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Not sure you would see a 404 if Nginx had crashed, do you have anything written to the nginx log? Check the permissions on /var/www/fusionpbx/app/xml_cdr/v_xml_cdr_import.php, in fact check it exists!

What is generating the 488, is it FreeSwitch or an endpoint? Most often a 488 occurs when there is no codec intersection between the endpoints. Check you have t38-passthru set to true and enabled in the relevant SIP profile.

Did you hit enter on the invites to look at the message exchange and see the detail in the packets?

Ok, that makes more sense, so we can see FreeSwitch overriding the SDP sent from the phone by substituting the office WAN IP: b6a7a9a2-3558-444e-bca8-659032801bc5 2020-05-24 07:54:39.832052 [INFO] switch_rtp.c:7682 Auto Changing audio port from 192.168.1.42:4010 to 50.75.94.242:4010 Clearly...

The only thing I would say, I that going "Non-Standard" always seems to make it more difficult to get help. Probably becauase fewer people will understand your environment.

Difficult to comment without fully understanding the network topology of your set up, from the log files it would appear there are two lots of RFC1918 (private addresses) in play, 192.168.1.42 and 172.26.3.197. I can only guess that MY.DOMAIN.NAME.COM resolves to 50.75.94.242, which I guess is...

99% of the time no audio issues are caused by NAT problems. The way to diagnose for sure is to get a packet capture of the SIP messaging and look at the SDP body to see where the RTP (audio streams) are being sent. If you see an RFC1918 address in the SDP body, that will be the cause of the...

Have a look at https://freeswitch.org/confluence/display/FREESWITCH/max_forwards

Works OK for me. (version 4.5.11). Trouble is, installing from Master, two installs two minutes apart could contain different code. Difficult to know where we are at.

I guess what you mean by "expose" is set up some sort of network share for the folder. There are many ways in which it could be done, but I guess if you are in "Windows land" then Samba may be your best option (https://www.samba.org/samba/what_is_samba.html). But believe me, you REALLY DO NOT...

Looking at the code. In themes/default/template.php there is the following line: $menu_brand_text = ($_SESSION['theme']['menu_brand_text']['text'] != '') ? escape($_SESSION['theme']['menu_brand_text']['text']) : "FusionPBX"; Which suggests that the hover text defaults to "FusionPBX" unless a...

If you mean stop the timestamp announcement when the voicemail is played back, then you may find this post helpful: https://www.pbxforums.com/threads/voicemail-disable-message-information.3746/

Your condition statement is testing for the existence of "100" occurring somewhere in the destination number. This is definitely wrong if the inbound route is taking calls from the public switched telephone network. You say your destination number does not contain "100", if this is the case...

Share a link to the work around you found, I'm interested...

Not exactly sure what you have done, a screenshot would help. My guess is that it is now working because the inbound number you are calling has a "100" string in it somewhere. The DATA part is looking for a regular expression so 100 will match any string of digits with a 100 in it. You should...

You should really start a separate thread for this but... Look at the FreeSWITCH event socket: https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket Also look at event hooks, we use an event hook to drive a call centre wallboard, totally separate from FusionPBX...

Your seconds per second still shows max 30. You can look at and change the sessions-per-second values at the fs_cli with: fsctl sps or to set it: fsctl sps 100

ngrep -W byline -td any . host 2.xx.xx.xx and port 5060 or if you want to watch multiple ports ngrep -W byline -td any . host 2.xx.xx.xx and \(port 5060 or port 5080\) You can also use sngrep, which provides a nicer graphical representation in terminal. You mention "puttyed", most people...

When you get the 503, do you see an entry in the FreeSwitch log file? What happens with the same number of concurrent calls but not hitting an IVR? Have a look at your default settings in /etc/freeswitch/autoload_configs/switch.conf you may be hitting the sessions-per-second limit.

You should never need to set registration expiration to such a low value as long as your keep alive pings are working correctly. If NAT holes are getting closed, it is better to properly investigate the cause rather than make your system work around it. Short registration expiry times puts...