That makes a ton more sense now. I assumed that they were one in the same, but it's good to know that they are perceived differently.

Thanks for your guidance!

Additionally, we do have a license in production for FOP2. This is just some testing we are doing.

Ah, I saw the "Hold Reporting" plugin, and this text is confusing me:

"This plugin will catch hold/unhold events in AMI and write a proper entry in the queue_log file for that event, so reporting tools like Asternic Call Center Stats can produce the relevant reports."

Is this reporting different than what I am thinking of?

We've installed the demo version of FOP2 onto our test system here. We're trying to get the MOH tracked through QueueMetrics, which requires that the queue_log have records in it for MOH.

We are running FreePBX 2.11.0.14 with Asterisk 11, and QueueMetrics 14.03.

For the life of us, we can't get the MOH to write to the Queue Log. We ran the debug with FOP2, and we see that MOH is being seen by FOP2:

127.0.0.1 <- Event: MusicOnHold^M
127.0.0.1 <- Privilege: call,all^M
127.0.0.1 <- State: Start^M
127.0.0.1 <- Channel: SIP/86000-00000002^M
127.0.0.1 <- UniqueID: 1410802601.30^M
127.0.0.1 <- Class: default^M

-and-

127.0.0.1 <- Event: MusicOnHold^M
127.0.0.1 <- Privilege: call,all^M
127.0.0.1 <- State: Stop^M
127.0.0.1 <- Channel: SIP/86000-00000002^M
127.0.0.1 <- UniqueID: 1410802601.30^M

It's being seen in the final part of the call as well:

127.0.0.1 <- Event: AgentConnect^M
127.0.0.1 <- Privilege: agent,all^M
127.0.0.1 <- Queue: 8000^M
127.0.0.1 <- Uniqueid: 1410802601.30^M
127.0.0.1 <- Channel: SIP/1584-00000003^M
127.0.0.1 <- Member: SIP/1584^M
127.0.0.1 <- MemberName: SIP/1584^M
127.0.0.1 <- HoldTime: 1^M
127.0.0.1 <- BridgedChannel: 1410802607.31^M
127.0.0.1 <- RingTime: 1^M

Attached is the capture.log file that we ran. Is this issue something that Demo mode has limitations on? or is there something we need to look at specifically?

We've confirmed that all permissions for the AMI user are correct, that FOP2 conf has been added to /etc/asterisk/extensions_override_freepbx.conf, and that everything seems to be included.

We're just scratching our head on this.