Calls dropping randomly

I have a scenario that receive 500 calls (incomming) daily from different DID numbers in a Queue with 6 endpoints registered in an Asterisk PBX server operating in a cloud server AWS over the Internet with PJSIP and Realtime with MySQL DB.

Endpoints are registered in a private network behind NAT.

Sometimes the administrator reports to me that some days a couple of calls drop randomly, as example 3 calls per hour.

I would like to know how can I detect if the calls was dropped by the originator of the call or by the endpoints registered in my PBX?

The queue app does log who hangup the call agent or customer.

how can I get the queues log with this level of informations?

How can I discover how hangup the call first by the SIP packets with the sip dump? If yes how?

asterisk -rx ‘pjsip logger on’
asterisk -r |tee >dump.log

Then you need to ask them to record the numbers that disconnected and the time

Thanks, and how can I detect how dropped the call when it happen by the SIP log?

Exemplo de DUMP:

0K<— Received SIP request (362 bytes) from UDP:201.Z.Y.X:5060 —>
BYE sip:54.Z.Y.54:5060 SIP/2.0
Via: SIP/2.0/UDP 201.Z.Y.X:5060;branch=z9hG4bK00B5db9051cb9f46346
From: sip:16988461239@201.Z.Y.X;tag=gK0034b6e5
To: sip:551639747828@54.Z.Y.54;tag=093f4f68-d655-40bf-91ec-58857604d520
Call-ID: 151031461_100655322@201.Z.Y.X
CSeq: 615759 BYE
Max-Forwards: 70
Reason: Q.850;cause=16
Content-Length: 0

0K<— Transmitting SIP response (364 bytes) to UDP:201.Z.Y.X:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 201.Z.Y.X:5060;rport=5060;received=201.Z.Y.X;branch=z9hG4bK00B5db9051cb9f46346
Call-ID: 151031461_100655322@201.Z.Y.X
From: sip:16988461239@201.Z.Y.X;tag=gK0034b6e5
To: sip:551639747828@54.Z.Y.54;tag=093f4f68-d655-40bf-91ec-58857604d520
CSeq: 615759 BYE
Server: Asterisk PBX

The hangup request is from my SIP provider. I am requesting analyze from them.

thank you.