk Mailing List / http://uc.org/asterisk *
