[Asterisk-Users] SIP RTP

Mike Hammett asterisk-users at ics-il.net
Sat Jan 14 19:59:41 MST 2006


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
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