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You are partially correct. If you use SRTP, you also need to register the phone using TLS. That will dictate which port is used. I would check FS CLI too. It sounds like you may be able to see some info there related to this. I can assure you it works on Yealinks and TLS with SRTP as I have it...

Is your call park working on another phone connected to the same domain?

This works for all my yealinks. See the screenshot

yes that's the one.

I want to highlight an important aspect of any PBX: we optimize server resource usage. Every stage of call transcription is managed through our scalable queuing system, which leverages Laravel queues, Redis, and Horizon. This setup ensures there are no noticeable spikes in CPU or memory...

Hey everyone, We’ve just released a new Call Transcriptions module in FS PBX, bringing AI-powered transcription and summarization directly into your PBX — no external scripts, no manual uploads, no patchwork integrations. Here’s what this release includes AI Voice Transcriptions (AssemblyAI)...

I'm in the same boat as you, and I've concluded that Freeswitch documentation is often outdated or non-existent, so no LLM models can be appropriately trained to produce good output. It's a hit or miss, and there is a lot of hallucination happening.

We have committed a fix for this and assisted you on GitHub. Adding a note here in case someone else is having the same issue.

There is a setting to keep the voicemail from answering on the other side in ring groups and follow me. I don't remember what it's called in FusionPBX. In FS PBX, we renamed it to make its purpose easier to understand. In our system, it's called Enable Answer Confirmation

We have SMS notification for voicemails in FS PBX (once a fork of FusionPBX that was heavily modified). It wouldn't be too hard to add transcripts to them. Reach out in DM if you are interested in this feature. We can work on implementing it.

show us your domain settings and extension settings for 1021 under that domain

Log out and log back in. Since the domain is create using the old page the new pages don’t know about it until you log in again. This will be addressed when we update the domain page next month.

This is not the only way, it's just one of the ways, and it works well when you have TTL set to a very low number, like 60 seconds. Then phones reregsitrer to the backup server fast. How do you do it? @s2svoip

That is correct. When we are done, there will be no FusionPBX left in the system.

Check the sip_from_user variable and make sure it doesn't match one of your extensions via regex before testing the "^(\d{3,20})$" condition. This should work to bypass your outbound rule. https://developer.signalwire.com/freeswitch/Channel-Variables-Catalog/sip_from_user_16354276/

@rcksl we are working on redesigning each page one by one. When we release them it’s not just a face lift but we also make sure all back end code is optimized. So it takes time. We started with all customer facing pages so they can do self management if needed. Then we moved on to the admin...

We’ve added documentation and new scripts for High Availability (HA) deployments in FS PBX With two FS PBX nodes, you can now run your platform in Primary–Standby mode — both servers stay synchronized, ready for instant cutover if one fails. Overview The HA setup uses bi-directional...

It'd be very hard for you to configure them to talk to each other if they are all connected to different servers. It's not entirely impossible through a clever use of bridge statements, but it's not the best practice.

You have a problem with one of the FreeSwitch files, hence all the errors. The phone won't register until you resolve your FreeSWITCH errors.

Why do you need three servers? Wouldn't it be easier to combine all into one server? Then they can talk to each other