[Asterisk-Users] Server side call waiting for SIP

[Alistair Cunningham]

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/