[Asterisk-Users] SIP RTP

[Douglas Garstang]

Reinvite doesn't happen until after the call is picked up. After it's picked up, new invites' are sent and the phones communicate directly. Sorry, I forget the details. It was a few weeks ago.
 
Doug
 

	-----Original Message----- 
	From: Mike Hammett [mailto:asterisk-users at ics-il.net] 
	Sent: Sat 1/14/2006 7:59 PM 
	To: asterisk-users at lists.digium.com 
	Cc: 
	Subject: [Asterisk-Users] SIP RTP
	
	
	According to this page:  http://www.asterisk.org/doxygen/Config_sip.html
	 
	canreinvite=yes redirects just the RTP.  I was under the impression that the entire SIP connection got redirected, therefore losing accounting ability.  Could someone clarify this?
	 
	--Mike