asterisk users - Oct 2010 - SIP disconnects after 20 seconds behind NAT

Hi,

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

Am 13.10.2010 19:50, schrieb Ahmed Ossama:
> Hi,
> 
> 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
> 
Hello ahmed,

sounds like the typical SIP ALG problem. Just configure your firewall to
do stupid plain nat and dont touch the sip headers. As you could see
this doesnt work.

if you turn on sip debug you will see several retransmits for the 200 ok
message which comes at the real beginning of a call (when you answer the
phone) cause the ACK package to this 200 ok could not be received.

same to Bye at the end of a call.

Best regards

Stefan Schmidt