I have been playing around with canreinvite=yes the past few days to see if I can stop using asterisk for media and instead get the audio directly from the carrier. If it makes a difference, I know for a fact that my carrier is passing audio from their carrier. I can also prove it from looking at the sip debug.
At what level (trunk, extension or both) does canreinvite=yes need to be set? And once it is set how can I tell if it works?
Thanks in advance
At what level (trunk, extension or both) does canreinvite=yes need to be set? And once it is set how can I tell if it works?
Thanks in advance