Recent content by pbxgeek

AI is expensive, but clients want it. Since this feature is so popular and clients want to pay for it, we decided to introduce it in FS PBX to make things easier. When I have time, I'll post a full video tutorial on YouTube. One of my personal clients said he would rather pay $500 for an AI...

Here is a possible use case and how it works. You have 50 DIDs for a client. All of them need to check business hours first and then route to their own destination. Old configuration (without dynamic routes module): You will have to create 50 Business hours sets and adjust the individual route...

The new FS PBX Contact Center update brings two major additions: callers can request a callback without losing their queue priority, and queue configuration and agent availability can now stay synchronized across redundant servers. These are capabilities people look for when comparing...

Connection timed out after 5002 milliseconds Yesterday, someone blocked their own IP in the firewall, and they got that exact error message. We unblocked it, and the ACME challenge succeeded. Other possibilities: 1) Port 80 is blocked 2) Port 80 is geofenced and not open to the whole world...

Ah, you got me caught red-handed. The features are there, but I haven't documented them yet. It's been busy here at FS PBX. I'll get around it soon.

@Starburst, GitHub issues are the best place to report small bugs like this. This way, they get proper attention and don't get lost. Can you share an example of the number that isn't properly formatted?

@Starburst NAT is a nasty thing that can affect different devices differently. Without a full SIP capture, we can't tell what's going on. You might not have RPORT enabled on the device, which could cause the same problem, and Cisco may have it on by default. There are so many possibilities for...

I'm glad it got resolved. Make sure to follow the official update guide in the future. This step is documented and needed for most updates.

You cut out the actual INVITE messages, so it's hard to tell what the phone is trying to dial. Digimap would only affect something if you are adding prefixes to dialed numbers. But you should still get replies even if the number is dialed incorrectly. Instead, you are sending a bunch of...

So the dead air is because the phone is actually not dialing anything? Does it dial when you press the Dial button? Those would be very different issues.

Yes, this old firewall that was inherited from FusionPBX needs to be majorly redesigned. It has many nuances that were missed. We are planning to improve all of them very soon. You can whitelist the IPs in the ACL list.

@willy Lafort, from the log file, I can see that you might be missing some migrations. And that's why you sometimes get a 500 error. run this to find out if you do: cd /var/www/fspbx php artisan migrate:status

Those failures are not the source of the issue. This is how the dialplan works. Each individual dialplan has conditions. If the condition is true, it executes. If not, you see the word FAIL, and it moves on to the next dialplan. It's desinged that way because not every dialplan needs to be...

Thanks for reporting this. Sofia-SIP compiled successfully; the installer failed because ldconfig was outside the shell’s PATH on your server. We’ve fixed the installer to use the absolute path and committed it to git. Please try again

Gateway Settings -> Advanced - Caller ID in From. It's set to true by default unless you are in a very old version of FS PBX