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The author of this thread and the error messages: 2023-04-08 19:41:31.011432 97.20% [DEBUG] sofia_glue.c:3422 IP 111.111.111.83 is on local network, not seting NAT mode. 2023-04-08 19:41:31.311356 97.20% [DEBUG] sofia_glue.c:3422 IP 111.111.111.84 is on local network, not seting NAT mode.

No, that would require some heavy modification to call_forward.lua to include ring_groups. What you need to do is use a Call Flow to allow users to toggle a call flow at the desired times.

rfc1918.auto - RFC1918 Space nat.auto - RFC1918 Excluding your local lan. localnet.auto - ACL for your local lan. loopback.auto - ACL for your local lan. Your external profile probably has localnet.auto in for the local-network-acl, and from your subnetting design that is...

Yeh *720800111111, or just enable (*72), disable(*73) or toggle(*74). You can modify the request_id=false line to set it to true, so anyone as long as you have the voicemail pin can enable a call forward.

Think you might also want: From User: <Your Username> From Domain: xxxxxx.phonestar.ch I think that's why it's defaulting to 'FreeSwitch'

You're usually looking for 'VAD' - Voice Activity Detection. Personally, I turn it off where possible. It usually results in clipping of voices at the start and end of sentences. You can force things with Comfort Noise Generation sometimes...

Autofill how? One up from the last one you defined?

First things first, remove the domain name from the ACL. You rarely ever want that there. Only the CIDR IP Addresses of the providers SBCs for IP Authentication. I don't think you want username authentication, that's what it looks like your trunk is doing with the username of 'FreeSWITCH' and...

Usually now is about time to look at: 1. The logs in /var/log/freeswitch/freeswitch.log, can you see the INVITE attempts? 2. Running sngrep from a root terminal, can you see the SIP invites, is FusionPBX responding? If you can't see either, then you know it's a network issue still. Unlikely...

Only in the interests of keeping this forum the premier location for FusionPBX knowledge! First few nights the support desk were not logging out of their phones like instructed so it got very confusing seeing yesterdays calls on the wallboard at 8am!

Think you'll want this on your outbound route: https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Dialplan/Variables-Archive/x__DTMF-Variables_16352562/#dtmf_type <action application="set" data="dtmf_type=rfc2833"/> The FreeSwitch documentation is a bit shitty for DTMF, so I'm...

Add the user to the ring group. As for forwarding via star code *72,*73,*74 are call forward enable, call forward disable and call forward toggle respectively.

Is that user assigned to any extensions?

Oh shit, yeh. Very important 'codec'. Weird they let you disable it without a fat warning.

Does anything show in the logs? The only thing I can suggest is trying <param name="pass-rfc2833" value="true"/> on the external profile, this disabled the de-encode re-encode of DTMF in the event the two legs have different methods.

Worked it out, logging in and logging out is the key.

There's no 2FA, but here are the password reset settings. You have to ensure SMTP is working before it will work.

Are you updating the menu_language table too? The only reason I say this, is I discovered that when I made a custom menu and users were set to 'British English' then the menu bar just would display blank.

That's not the provisioning settings for the phone. That's for the web interface. All of the settings you need are in the Provision section.

Makes sense, I was looking through the source code and that definitely does work so i was confused. I always forget about that cache.