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Make sure FreeSWITCH is selecting this route when the fax call goes out. I have the service running smoothly for clients that send thousands of faxes every day.

You need a route that is prefixed with 9999. See the screenshot attached. To separate regular routes and fax routes, we use a prefix.

Are you entering a US-based number or another country? I can check if I get the same error and fix it for you?

It's in this directory. You are correct. It's hidden. Run this command, and it will open. nano .env

FS PBX (https://github.com/nemerald-voip/fspbx) We are currently developing the API. It should be released soon. FS PBX is a forked FusionPBX with lots of additions targeting specifically external integrations

I sent you a private message regarding the rest of your questions.

I already have 95% of it built in the FS PBX Contact center module.

What specific features in FOP2 are you looking for that are not available in FusionPBX?

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.