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().