There are lots of sip phones which do not support conference calling and I would like to know if asterisk can support that (i.e. perform the mixing) on behalf of the phone? For example if
RTP
A <------ * ----------> B
and then C tries to call B where the B phone can support one call at a time, would * support:



C \
\
\ RTP
\
* ----------> B
/
/
/ RTP
/
A


thanks
moe smadi
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