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

[Jean-Hugues ROBERT]

Hello,

It is not obvious to do I guess.
If the channel to the SIP phone is in a Dial commmand in
the dialplan, one complex solution would be to redirect
it to a Meetme me room in quiet mode and then play a call waiting tone
in that room (probably using some local channel dialed
into the meetme room and used only for playing media in
the room). When the user should switch, things would be
more or less complex depending on what happens to both
calls (the first & the second). It get worse if there
are multiple calls waiting.
I would be interested to hear about any  simpler solution.

At 18:28 16/07/2005 +0100, you wrote:
>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/
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