Using jssip.net as a SIP WebRTC client with asterisk 16.10 and PJSIP channel driver we are able to do a SIP Attended Transfer and asterisk generates 3 CDR records for this type of transfer.
Based on Uniqueid and Linkedid, as you can see below, we are not able to link these 3 records as a single call.
| src | dst | clid | channel | dstchannel | call_start | call_answer | call_end | duration | billsec | disposition | uniqueid | linkedid |
|---|---|---|---|---|---|---|---|---|---|---|---|---|
| 1004 | 1009 | 1004 <1004> | PJSIP/1004-00000002 | PJSIP/1009-00000003 | 9/22/2020 14:55 | 9/22/2020 14:56 | 9/22/2020 14:56 | 49 | 39 | ANSWERED | 1600779350.30 | 1600779350.30 |
| 1009 | 6002 | 1009 <1009> | PJSIP/1009-00000004 | PJSIP/6002-00000005 | 9/22/2020 14:56 | 9/22/2020 14:56 | 9/22/2020 14:56 | 31 | 27 | ANSWERED | 1600779368.50 | 1600779368.50 |
| 1004 | 1009 | 1004 <1004> | PJSIP/1004-00000002 | PJSIP/6002-00000005 | 9/22/2020 14:56 | 9/22/2020 14:56 | 9/22/2020 14:57 | 30 | 30 | ANSWERED | 1600779350.30 | 1600779350.30 |
From the second record, judging from the UniqueId and Linkedid you cannot tell that this records belong to the same call.
Is there a way to link these 3 CDR records to represent a single call ?
Maybe having an extra field, something like CallID that it must have the same value for all CDR records that belong to the same call ?
Thanks in advance!