Loosing Audio From One End After 5 Min.
Hi all,
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?
3 thoughts on - Loosing Audio From One End After 5 Min.
1) Does it happen every time at the 5 minute work?
2) Have you done a dump on the client side to see if the NAT device is dropping the packets?
3) Is the phone behind a load balance internet connection and is the RTP
port changing?
Hi
Is the keep alive activated on the phone?
Just tested the connection in the other direction and when calling out there is no problem. only when calling in.
—— Original Message —-