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Well, I jumped the gun yesterday. I started receiving numerous reports that speech generation had stopped working. After looking into it further, we found that the old models created by OpenAI appear to be throttled. This isn’t official, but it’s what we discovered. Generating a 20-second...

There are no plans to drop OpenAI support for voicemail transcription or speech generation. Could you point me to what in the repository gave you that impression? I’d be happy to clarify. The current integration works well, and switching to newer models would increase costs by roughly two to...

Ah, that's too bad. I don't use them personally, but I know a lot of folks who do.

A small correction. I think @kenn10 posted the wrong URL. The official upgrade guide for FS PBX is here: https://www.fspbx.com/docs/updates/ The URL above is for upgrading FreeSWITCH; if you decide to run it, you'll have their bug fixes too.

You need to be on one of the recent versions and follow this guide for it to work. https://www.fspbx.com/docs/configuration/contact-center/agent-presence-buttons/ Pressing the button should work unless you don't have one of the recent updates that replaced the old dialplan pointing to...

Sharing a phone number doesn't work well and will give you more frustration than it should. I recommend using different numbers for calls and faxes

No T.38 negotiation occurred on this call. I do recommend T38 for faxing. In my experience, it always renders better results. In your case, I would address the decline code first and then try switching to t38. The decline code is 603, but this log does not reveal the specific reason or whether...

@yaboc maybe start by posting some logs with failed faxes. We can't magically know what the issue is. So any solution would be a guess. I have a guy who sends 30K pages a month from FS PBX. So the fax system works very well but always requires some tweaking based on individual cases and provider...

In the caller ID tab, you can set the external caller ID for each extension. Internal caller ID always matches the extension number.

You need to use variables from the Default settings there if you want to override them. The Default settings also set the SIP port, transport, and expiration. You don't need a custom template for that. See attached screenshots:

I was too scared of updates and tired of constant bugs in the past. It cost me a few clients...

Check the generated inbound route and make sure it uses sip_to_user. In my screenshot below, I'm using the default sip_req_user. From there, you just need to make sure the regex matches the DID being passed in the header. Reach out to FS PBX support if you get stuck.

You did restart FreeSWITCH after the update, right? Does this happen every time you restart FreeSWITCH? Or was it a one-off issue after the upgrade?

I found your issue 404 not found 199.87.144.69 destination_number CallCentric-1777 sip_to_user CallCentric-DID sip_req_user gw+0ef24007-6de9-4969-951d-f267f96d0831 I assume CallCentric-DID is the real DID you are dialing, and you just replaced it for privacy. So that means CallCentric is...

Attach screenshots so we can see what you are doing. The rule remains the same. Do not modify inbound routes manually. Create a phone number, and it will create an inbound route for you. In your case, it's best to delete phone numbers from the inbound routes and phone numbers page and start from...

Awesome to hear that @kenn10. Thank you for giving it a try. Don't forget we use our own fork of FreeSWITCH, and it includes bug fixes we've discovered over the years. We've had many conversations with FreeSWITCH support about adding those to their main repo and have submitted PRs, but they've...

FS PBX v2.0.1 now supports FreeSWITCH 1.11.3. Upgrade FreeSWITCH separately from FS PBX, following our official guide.

@kenn10, we discovered that almost immediately. Even before we merged the updates into our fork. This is one of the things we are testing to make sure the update goes smoothly. We always read the docs carefully, along with the code itself, to understand what is new and what's required.

We have started our testing environment. It's not ready yet for release

Make yourself a menu link to the /speed-dial page