On your router/firewall, you will need to open port 5060 and forward
it to your Asterisk box to allow SIP messages through. You will then
need to open and forward your RTP ports to allow audio through (the ports
are configurable in rtp.conf. See chapter 3 for more information).
The default RTP ports are the range of 10000 -> 20000.
In your sip.conf, you need to add three lines to your [general] context.
[general]
port=5060 ; make sure you have this line
externip=my.domain.com ; this can be either external IP address, or FQDN
localnet=192.168.0.0/mask ; local network your Asterisk server is in, plus network mask.
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