Sound goes off after few minutes into a call

I have an asterisk installation that I’m using for months now. During this week, I have a strange problem. After two or three minutes into the call, the other side doesn’t hear me. But I can hear what he is saying. Sound on one side goes out. What could be wrong? I don’t see anything special in the asterisk logs.

During the call, you can try running the “rtp set debug on” command in the Asterisk CLI to see if the media stops hitting your box. If it does, then the problem is likely with your firewall setup and/or ITSP. Did anything recently change with your firewall?

Its due to NAT. Use Externip to resolve this we had is issue in our environment.

Use it in [general] section of SIP.CONF

externip = ; Address that we’re going to put in outbound SIP ; Alternatively you can specify a domain
externrefresh=10 ; How often to refresh externhost if
localnet=; All RFC 1918 addresses are local networks
localnet= ; Also RFC1918
localnet= ; Another RFC1918 with CIDR notation
localnet= ;Zero conf local network