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The only way to or conditions in xml dialplan is by using the regex=any condition style. <condition regex="any"> <regex field="destination_number" expression="^01324334832$"/> <regex field="destination_number" expression="^01742743501$"/> <action application="transfer" data="3000...

Freeswitch has this feature it is called "call camping" Here's the link https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Modules/mod-dptools/bridge/Call-Camping_16354681/#docusaurus_skipToContent_fallback

No need for dialplan the simplest is to use extension settings.

You can set proxy_media=true in your dialplan when sending faxes so freeswitch wouldn't care about the media type. Also, freeswitch has clear logs on T38 negotiation and media re-negotiation. If you can post them here I can try to help you.

If you want to act when no one answeres the ring group, you must use the timeout destination. Why should the call execute the time condition alternate destination when the ring group times out? Once the time condition was matched, the call was transferred to the ring group.

Can you post the SDP of your initial invite (before the Re-Invite)?

mod_spy is not a maintained module I wouldn't recommend it

It uses lua to get the xml from the file cache or the database and passes it to freeswitch

checkout https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+eavesdrop In fusion it's *33 followed by the extension to tap into

I wrote a simple bash function that will make the output of the freeswitch log files colored like fs_cli does. function fs_log() { if [ $# -eq 0 ] then echo "Error: Missing required argument 'file'" return 1 fi if [ ! -f $1 ] then echo "Error: No...

No, you need to set it on the dial string of the ring group originate. It will require some code modification in the Lua script. I have it working on my servers.

For video to work on ring group enterprise, you'll need to set in the dial string. absolute_codec_string The reason is that the enterprise legs are running in their own thread, so they don't take the codecs from the A-leg.

+OK is the result of Freeswitch API commands. you probably have somewhere in your dialplan where you set the caller id or other variable through an API command and that gets used as the caller id field in the cdr.

Use opensips dispatcher module that has builtin FreeSWITCH support https://opensips.org/html/docs/modules/3.1.x/dispatcher.html#param_fetch_freeswitch_stats

You're right it's failing I just submit a pull to fix it https://github.com/fusionpbx/fusionpbx/pull/5503

Freeswitch documentation gives clear instructions how to do it https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer

I know from this bug since it was committed just comment out lines number 282 and 284 of /usr/share/freeswitch/scripts/app/xml_handler/resources/scripts/configuration/ivr.conf.lua that should solve it

I think for paid members there's something for this

Can you confirm the nginx service is running systemctl status nginx.service

What means the GUI is not loading, is there any error you see in the browser when you try to get to your login is it a browser error or a server error nginx, php?