On 1/14/06, Mike Hammett <[EMAIL PROTECTED]> wrote: > 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? >
This isn't correct. While RTP goes away on a successful reinvite, Asterisk never gets out of the middle of the SIP signaling path because chan_sip is a B2BUA and not a SIP proxy. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
