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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.

You may need to make sure the button is configured to dial that star code.

@marc8lange Freeswitch supports multiple registrations. You may need to adjust your SIP profile settings to enable it. Although, unless you misconfigure something, it should work out of the box with vanilla installation. Check these threads...

Hi @marc8lange, we don't have an official restore script. The backup archive is best used for disaster recovery or off-site backups. A more straightforward way to restore your server is to install a new version of FS PBX and then restore the database from the backup folder. Our latest version of...

If you paste your FreeSwitch log here, we can help you. Without seeing any logs it could be anything like you said.

I agree @voipBull. I can't find the roadmap link. Can you share it?

OMG @voipBull, how long has this been around? This is groundbreaking. I watch new softphone offerings all the time, and I've never heard of this one.

The project is not maintained. It was written for older versions of both SIP and PHP. Without modifications and porting to new libraries, I doubt it will work very well. If the browser-based-phones project takes off, it will implement all required modifications.