Just installed Asterisk 13 on CentOS 7 and have run into a problem.
The Scenario is this:
Asterisk is on the internet the Phone, a D40, is behind NAT
So someone calls the number and Asterisk routes the call to the D40
Everything works fine and the call is established, but then after 5 min. the caller stops getting audio from the D40 but there is still audio to the D40.
using both RTP and SIP debug on the Asterisk console does not reveal anything. Actually I can see from the RTP debug that RTP packages are send and received even after lose of the audio.
So does anyone have any ideas where to look for the problem or perhaps a solution?