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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.

I'm pretty sure the name is the 'profile' name. I've only used it in FreeSwitch myself. So if I created a rule called GB: Translate:GB,XML

Each vendor has their own definitions, so you'll have to set their definitions as per the vendor documentation. I only use Yealink myself.

To use mod-translate to modify inbound calls before they hit the dialplan the following setting for the SIP-profile must be modified: dialplan “XML” -> dialplan “Translate,XML” With FreeSwitch 1.8.x it is now possible to specify the translation profile to be used: dialplan “XML” -> dialplan...

For the most part, set them in Variables and use them in the dialplan like ${variable_name}. You can also set them in channel: <action application="set" data="var_name=var value"/> <action application="log" data="INFO The value in the var_name chan var is ${var_name}"/>...

Removed a long time ago due to security/maintenance issues along with the SQL editor and that XML editor. You have to look at the configs via SSH SSH on to the server, cd /var/www/fusionpbx/resources/templates/provision/yealink We're getting beyond ourselves here... the default FusionPBX...

Are you certain? Because that's all I set, along with the timezone number as 0. What is showing on the Yealink? Has it actually taken the config?

You would hope the SIP Trunk provider is normalising that tbh, anyway... There is a number translation application, based on a FreeSWITCH module which has some examples in: https://docs.fusionpbx.com/en/latest/advanced/number_translations.html

yealink_time_zone_name is not set in your picture. Change it to: United Kingdom(London) and enable it.