It just re-directs the RTP stream. The SIP stream still goes through *.
Mike Hammett 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?
--Mike
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