[Asterisk-Users] three way mixing / conferencing

[m. smadi]

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