Newbie question SIP and NAT

Asterisk Asterisk Support
1

I have a Fedora 20 box with two NICs. NIC A is connected to a local network and has a 192.168.3.1 address. NIC B is connected to the Internet and has a DHCP address handed out by our WAN. I have a phone connected on the 192.168.3.x network and it has a SIP calling application. I have a Media Gateway out on the internet that is going to hook up the SIP call to the PSTN. Currently SIP calling doesn’t work. I was told that I have to have global IP addresses on both ends of a SIP call for the call to work. Can I configure Asterisk on the Fedora 20 box to translate the call parameters? I was thinking that since Asterisk has access to both NIC A and B on the Fedora box it could translate the call across NAT…

Thanks in advance.
Y-

2

Look at exernip, extenhost and stunaddr, together with localnets.

3

I am new to this type of configuration. Is there FAQ or HOW-TO on setting up Asterisk to forward or proxy calls across NAT? So I assume that Asterisk will take the SIP call and convert it so that it has an originating address on NIC B, right? I will read the docs on exernip, extenhost and stunaddr, together with localnets.

Y-