Problem probably with NAT


#1

Hello. I could configure asterisk and redirect my internet server to call from a FWD soft phone to a PSTN phone (X100P) and from a PSTN phone to a FWD soft phone correctly.

  1. FWD1(remote PC) ---------> FWD2 (Asterisk) --------> ZAP

  2. ZAP ----------->Asterisk ------------> FWD1(remote PC)

But suddenly asterisk didn’t leave me speak in any way.
I’m using Pulver Communicator and i can call, asterisk answer the call but strangely if i call logged out i can hear the “demo-congrats” message but
if i call logged in i can’t hear the “demo-congrats” message and i can’t dial numbers to establish the communication that i want in any way.
I don’t know what the problem was because i didn’t change the configurator files after i could call and speak.
i need some help.Thanks.


#2

first, in sip.conf set localnet= externip= and set nat=yes. Then in your router, forward UDP ports 5060 and 10000-20000 to your * box. Does that do anything?


#3

I did it and it ran but now it doesn’t. I’m sure i didn’t change any file after it ran…