asterisk users - Jul 2005 - Server side call waiting for SIP

Has anyone implemented call waiting on the server side for calls to SIP 
phones? I.e. where only one call is delivered to the phone, and the 
called party hears a tone for subsequent calls, and they can press a key 
sequence to switch between them, the switching being done on Asterisk 
rather than the phone.

On a related topic, if I were to implement it myself, is there a clean 
way to play a tone to an arbitrary channel from an AGI script? I could 
use the manager interface and redirect the call to a Playtones extension 
then back again, but a neater way would be good.

-- 
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/