I have a new scenario of remote extensions that I am trying to fix, I have an elastix pbx 1.5 at my home with a static 20/5 Mbps internet connection. I have 5 static ip's one of them is used for the router and the other 4 are for other boxes, One points to my pbx and I use sip trunks for pstn. if I leave sip_nat.conf blank my two remote ip phones will register along with stun = true, and I can make calls out just fine, but cannot extension dial from IVR to the phones. This is a problem. So I remembered to add the statement in sip_nat.conf of externip, localnet, and nat = yes. Once I restart asterisk now I can extension dial but with no audio both in or out, and can make calls out but no audio cant even hear the dialing tones. They remote site has the same internet service but its dynamic instead of static. Would this make a big difference. Its a different scenario for me compared to last time. But maybe not, its behind the same natted firewall that my pbx is just dynamic instead. When I do sip show peers I see the remotes using 2054 and port 49152 and I know I have only 10000-20000 udp to my pbx in my router. I have tried every other solution I have found in the forums and have had no luck. The last time I got it to work with 1.3, is something different in 1.5 that I would need to make another change to get this. Thanks all.