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