-
-define and change dsp conference ids
-
make asterisk call implementation
-new interface.conf
-
-reduce mixer
-
-call recording
+new interface.conf (add remove ports by admin)
call to multiple endpoints (extensions)
- application process (action)
- bchannel control (tones, dsp, filter, activation/deactivation)
-sip raus, h323 raus
-
avoid disconnect-collision (release if disconnect from both sides)
+display message during nothing/play
-
+Port -> Channel
+Call -> Link