Alistair Cunningham
2005-Jul-16 03:51 UTC
[Asterisk-Users] 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/