asterisk2*CLI> sip show history 
000f8fe9-22b60018-0a82291c-2b7db72a@137.28.94.174  e89dc1b2af604a25b0956e5dbd6f9264                   73b6b3350ea1dc283968224b35d64692@172.28.1.42       
asterisk2*CLI> sip show history 000f8fe9-22b60018-0a82291c-2b7db72a@137.28.94.174
asterisk2*CLI> 
  * SIP CallI> 
1. Rx              INVITE / 101 INVITE / sip:999999@172.28.1.42
2. AuthChal        Auth challenge sent for  - nc 0
3. TxRespRel       SIP/2.0 / 101 INVITE - 401 Unauthorized
4. SchedDestroy    32000 ms
5. Rx              ACK / 101 ACK / sip:999999@172.28.1.42
6. Rx              INVITE / 102 INVITE / sip:999999@172.28.1.42
7. CancelDestroy   
8. Invite          New call: 000f8fe9-22b60018-0a82291c-2b7db72a@137.28.94.174
9. AuthOK          Auth challenge succesful for 364774
10. NewChan         Channel SIP/364774-09aa5d58 - from 000f8fe9-22b60018-0a82291c-2
11. TxResp          SIP/2.0 / 102 INVITE - 100 Trying
12. TxRespRel       SIP/2.0 / 102 INVITE - 200 OK
13. Rx              ACK / 102 ACK / sip:999999@172.28.1.42
14. Rx              BYE / 103 BYE / sip:999999@172.28.1.42
15. RTCPaudio       Quality:ssrc=1494421050;themssrc=3055741327;rxjitter=0.000000;r
16. RTCPaudioJitter Quality:rxjitter=0.000000;
17. RTCPaudioLoss   Quality:lost=0;expected=1;
18. RTCPaudioRTT    Quality:Not available
19. TxResp          SIP/2.0 / 103 BYE - 200 OK
asterisk2*CLI> sip show history e89dc1b2af604a25b0956e5dbd6f9264
No such SIP Call ID starting with 'e89dc1b2af604a25b0956e5dbd6f9264'
asterisk2*CLI> sip show history 73b6b3350ea1dc283968224b35d64692@172.28.1.42
asterisk2*CLI> 
  * SIP CallI> 
1. NewChan         Channel SIP/exchange2-09aace10 - from 73b6b3350ea1dc283968224b3
2. TxReqRel        INVITE / 102 INVITE - INVITE