Finally, I have my system running and I have a SIP trunk, one local SIP phone and a remote SIP phone. Al wors excep that I have:
In rtp.conf ports from 20000 to 22000
and my external phone y “correctly” connected with my ASterisk but not by 5060 if not by 64185, 1080 and something like that but my trunk is conected via 5060 and the local IP phone by 5062.
I can call but I hear a little noise.
In the router I have redirected the 5060-5080 UDP and 20000-22000 UDP ports to my Asterisk server.
Ah, in the configuration of trunk and extensions I have the port as 5060.
I think this is not normal.