I have my FreePBX appliance connected to my Avaya Session manager via SIP as a trunk for outgoing call processing. It works great, except I have had two incidents over ~8 days (thousands of calls in between) where the connection shows as “OK” on the Asterisk side, but not connected on the Avaya Session Manager side. Almost like my 5060 port is closing and I need to reboot FreePBX to resolve the issue (losing pending and active calls).
In my forum crawling, I noticed people indicating that the issue could be an issue with qualify and the default of 60 seconds being too long. Below is a copy of my trunk outgoing SIP settings:
It’s pretty basic. Does this look correct? Is anything missing?
The last question I had was if there was a way to have the asterisk trunk status have a more accurate reflection of what is going on (versus just showing as “OK”)?
Thanks for your time and any guidance you can provide.