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Use small letters in the search. It should find it. I'll update the instructions

Here are the instructions on how to set up DISA for FS PBX: https://www.fspbx.com/docs/additional-information/disa-setup/

You may need to explain to Freeswitch that it is behind NAT. Add these variables. Flush the cache and restart FreeSwitch. Adjust your SIP profile accordingly.

I got it working on my system, but had to push a small fix to make the process user-friendly. SO make sure you have the latest commit. Stand by while I'm writing the article with instructions. I'll be back shortly

Let me take a look at this and get back to you. I want to make sure it's 100% functional.

FusionPBX and FS PBX are exactly the same when it comes to SIP profiles. As a matter of fact, SIP profiles are a Freeswitch thing and have nothing to do with either FS PBX or FusionPBX. By default, your external profile uses port 5080 unless you change it. Adrian is correct in pointing out that...

This is a public forum. Don't post your credentials here. Remote sessions are reserved for customers subscribed to a support plan. I'm happy to do best effort support on the forum for free, but you need to provide the required logs.

It's fine to use the log viewer, but just don't paste all 59000 lines. The log rotates when it fills up, allowing it to store weeks' worth of calls. If you can isolate just the call in question and paste it here, it would be best. I personally prefer FS CLI because I can start it before the call...

There are 59000 lines in this log. Please isolate the log for one call. I can't possibly read all of those lines. Turn on FS CLI right before the call and turn it off after the call ends. Then copy and paste only those lines.

Calls failing are usually a Freeswitch or dialplan issue. Please post your FS CLI log when making a call, and I'll try to help you.

This is very likely an issue with NAT and Docker. Try testing WSS from an external service and see if you have the same problem.

Yes, you are missing IPv4. Likely misconfiguration in Variables or SIP profiles. It can't properly start. If you watch the Freeswitch CLI while reloading sofia you will see errors that might point you in the right direction.

Use late negotiation. It will negotiate the codec supported by both sides. They will probably decide on G. 711. No transcoding will be needed and it will free up your CPU.

Do you see any activity in Freeswitch CLI?

Did you flash the cache and restart FreeSwitch after making these changes?

If traffic is directed to your box, it doesn't yet mean it reaches it. You need to run SNGREP to confirm that. Once you confirm, then check FreeSwitch running as @DigitalDaz suggested. Also, check the ports it's listening on.

In your crontab, you should have this job scheduled to run. * * * * * cd /var/www/fspbx && php artisan schedule:run >> /dev/null 2>&1 That's all you need to have all FS PBX jobs run. If backup is enabled in the Default Settings, it will execute every night. You can post your restore job here...

They switched how the active calls page works in the recent version to web sockets. It requires some setup to get it going. Search this forum for posts about it.

Did you check Iptables? Is the port open? Also, check SNGREP to ensure the traffic is directed to your box.

If you end up with a final result at some point, send me a pull request to add the script to the repository. Others may find it very useful. Thank you for sharing.