Sip Calls disconnect after 2 minutes

Hi I have latest distro installed. Every time I make a call to phone passed through a sip trunk, the call terminates after 2 minutes. I noticed in the astersik:
[2012-10-04 16:10:15] NOTICE[3625]: chan_sip.c:25925 check_rtp_timeout: Disconnecting call ‘SIP/35544520512-00000ee6’ for lack of RTP activity in 31 second s

Is the problem in the freepbx distro or I should call the sip provider?

The problem is probably the NAT in your firewall.

What kind of firewall? Make sure you have consistent NAT turned on (so the firewall doesn’t randomize ports), and also make sure that the UDP timeout on the firewall isn’t shorter than the timeout on the Asterisk server (default is 60 sec, my Sonicwall defaulted to 30sec).

I want to use the distro only for extentions within the lan not outside. I have 2 trunks with real ip-s.
Should I configure it Nat and public ip ?!
we have a static public ip. but I don’t want freepbx to be shown outside except for the 2 trunks which I register with publick ip 80.xx.x.xx