We just set up a FreePBX on a Windows HyperV as a VM. Everything is configured and working. We started using the server for outbound calls using Softphones to ensure we would not have an issue when changing over to have inbound calls come to the new server
But are having problems now with outgoing calls they consistently drop at about 10 minutes into the call. It happened 4 times already where I’m on a call and then the drops. The truck stays active, but I can’t hear the other person, and they can’t hear me. I open the call again and then the same scenario happens
I would start by looking at your router/modem configuration for any timers set for 600 seconds on your udp/rtp connections then figure out why it decides to close the connection.
Then you will need to find out what “router” between the two endpoints are closing the connection at 15 minutesd, I very much doubt it is asterisk per se, so that leaves your VSP and the destination extension and possibly your trunk definition. Check with them
Edit, wait . . . . did you say you disabled “. …Enable consistent NAT…”
Its only been 3 phone calls so far I have not exactly timed it but it is somewhere around 10-15 minutes. I had an 40-50 minute call and I has to call back 2 times.
We are using Untangle for a firewall and also have an old Trixbox PBX with no problem. But that could because with it is an older version of Asterisk.
I timed a call now and it is 15-16 minutes when it drops. I’m using a softphone for testing so not sure if that is it. I will test with a hardware phone and see what I get. Diabeling SIP helper in Untangle did not help
I have tried a few possible solutions, from changing the UDP timeout on the router, to decreasing SIP registration time in an effort to force it to register more often. I still have calls dropping after 15mins.
I’ve been told to try session-timers=refuse as a possible solution.
Might not be helpful in your circumstance, but we use Cisco ASA firewalls with SIP inspection / processing enabled and things are flawlessly solid with several dozen users. Swapping firewalls is an extreme solution and only to be considered when all other avenues are exhausted, but having functional SIP inspection for NAT connections is A Good Thing (at least from a Keep It Simple, Stupid standpoint). Let the firewall do the work instead of playing games with quirky NAT workarounds.