Recent content by markjcrane

You asked for best practices. - FritzBox Gateway should only be a modem with no NAT or it should be bridged and skip NAT - NAT isn't friendly with SIP and RTP so double NAT is not your friend. - Double NAT will make it more difficult to get it working.

Spanish language files need to be installed. Then use sox to convert the files to 8khz and 16khz and put them in the right sounds directories.

The additional languages need to be installed. They aren't installed by default.

The size of zero bytes is telling you that it can't find the voicemail. Try this from the command line as root. After doing this leave a new voicemail and then see if it shows a size. If it shows a size then try downloading the file.

After adding the Provider IP address to the providers access control list. Then go to Status -> SIP Status and press the RELOADACL button.

Your screenshot shows that you haven't assigned any agents to the call center queue. Assign agents to your queue and then press the reload button.

Oh, so you are trying to block out bound calls. If that is the case Delete the old call block dialplan then run upgrade -> app defaults. Confirm there is a new call block dialplan Go to Status -> SIP Status and press the FLUSH CACHE button Then try making a new call.

Screen shot of your call block entry would likely help.

Is this a new install or an existing install that has been upgraded for a few years?

Make sure to leave the caller ID name field empty. Its more effective to block using the Caller ID number.

In default settings under category provision there is a setting enabled that needs to be set to true for provisioning to be enabled. This was put there to only enable provisioning when you have secured it in some way.

We have people using FusionPBX for SIP trunking first person I know of to do this was over 8 years ago. You can use SIP registrations or IP authentication (preferred) using the CIDR field on the extension to authorize the IP address of a remote PBX. My point for adding my comment here is to say...

You can make outbound route that uses FreeSWITCH time conditions. Here is the documentation. - https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Dialplan/Time-of-Day-and-Holiday-Routing_6586868/#docusaurus_skipToContent_fallback

Basic Upgrade Source Code - Required needs to be updated. May require a Git branch change. Upgrade Schema - The required database structure needs to exist. Data types don't change often but recommend updating data types. App Defaults - This doesn't set back to default it looks for missing...

I'll try and be more clear this time. Extension Caller ID input type text If you want extension caller ID name and number to be a input type text then make sure permission outbound_caller_id_select assigned to groups in Groups Manager. By default outbound_caller_id_select is not assigned to...