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Hey everyone, We’re excited to announce the release of FS PBX version 0.9.57! This update brings a major new feature that’s been highly requested, as well as a round of bug fixes to make your experience even smoother. Polycom ZTP Integration Is Here! With this release, you can now fully...

Hi @Raven, by default, everything is preconfigured to access the system from an external IP. You can follow this two articles to configure access by domain name or with your internal IP address. https://github.com/nemerald-voip/fspbx/wiki/Troubleshooting-Error-419-(Page-Expired)...

As far as I understand, this isn’t possible. The stream in FreeSWITCH is continuous, meaning you can join or leave at any time, but you cannot pause or rewind it. It functions much like a traditional TV channel, which plays whatever content is live at the moment you tune in.

In FusionPBX, it's a member-only feature. You have to pay for support to get access to API

You can navigate to Advanced -> Provision Templates to view all available templates. If the menu link is missing, download the latest updates cd /var/www/fspbx git pull php atisan app:update and then run this command to refresh the menu links php artisan menu:create-fspbx Log out and...

@jimfIre17 You can theoretically use any SIP client. The most popular apps that can be used as a generic version and also white labeled are Acrobits and Ringotel.

@whut The original question was about call transcriptions. Not voicemail transcriptions. I agree that voicemails are short and there is almost no delay in returning a result. I would still have a queue for that as well. However, when you transcribe an hour-long conversation, there will be some...

what code do you have on line 115 in this file? /var/www/fusionpbx/resources/classes/config.php

Thank you for reporting this. I tried to push out the fix as soon as possible.

You need to point the phone to your provisioning server. It needs to know where to go and download the config

It's probably doable with some custom LUA script and a database table where you can keep the records of who called whom. You would need to hire a developer. This is a dynamic system that will change daily. If you want something simple, you can assign direct phone numbers to each person and...

You for sure need PHP 8.1 to start. 7.4 will likely to have errors. There might be other things you will need to adjust.

I would be very careful with that module. No proper queues have been implemented yet to process transcriptions. If you happen to have 1000 pending transcriptions, it will overload the server. I hope to see FusionPBX to have that fixed soon

It's usually passed over HTTPS, which is secure, but if you want even more secure, have a private network, and then the data will travel without passing through the public internet.

Are you using FS PBX?

I have a script that pushes recordings to AWS servers. It can easily be repurposed for any other servers. It's for FS PBX though. Not for FusionPBX. FS PBX is a fork of FusionPBX

CASCADE requires special handling when the database is restored. For an easier restoration process, developers of FusionPBX opted not to have any CASCADE records. At least this is my best guess as to why they did it. Very good script by the way. Thank you for sharing.

We can do a Zoom call too, so I can show you around. The system has so much to offer, it's hard to describe in forum posts and a few videos that we have.

you are missing something. The caller ID needs to be assigned to the extension and the setting I posted above also has to be added. Restart your gateway after changes and just for good measure, flush the cache

Did you add your provider to ACL list? Also, flash cache, reload XML?