[asterisk-users] SIP disconnects after 20 seconds behind NAT

Ahmed Ossama ahmed at master-zone.net
Wed Oct 13 12:50:25 CDT 2010


I have an asterisk server sitting behind a pfsense firewall, I have 
successfully configured pfsense for NAT traversal, and clients from the 
internet can call clients inside the network of asterisk, as well as 
other clients registered with this asterisk server on the internet.

The problem now is when a client from the internet do a call, the call 
disconnects in 10~20 seconds, but during this period the call goes fine 
and voice is heard on both ends; But when a client on the same network 
of asterisk calls another client registered from the internet, the call 
is established without any issues, and it doesn't disconnect.

I have also noticed that when internet clients do calls, and the call is 
established on both ends, if one of the two parties hang up, the other 
end isn't notified and the call stays opened at this end.

I could provide config files if needed.

Please advice about resolving this issue.

Ahmed Ossama

More information about the asterisk-users mailing list