Recent content by ja133

For VM transcriptions you can just change the OpenAI Transcription Model model to gpt-transcribe I saw a pull request that starts working on changing the other models that are depreciated but I am sure the developer will chime in on this as well

Dialplan —> global-variables Add a “set” rule: ringback=$${us-ring} Flush cache and give it a shot. Issue resolved

FusionPBX API requires a subscription. FS PBX API is open source

How much utilization do those servers have? We have over 1500 endpoints

Does #1 address the CPU spike when leaving freeswitch running for several weeks without a reboot?

As a safety measure, if an email address is associated with an extension , and then the email is later removed/changed, if there is an associated user login with that email, it suspends the account

Great question. You would have to check with LE directly.

You need to open port 80 to all countries temporarily while the script runs. It looks like letsencrypt cant reach you on port 80

No issues here when I just tested

This is what I have for proxmox, though I have not yet upgraded to 1.11.1, but I have been using Debian 13 since December /etc/pve/qemu-server/<vmid>.conf boot: order=scsi0;net0 cores: 16 cpu: x86-64-v2-AES memory: 131072 meta: creation-qemu=10.0.2,ctime=1759763031 name: pbx02 net0...

Postmarkapp.com

Before the answer command in the dialplan try to set or export (i forgot which one does the trick) absolute_codec_string=PCMU assuming the SIP call to the outside is using G711U

I think 4 GB of ram is the minimum requirement, otherwise it will work fine on AWS. No, it is not ram intensive while running normally; it’s just the initial build process that demands the ram

This issue has existed for years already. I found myself having to make the changes from the database manually. FS PBX, an alternative to what you are using today, solves a lot of these bugs including this one.

If call forward is being set with the system feature code and NOT through the endpoint itself, you can potentially check for the SIP diversion header as a condition