Hey guys, i’m new, and was noticing that when people called me via sip to sip client, that i could answer the call, but as soon as i answered it, it would disconnect. Then i noticed that if i set their clients to use a stun server at stun.example.net and opened all ports 10,000 to 10,024 on my firewall, that i could finally hear their voices.
I’m not sure which one is working. All of the sip clients were originally set at port 8,000 for rtp, so i’m not sure that i even needed to open the ports. Also, i was wondering if Asterisk did STUN for the clients. I do not like using a third party’s stun server. So i was wondering if anyone knows how to get this to work with asterisk.
Oh, here is my setup. My computer has MANY servers, and asterisk is one of them. It is behind a firewall/router and i also use an sip CLIENT from it. The rest of the computers are different places in the world, and all behind routers. So, what would i have to do to get this working without 3rd party stun? I seen that there is a stun-server for Fedora (which is what i use), but i am still trying to find it’s configuration files, and process name.
Thanx!