Hello
When I make calls from home to the PSTN by going through the Net ->
Asterisk -> the Net -> VoIP provider -> PSTN, I get no sound either way. I
assume it's because I must tell Asterisk to use fixed ranges of UDP ports
and map ports accordingly on the NAT firewall under which it is located on
the LAN at work.
Here's the schema:
home > NAT > Internet > NAT > Asterisk > NAT > Internet > VoIP provide >
PSTN > callee
I took care of the NAT at home by using fixed ports in X-Lite + used STUN,
so I guess the problem is located on the Asterisk side.
1. What are the settings (in sip.conf?) to tell Asterisk to use specific
ports for RTP?
2. With this kind of setup, does Asterisk stay in the loop to forward RTP
packets, or do X-Lite at home and the VoIP provider send RTP to each other
directly?
Thank you.
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