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Hi Andy, glad I could help. I try to do my bit to keep community support alive for FusionPBX.

The api key can be any string of digits or numbers or whatever, originally it was a uuid but you can just type in to that field any string of random characters. The key can be whatever you want it to be. If you want to use a uuid there is an online uuid generator tool here...

It looks like you have a later version of edit_user.php, more like the version that is currently on the master branch on Github: https://github.com/fusionpbx/fusionpbx/blob/master/core/users/user_edit.php This code seems to create a password using a php function called generate_password at the...

Mmm... Just looking at the PHP code on one of my test servers (FusionPBX V 4.5.11) built on 14th Feb 2020: /var/www/fusionpbx/core/users/user_edit.php At line 988, thi sis the code that creates the api_key edit control and button if (permission_exists('api_key')) { echo "...

That was taken from the "view page source" option in the browser whilst on the edit user page. The script for the uuid() function was very near the bottom of the page source.

I'm sorry I have no idea why this is not working for you but the problem may be browser related. It is pure javascript that populates the API key field when you click the button: <input type='text' class='formfld' style='width: 250px;' name='api_key' id='api_key' value="" ><button...

I'm not aware of any publicly available blacklist. It may be worth looking at the Telephone Preference Service website: https://www.tpsonline.org.uk There is a list of dummy numbers that can be used in TV dramas and films, dodgy callers sometimes present one of these as their CLI...

I agree with @DigitalDaz about the NAT in the DC. I always insist on a raw internet connection in the DC so I know nothing has been manipulated by their routers. To go back to your question, I think I would just restore the backup and then run some SQL to alter the SIP profile settings. Here...

There are many different ways you could do this. (I see someone has also posted while I was typing this). Probably the simplest would be to set up an outbound route with the destination_number condition (Dialplan Expression) set to your short number, for example ^17$. Then wherever $1 is...

@sokalsondha The short answers to your questions are: Unlikely, no idea, and no. RTP (Real Time Protocol) is required for your audio transmission, is as it's name suggests, something that must be processed in real time. You cannot guarantee ANY real time operation on a vcpu, unless you have...

Whatever happened to community spirit? Sadly this is how many good projects shoot themselves in the foot.

Of course you can, but you will need to know where to put the file and how to create the correct record in the database so it will show up as userble in the GUI.

Check to see if the Nginx logs show an error. Depending on the Linux distribution you are using, your php.ini file may be limiting your maximum file upload size and / or your maximum POST size. Check the settings: upload_max_filesize post_max_size

I believe membership is essential if you are using FusionPBX for business with customers that are relying on you. It's not just about training videos, it provides detailed information needed when you perform upgrades etc. If you are in doubt, ask @markjcrane directly or call the FusionPBX...

Freeswitch will allow you to set a diversion header with an invalid URI prior to the bridge statement. A valid format would look something like this: sip_h_Diversion=<sip:somecallerid@somedomain>;reason=unavailable

*33 in the default dialplan executes eavesdrop.lua. You can find this script in /usr/share/freeswitch/scripts. Most of the .lua script is code for handling the pin number, the .lua script eventually calls the mod_dptools eavesdrop application. Details about the application can be found here...

Caller gets put through to voicemail/divert etc. immediately rather than waiting for a 408 Request Timeout?

Have a look at the setting "unregister-on-options-fail" in your internal SIP Profile. https://freeswitch.org/confluence/display/FREESWITCH/mod_sofia

Will this one do for now... https://www.a2es.co.uk/img/pbx-forums-logo.png

Sorry, I have had no feedback whatsoever...