Noone knows that? Where/whom could I ask?
Thanks
Il 10/02/2012 12:30, Matteo Fortini ha scritto:
Hi,
I'd like to implement some way of controlling remote SIP clients while
in a call, to execute remote commands.
The call topology (think of a PA system) is this:
* the caller is in a MeetMe() conference room
* the callees are Page()d, then the dynamic conference room is
connected to the previous one
I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the
caller to the callees. I found option 'F' for MeetMe, but I have no
control on Page().
TIA,
Matteo
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
