SIP Attended Call Transfer with RFC3891

Hi

I would like to know if Asterisk PBX supports SIP Attended Call Transfer method using RFC 3891 (Replaces Header)?

tech-invite.com/Ti-sip-service-5.html

If it does support this supplementary service, what is the call flow (message sequence diagram) when Asterisk is used?

Regards
Saurav

google is your friend :

asterisk.org/doxygen/chan__s … ource.html