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Some times ims_charging module not sending ccr terminate request to Diameter Server upon receiving the BYE Request .
ISSUE Description:
1 ) Consider User A registered with kamailio.
2 ) A called PSTN number ..Initial CCR request went to diameter server successfully.
3 ) PSTN number hangup the call....Here After BYE Transaction is done...Kamailio is generating a new BYE request to itself and it is retransmitting it four times.
4 ) I think this is the reason Ims_cahrging not generating ccr terminate request.
Please find below sip traces. [Proxy is : 7080 , gateway : 5060]
BYE sip:+xxxxxxxxxxxx@xxxxxxxxxxxx:35465 SIP/2.0
Via: SIP/2.0/UDP xxxxxxxxxxxx:5060;branch=z9hG4bK05c0a4d6
Route: sip:xxxxxxxxxxxx:7080;lr;transport=UDP;did=492.d7f1
Max-Forwards: 70
From: sip:xxxxxxxxxxxx@xxxxxxxxxxxx;tag=as4ab1ab4b
To: "+xxxxxxxxxxxx" sip:+xxxxxxxxxxxx@xxxxxxxxxxxx;tag=8110a0S32NHgp
Call-ID: 8fa995a0-061c-11ea-8d83-00505697042b
CSeq: 102 BYE
User-Agent: SM SoftSwitch 11-06C13
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxxxxxxxxxxx:5060;rport=5060;branch=z9hG4bK05c0a4d6
From: sip:xxxxxxxxxxxx@xxxxxxxxxxxx;tag=as4ab1ab4b
To: "+xxxxxxxxxxxx" sip:+xxxxxxxxxxxx@xxxxxxxxxxxx;tag=8110a0S32NHgp
Call-ID: 8fa995a0-061c-11ea-8d83-00505697042b
CSeq: 102 BYE
User-Agent: Janus WebRTC Gateway SIP Plugin 0.0.6
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, UPDATE
Content-Length: 0
BYE sip:xxxxxxxxxxxx:7080;lr;transport=UDP;did=492.d7f1 SIP/2.0
Via: SIP/2.0/UDP xxxxxxxxxxxx:7080;branch=z9hG4bK461c.5003522650d9dbb33e47c6d83d7465b8.1
Via: SIP/2.0/UDP xxxxxxxxxxxx:5060;rport=5060;branch=z9hG4bK05c0a4d6
Max-Forwards: 69
From: sip:xxxxxxxxxxxx@xxxxxxxxxxxx;tag=as4ab1ab4b
To: "+xxxxxxxxxxxx" sip:+xxxxxxxxxxxx@xxxxxxxxxxxx;tag=8110a0S32NHgp
Call-ID: 8fa995a0-061c-11ea-8d83-00505697042b
CSeq: 102 BYE
User-Agent: SM SoftSwitch 11-06C13
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
BYE sip:xxxxxxxxxxxx:7080;lr;transport=UDP;did=492.d7f1 SIP/2.0
Via: SIP/2.0/UDP xxxxxxxxxxxx:7080;branch=z9hG4bK461c.5003522650d9dbb33e47c6d83d7465b8.1
Via: SIP/2.0/UDP xxxxxxxxxxxx:5060;rport=5060;branch=z9hG4bK05c0a4d6
Max-Forwards: 69
From: sip:xxxxxxxxxxxx@xxxxxxxxxxxx;tag=as4ab1ab4b
To: "+xxxxxxxxxxxx" sip:+xxxxxxxxxxxx@xxxxxxxxxxxx;tag=8110a0S32NHgp
Call-ID: 8fa995a0-061c-11ea-8d83-00505697042b
CSeq: 102 BYE
User-Agent: SM SoftSwitch 11-06C13
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
BYE sip:xxxxxxxxxxxx:7080;lr;transport=UDP;did=492.d7f1 SIP/2.0
Via: SIP/2.0/UDP xxxxxxxxxxxx:7080;branch=z9hG4bK461c.5003522650d9dbb33e47c6d83d7465b8.1
Via: SIP/2.0/UDP xxxxxxxxxxxx:5060;rport=5060;branch=z9hG4bK05c0a4d6
Max-Forwards: 69
From: sip:xxxxxxxxxxxx@xxxxxxxxxxxx;tag=as4ab1ab4b
To: "+xxxxxxxxxxxx" sip:+xxxxxxxxxxxx@xxxxxxxxxxxx;tag=8110a0S32NHgp
Call-ID: 8fa995a0-061c-11ea-8d83-00505697042b
CSeq: 102 BYE
User-Agent: SM SoftSwitch 11-06C13
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
BYE sip:xxxxxxxxxxxx:7080;lr;transport=UDP;did=492.d7f1 SIP/2.0
Via: SIP/2.0/UDP xxxxxxxxxxxx:7080;branch=z9hG4bK461c.5003522650d9dbb33e47c6d83d7465b8.1
Via: SIP/2.0/UDP xxxxxxxxxxxx:5060;rport=5060;branch=z9hG4bK05c0a4d6
Max-Forwards: 69
From: sip:xxxxxxxxxxxx@xxxxxxxxxxxx;tag=as4ab1ab4b
To: "+xxxxxxxxxxxx" sip:+xxxxxxxxxxxx@xxxxxxxxxxxx;tag=8110a0S32NHgp
Call-ID: 8fa995a0-061c-11ea-8d83-00505697042b
CSeq: 102 BYE
User-Agent: SM SoftSwitch 11-06C13
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
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