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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

To add to that, 6 more pages were just converted in the last week. We manage over 3000 endpoints with FS PBX and we will never look back. It's probably the only open source front end that is literally ready to go once you install it and plug in the core information. It is also being developed by...

Freeswitch can crash and yet still respond to the SIP tcp port that you’re monitoring in AWS. How are you handling such instances? Have you found a way for route 53 to specifically check for SIP options?

with an SBC

We service >3000 endpoints in production and moved all those PBXs to FS PBX

nothing wrong with going in that direction either. Maybe just have a primary and failover server. All traffic routes through the primary. Failover is only when there’s a problem or scheduled maintenance and you make dns change so that no one will ever need to know about the second login

How about two central DB servers that are replicated against each other in two separate geographic regions, and then spin up as many PBX servers as you desire, that will either connect to centralized DB server A or centralized DB server B?

That’s exactly what I did. May not be the most perfect solution but it does the job. The alternative that comes to mind is to create a lua script for each outbound call. The lua script runs a Postgres query against your configured DIDs. If there’s a match, then no diversion header is set (and...

What I do is I set a tag on all inbound calls. When placing an outbound call, if the tag condition is met, I automatically add diversion header, as this assumes that the call is being forwarded. Any call without a tag means that the outbound CID was set from the system and can safely be passed...