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The Freeswitch documentation for SIPS/SRTP has a good writeup on creating self signed certs for voice comms with TLS.

Wow. Second option (a little hacky), back in the dialplan add a new row, with: Action - set - sip_from_user=400 Of course mate, not a problem :)

I have not had to enable delete permission as superuser any time I've deleted a domain. YMMV.

Even when your IP is on the line ignoreip=, it will still end up the file /var/log/fail2ban.log. E.g. [nginx-dos] Ignore 127.0.0.1 by ip You will still be able to see what is triggering the ban in the future @sbenwellp. This comes down to basic admin. Manually check those files to spot the...

Title case works better on the internet for titles. In theory, this should work as you intend: 1. copy the default voicemail dialplan entry 2. Edit the copied dialpan, noting the changes highlighted: *edit: "access" was meant to say "ask us".

Delete the domain from Advanced>Domains. Note you can't delete a domain you have selected as your current context.

I know you didn't ask about this, but... Check out Advanced > Variables. I've not had an issue with the busy tone, so I'm not sure where to point you. (Maybe changing the locality settings will change the BUSY tone?) For the second problem, maybe make a simple Call Queue and log your only...

First time I'm hearing of this so I probably misunderstand what is being requested; but isn't this what you're after? https://docs.fusionpbx.com/en/latest/applications/call_broadcast.html

Just make an alternate outbound route where you don't override the CLI.

My guess is your firewall rules. Have a look at your Security Groups applied for this EC2. Make sure you are allowing the whole Freeswitch RTP range. You can find this on the Freeswitch documentation for firewalls. Maybe do a test call allowing 0.0.0.0/0 and all ports first. If that works then...

"gateway" can be many things. In your case, "gateway" will push calls though an extension. I cannot tell you how to do this on the "alcatel-lucent pbx", but I can tell you how to do this on FusionPBX. 1. Create the extension on the destination PBX 2. Go to FusionPBX Accounts > Gateways, and...

Depends on your needs. Obviously Debian 8 + FS 1.6 (or whatver) may have features missing that you may need. I am building my new ones with Debian 9 + 1.8.4 (source) + master as per the current FusionPBX installer.

I reread your question. Your agents dial a prefix to reach a specific outbound route, not they dial a prefix to record. My apologies. You could try this answer? <action application="export" data="execute_on_answer=record_session...

This would record all calls. You would need to check the /var/log/freeswitch/freeswitch.log for your test calls and see if there is an error or an attempt at recording. Probably a file permission error. If you wanted to: Most modern SIP phones can record with SIP signalling. Yealink have a...

In my use case, it was CTO with outdated mentality of "technology is too quick, and no ringing is unrealistic for callers". Oh, I definitely agree with you on that haha.

I believe this is the only 'stock' way of doing this on FusionPBX (Freeswitch). You would need to write something custom to override the existing functionality of turning the button 'green' when the user becomes active again.

I think a ring group with no valid extensions will instantly hit the exit action @Kenny Riley, despite the delay. My lazy solution was to upload a WAV of a ringtone and play that in an IVR before hitting the next destination.

You register a new Gateway with the SIP extension details from the Alcatel, then create a new outbound route pointing to that gateway.

On Yealink using the built in Remote Directory for extensions, my phone does a look up against the extension and shows the name of the person I'm calling. So long as the Extension has the correct Effective Caller ID Name, it will also show the right name. Not sure if that's an option you could...

Have a look at this post I just made @rubberducky, should clear up some things. https://www.pbxforums.com/threads/call-block-or-do-not-call-number.3313/post-10871