I have a small FreePBX system that I am running. We have 4 POTS lines and sip lines. When we use the POTS lines for outbound calls it takes about 8 seconds for call termination. When i change to SIP lines, the termination can range between 30 to 60 seconds. Any idea what could cause this problem only or SIP calls?
In Settings ----> Asterisk SIP Settings, do you have anything entered for the STUN Server Address? I had extremely long pauses before SIP calls (even endpoint to endpoint) would connect when I had a STUN Server Address entered. Apparently, my firewall (external device, not the PBX) was causing some issues with it. I removed the STUN Server Address and all went back to normal.
Ill be… I removed the STUN server address and it all went away. Thank you very much for the help.
Any idea if the STUN server is actually needed?
I’m not sure if the STUN server address is needed. I have not experienced any problems without it though.