[Asterisk-Dev] Fwd: Server side call waiting for SIP

[Alistair Cunningham]

Has anyone on Asterisk-dev implemented this?

-------- Original Message --------
Subject: Server side call waiting for SIP
Date: Sat, 16 Jul 2005 11:51:34 +0100
From: Alistair Cunningham <acunningham at integrics.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users at lists.digium.com>

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/