FOP2 Community Forum

Of course not everyone uses FreePBX. If this setting is not one that can have a good default based on deployed system, then maybe make a help note someplace so admins can know where to set this.

It seems very random m, so I suspect not. I will verify though.

I can and will change this, thanks. But I think this also needs to be default if you want things to work with FreePBX still.

When I record a call within FOP2, it does correctly record the call and write the filename to the CDR record. Unfortunately, FreePBX 15 requires that the seperator be a - and not a _ now. Additionally, the call needs a field added prior the to the name ( i used fop2 in my manual test) and it needs the date. Once fixed, like this, the recording shows up in the CDR search as well as the UCP.
Original filename: 219_219_111806_1638551871.156513_GENERAL.wav
Working filename: fop2-219-219-20211203-111806-1638551871.156513-GENERAL.wav
The issue is how the CDR (and UCP as it worked but I haven't checked the code) finds the file.
https://github.com/FreePBX/cdr/blob/2969c0781e0322af94048380b6f3adde18f23621/Cdr.class.php#L489

When I click on call recording or call history, i only get a white screen.

Using FreePBX 15, Asterisk 18.2 and FOP2 2.31.30

Absolutely. This product is now a zombie. I wish it was opensource licensed so that someone else could pick it up.

I asked about this a year ago. Forgot about it actually until I saw your post. Dev did not reply until December
It is an advanced setting apparently..
http://forum.fop2.com/discussion/4823/dial-from-fop2-doe-snot-support-calling-to-multiple-pjsip-aors/

I have3 or 4 extension on this system that never show the call info. Only the users turning orange form green to indicate they are on a call.
System is FreePBX 15 (distro), with Asterisk 18.1.1 but this issue has been going on for more than a year.

It seems something about the subsequent if logic to after the amportal check is not working. Going only to the /etc/freepbx.conf does work as I simply commented out the block and added the if because it does not apply to the install. but obviously, you need to actually fix the logic so it works for various systems.
This is the error I was getting

FreePBX moved the DB credentials out of etc/amportal.conf quite some time ago now.
But they left the file in place with other information still. Because of that, the logic in config.php is no longer valid.

    if(!$db->is_connected()) {
        // Database connection details from amportal
=begin comment
This logic is invalid on FreePBX 15 as /etc/amportal.conf still exists.

        if (is_readable("/etc/amportal.conf")) {
            $amp_conf = parse_amportal_conf("/etc/amportal.conf");
            $DBHOST = $amp_conf['AMPDBHOST'];
            $DBNAME = $amp_conf['AMPDBNAME'];
            $DBUSER = $amp_conf['AMPDBUSER'];
            $DBPASS = $amp_conf['AMPDBPASS'];

            if(!isset($amp_conf['AMPDBENGINE'])) {
                if(is_file("/etc/freepbx/freepbx.conf")) {
                    $amp_conf = parse_amportal_conf("/etc/freepbx/freepbx.conf");
                } else if(is_file("/etc/freepbx.conf")) {
                    $amp_conf = parse_amportal_conf("/etc/freepbx.conf");
                }
                $DBHOST = $amp_conf['AMPDBHOST'];
                $DBNAME = $amp_conf['AMPDBNAME'];
                $DBUSER = $amp_conf['AMPDBUSER'];
                $DBPASS = $amp_conf['AMPDBPASS'];
            }

	} else if (is_readable("/etc/freepbx/freepbx.conf")) {
            $amp_conf = parse_amportal_conf("/etc/freepbx/freepbx.conf");
            $DBHOST = $amp_conf['AMPDBHOST'];
            $DBNAME = $amp_conf['AMPDBNAME'];
            $DBUSER = $amp_conf['AMPDBUSER'];
            $DBPASS = $amp_conf['AMPDBPASS'];
        } else if (is_readable("/etc/freepbx.conf")) {
=end comment
=cut
    	if (is_readable("/etc/freepbx.conf")) {
            $amp_conf = parse_amportal_conf("/etc/freepbx.conf");
            $DBHOST = $amp_conf['AMPDBHOST'];
            $DBNAME = $amp_conf['AMPDBNAME'];
            $DBUSER = $amp_conf['AMPDBUSER'];
            $DBPASS = $amp_conf['AMPDBPASS'];
        } else if (is_readable("/etc/asterisk/snep/snep-features.conf")) {
            $DBHOST = 'localhost';
            $DBNAME = 'snep';
            $DBUSER = 'snep';
            $DBPASS = 'sneppass';
            $CDRDBTABLE = "snep.cdr";
        }
	$db = new dbcon($DBHOST, $DBUSER, $DBPASS, $DBNAME, true, true, $FORCE_UTF8);

Lots of issues on FreePBX 15 is seems.
I can tell a call to record and it goes to the database.

But the recordings is empty.

ran the upgrade command on the old server, and I was able to revoke.
wget -O - http://download.fop2.com/upgrade_fop2.sh | bash
But on the new server, it would still not register from the command line even after running the upgrade command.
Tried to register from the GUI and it worked.

@Kirk_S said:
Great feedback JaredBusch! The exact scenario "Having an operator change it in FOP2 and then have the phone reflect the same status." works for us on hundreds of systems with the setup I described. Good luck with your "cludgy" code.

Changing the DND status in FOP2 does not cause the phone to to come off of DND. It does cause the PBX to not see the extension in DND.

This post is now two week old and I am still unable to activate or revoke service.

I migrated servers and need to move the license.

@Kirk_S said:
If you change your Yealink to use the system feature codes then the handset and FOP2 DND status will be synced and there is no need to virtually press the DND button remotely on the device.

That is standard, and does nothing for the point of the post. Having an operator change it in FOP2 and then have the phone reflect the same status.

@Posi said:
How can I stop people from accessing the admin page?
Can you add a user and password?

If you are using FreePBX, the access to /admin is tied to the admin login of FreePBX. I never tested if being logged in to the FreePBX UCP also lets you access /admin though.

I do not have any issue with this using 2.31.28

If you use PJSIP for extensions and have more than one extension registered, only the first AOR listed in pjsip show aor XXX will be dialed.
FOP2 is likely using DIAL and it it needs updated. An example can be found in this bug report:
https://issues.freepbx.org/browse/FREEPBX-20576

Yeah, what I posted works great. But every time I update FOP2 I'm worried I will lose my changes to /var/www/html/fop2/admin/plugins/presencedndfpbx/presencedndfpbx.pl. That is why I was wondering if there was a way to make this into a plugin.
I get the issue with creating contexts. Why not have the plugin create its own conf file in /etc/asterisk and then have the plugin modify /etc/asterisk/extensions_override_freepbx.conf to include it.

This is definitely not something everyone will use. But I have found it quite helpful for places where users only have their dsk phone and the operator has the FOP2 panel.