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This was built on a separate branch and then added to the master branch. The master branch is the development branch. New installs haven't used the master branch for a couple of years. They use the release branches. The current branch used by the install at the moment is 5.4. Previously, we...

Thanks for reporting this issue. Finished checking all other Lua files. Alex, a member of the FusionPBX, saw your message and relayed it to me.

The issue has been fixed and committed. Checking all other Lua scripts to ensure there are no others with a syntax issue.

Matt, I'm taking a look at this now

Use the device edit page, and you can download the files needed. - Save provisioning config files using the drop-down link. - Move the files to a TFTP server on the same network as the phones. - Use DHCP option 66 to point the phones to your TFTP server.

@mdoupe, you can reach me with information here to discuss this further. https://www.fusionpbx.com/support

Bug reports should be in their own thread. So they can be discussed on their own. The user_record dialplan is not a bug. You can review the complete user_record dialplan here. https://github.com/fusionpbx/fusionpbx/blob/master/app/dialplans/resources/switch/conf/dialplan/050_user_record.xml...

Yes, that is what they are doing. I'm happy it's working for you.

Go to Advanced -> Default Settings and search for sip_to_user. Change the value to sip_to_user if it is not already that value.

This means you need your inbound routes to use a different variable. Search for sip_to_user, and that will find a variable in the dialplan category. With the name of the destination. Edit this and set the value to sip_to_user. Then press the RELOAD button or log out and back in, then go to your...

The provider's ACL is for your carrier's IP addresses. The domain acl was renamed to providers to attempt to make it more intuitive as to what belongs in that Access Control List. Adding your subnet to rfc1918 list is usually not needed. In most cases, this is just left with the defaults. If...

Update your version of FusionPBX to get the latest source for version 5.4 or the master branch. Then resave you fax server and it should work.

The screenshot shows where it is configured wrong. My previous message said this The FAX Server Extension number is 18801. Now set the Destination Number to 18801. Then try calling it again. You should get FAX tone on your next test after you correct the setting.

UDP has a limit for the header size of the packet. SIP can easily exceed that limit. When the limit is exceeded then the packet doesn't reach its destination.

On your fax server what is the extension number and destination number? In most cases they should match with the same value. Have you tried calling the fax server extension number from a local extension on the same system and domain to see if you get a tone?

You should definitely hear a FAX tone. You can try calling the FAX extension number from a registered phone on the server. Then you should hear FAX tone. Advanced -> Modules page, you should see one that says SPANDSP. This module is required for FAX.

If you are directing one of your numbers to a FAX Server I recommend using a dedicated phone number for FAX. When using a dedicated phone number for faxing then you can skip fax detection. The correct FAX settings are dependent on your VoIP provider. You should look for their documentation on...

Yes, that looks correct.

The new active calls works but require setting up the websocket configuration in NGINX. Look for the websocket section at the following link. https://docs.fusionpbx.com/en/latest/advanced/version_upgrade.html Make sure you are using the latest code on the git branch you are using. If you are...