Calls dropping randomly

[lmoraes]

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?

[khamlichi.khalil]

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

[lmoraes]

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

[lmoraes]

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

[khamlichi.khalil]

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

[lmoraes]

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

[lmoraes]

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.