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From your last post I am really struggling to understand what you are doing. It may help if you share some screenshots of your complete SIP profile configuration and not just the SIP profile section of the variables page. Do you have a static external IP address? Describe your topology, are...

It looks like your "random" port could be a WAN facing port on the other side of your NAT. I notice you are using STUN for the Ext-RTP-IP, are you doing similar for your Ext-SIP-IP?

There have been some new permissions added. From Group Manager click on the permissions link and scroll down to Time Conditions. You should see a permission called "time_condition_destinations", check this and then log out and back in again.

@andycol Thank you, your input is very much appreciated. Examples of real world deployments are useful to see, but have to be taken with reference to the caveats on actual usage, BLFs, registration expire times etc. mentioned in posts above. Thank you again.

@andycol thank you for sharing. How many extensions are registered? How many SIP profiles are you using? How many CPUs/Cores does your 1 server have?

As I said before, it does look like a permissions issue to me (but I know nothing about Proxmox). I assume it has failed to create the directory /var/run/freeswitch. I believe the FreeSwitch startup process needs to start as root in order to execute things like /bin/mkdir -p...

What do you see if you run "systemctl status freeswitch.service" and "journalctl -xe" ?

It is relatively straight forward, copy you certificate files onto your system by whatever means suits you, then check they have the correct permissions and copy them to the locations required, example script below: #!/bin/bash # make sure we have the correct permissions chown root:ssl-cert...

I don't know the answer. I know @DigitalDaz has some experience with Proxmox, maybe he (or others) will be able to provide a more educated guess than me.

Difficult to call, but this may be a permissions issue. You really need some input from a Proxmox expert.

Last person I remember having that sort of trouble had accidentally installed the 32 bit version of Debian.

My only (tentative) observation would be that as all your phones are on the local LAN, you would want to set both ext-rtp-ip and ext-sip-ip in your internal profile to your local LAN IP. I say this because ext-rtp-ip and ext-sip-ip govern what address is put in the contact headers and in the...

It will be a permissions issue. When you say admin for the domain, I assume you mean a user that is a member of the admin group. Go to Advanced -> Group manager and then click on permissions for the admin group. Scroll down to XML_CDR and choose the permissions you require, you will probably...

If you have a default install, you will have a shell script in /etc/cron.daily/ called fusionpbx-maintenance. Below is an extract of this shell script showing the part that is deleting your recordings after three months: #delete call recordings older than 90 days if [ .$switch_package = .true...

The error is because you have not escaped the + it should be \+ You may find this post enlightening: https://www.pbxforums.com/threads/regex-for-one-single-outbound-route-uk.3819/

Agree with @ad5ou it is more likely to be a routing loop becaise the bridge application only appears on the first line.

You can verify what jails have been set up and what IPs have been put in them with: iptables -nvL It's also worth noting that later versions of fail2ban do not create jails unless it has something to put in them.

You may need to examine more of the FreeSwitch log to understand what is happening. You may also need to get a packet capture of the messages being exchanged between FreeSwitch and your carrier. It could be something as simple as Carrier sends and INVITE to FreeSwitch, the resonse from...

Unless it has been recently updated, I don't believe the GS wave app (/var/www/fusionpbx/app/gswave) uses a provision template, instead it builds (in index.php) an XML document in memory that is sent to the QR code generator. It may be possible to add some more settings to the XML, but I don't...

We include a limited amount of SIP Options monitoring within our Nagios set up. https://exchange.nagios.org/directory/Plugins/Check-Sip-Options/details As for monitoring calls, we actively monitor the raw accounting data (OpenSIPS) for 408 Timeouts and any other response > 499.