[asterisk-users] DTMF forwarding and Page

[Matteo Fortini]

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