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    Who Ends the Call First ?!

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    • AlyRagabA
      AlyRagab
      last edited by AlyRagab

      Dears,
      In Asterisk , is there any availability to know who ends the call first ?
      it will be useful for any call center environment so i am searching for an answer for this question 🙂

      Thanks a lot 🙂

      1 Reply Last reply Reply Quote 2
      • scottalanmillerS
        scottalanmiller
        last edited by

        It's not generally shown anywhere other than the logs. But the end signal should be visible in the logs.

        AlyRagabA 1 Reply Last reply Reply Quote 0
        • AlyRagabA
          AlyRagab @scottalanmiller
          last edited by

          @scottalanmiller which logs can view that in asterisk ?

          1 Reply Last reply Reply Quote 0
          • scottalanmillerS
            scottalanmiller
            last edited by

            /var/log/asterisk/full

            1 Reply Last reply Reply Quote 0
            • scottalanmillerS
              scottalanmiller
              last edited by

              Look for lines like this:

              [2016-12-11 17:29:58] VERBOSE[5744][C-000031cf] pbx.c:     -- Executing [s@from-trunk:7] Hangup("SIP/162.209.2.96-000014e6", "") in new stack
              AlyRagabA 2 Replies Last reply Reply Quote 2
              • AlyRagabA
                AlyRagab @scottalanmiller
                last edited by

                @scottalanmiller Greet , so if the Agent who ended the call it will show that the SIP/IP who hanged up the call
                thanks Scott for this point 🙂

                scottalanmillerS 1 Reply Last reply Reply Quote 1
                • scottalanmillerS
                  scottalanmiller @AlyRagab
                  last edited by

                  @AlyRagab said in Who Ends the Call First ?!:

                  @scottalanmiller Greet , so if the Agent who ended the call it will show that the SIP/IP who hanged up the call
                  thanks Scott for this point 🙂

                  De nada

                  1 Reply Last reply Reply Quote 3
                  • AlyRagabA
                    AlyRagab @scottalanmiller
                    last edited by

                    @scottalanmiller said in Who Ends the Call First ?!:

                    Look for lines like this:

                    [2016-12-11 17:29:58] VERBOSE[5744][C-000031cf] pbx.c:     -- Executing [s@from-trunk:7] Hangup("SIP/162.209.2.96-000014e6", "") in new stack

                    but of course we will need to access this log file in a GUI to be easy to check this issue and search by date and time and by extension.

                    scottalanmillerS 1 Reply Last reply Reply Quote 0
                    • DashrenderD
                      Dashrender
                      last edited by

                      it's a log, open it in notepad/wordpad/excel, etc

                      It's just raw text, nothing special.

                      If you want a fancy GUI around it, then use something like LogStash to send the logs to, then use it's GUI.

                      1 Reply Last reply Reply Quote 4
                      • scottalanmillerS
                        scottalanmiller @AlyRagab
                        last edited by

                        @AlyRagab said in Who Ends the Call First ?!:

                        but of course we will need to access this log file in a GUI to be easy to check this issue and search by date and time and by extension.

                        Of course. Good logging practice is always to send your logs to a logging system like Graylog or ELK.

                        But looking at it via the command line is actually far easier than any local GUI. GUIs just slow down looking at text files.

                        1 Reply Last reply Reply Quote 1
                        • JaredBuschJ
                          JaredBusch
                          last edited by

                          That hangup line is not specific. all calls run this macro i believe. i would need to dig deeper in the logs to verify that though.

                          1 Reply Last reply Reply Quote 2
                          • JaredBuschJ
                            JaredBusch
                            last edited by

                            So yeah, there is not way to know this detail just from watching the command line (asterisk -rvvvvv) in Elastix 2.4

                            Call terminated by the person calling in:

                              == Using SIP RTP CoS mark 5
    -- Called SIP/5199
       > 0xa868640 -- Probation passed - setting RTP source address to 208.100.39.53:15750
    -- SIP/5199-000057f0 is ringing
       > 0xa868640 -- Probation passed - setting RTP source address to 208.100.39.53:15750
    -- SIP/5199-000057f0 answered SIP/voipms-000057ef
       > 0xa868640 -- Probation passed - setting RTP source address to 208.100.39.53:15750
       > 0xaecf2e8 -- Probation passed - setting RTP source address to 10.254.103.10:12560
    -- Executing [h@macro-dial-one:1] Macro("SIP/voipms-000057ef", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/voipms-000057ef", "1?endmixmoncheck") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/voipms-000057ef", "End of MIXMON check") in new stack
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/voipms-000057ef", "1?nomeetmemon") in new stack
    -- Goto (macro-hangupcall,s,28)
    -- Executing [s@macro-hangupcall:28] NoOp("SIP/voipms-000057ef", "End of MEETME check") in new stack
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/voipms-000057ef", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,34)
    -- Executing [s@macro-hangupcall:34] NoOp("SIP/voipms-000057ef", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:35] GotoIf("SIP/voipms-000057ef", "1?noautomon2") in new stack
    -- Goto (macro-hangupcall,s,41)
    -- Executing [s@macro-hangupcall:41] NoOp("SIP/voipms-000057ef", "MONITOR_FILENAME=") in new stack
    -- Executing [s@macro-hangupcall:42] GotoIf("SIP/voipms-000057ef", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,45)
    -- Executing [s@macro-hangupcall:45] GotoIf("SIP/voipms-000057ef", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,48)
    -- Executing [s@macro-hangupcall:48] GotoIf("SIP/voipms-000057ef", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,50)
    -- Executing [s@macro-hangupcall:50] AGI("SIP/voipms-000057ef", "hangup.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
    -- <SIP/voipms-000057ef>AGI Script hangup.agi completed, returning 0
    -- Executing [s@macro-hangupcall:51] Hangup("SIP/voipms-000057ef", "") in new stack
  == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/voipms-000057ef' in macro 'hangupcall'
  == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/voipms-000057ef'
  == Spawn extension (macro-dial-one, s, 37) exited non-zero on 'SIP/voipms-000057ef' in macro 'dial-one'
  == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/voipms-000057ef' in macro 'exten-vm'
  == Spawn extension (from-did-direct, 5199, 1) exited non-zero on 'SIP/voipms-000057ef'
localhost*CLI>

                            Call terminated by the answering agent:

                              == Using SIP RTP CoS mark 5
    -- Called SIP/5199
       > 0xa868640 -- Probation passed - setting RTP source address to 208.100.39.53:15338
    -- SIP/5199-000057f2 is ringing
       > 0xa868640 -- Probation passed - setting RTP source address to 208.100.39.53:15338
    -- SIP/5199-000057f2 answered SIP/voipms-000057f1
       > 0xa868640 -- Probation passed - setting RTP source address to 208.100.39.53:15338
       > 0xaecf2e8 -- Probation passed - setting RTP source address to 10.254.103.10:12564
    -- Executing [h@macro-dial-one:1] Macro("SIP/voipms-000057f1", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/voipms-000057f1", "1?endmixmoncheck") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/voipms-000057f1", "End of MIXMON check") in new stack
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/voipms-000057f1", "1?nomeetmemon") in new stack
    -- Goto (macro-hangupcall,s,28)
    -- Executing [s@macro-hangupcall:28] NoOp("SIP/voipms-000057f1", "End of MEETME check") in new stack
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/voipms-000057f1", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,34)
    -- Executing [s@macro-hangupcall:34] NoOp("SIP/voipms-000057f1", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:35] GotoIf("SIP/voipms-000057f1", "1?noautomon2") in new stack
    -- Goto (macro-hangupcall,s,41)
    -- Executing [s@macro-hangupcall:41] NoOp("SIP/voipms-000057f1", "MONITOR_FILENAME=") in new stack
    -- Executing [s@macro-hangupcall:42] GotoIf("SIP/voipms-000057f1", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,45)
    -- Executing [s@macro-hangupcall:45] GotoIf("SIP/voipms-000057f1", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,48)
    -- Executing [s@macro-hangupcall:48] GotoIf("SIP/voipms-000057f1", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,50)
    -- Executing [s@macro-hangupcall:50] AGI("SIP/voipms-000057f1", "hangup.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
    -- <SIP/voipms-000057f1>AGI Script hangup.agi completed, returning 0
    -- Executing [s@macro-hangupcall:51] Hangup("SIP/voipms-000057f1", "") in new stack
  == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/voipms-000057f1' in macro 'hangupcall'
  == Spawn extension (macro-dial-one, h, 1) exited non-zero on 'SIP/voipms-000057f1'
  == Spawn extension (macro-dial-one, s, 37) exited non-zero on 'SIP/voipms-000057f1' in macro 'dial-one'
  == Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'SIP/voipms-000057f1' in macro 'exten-vm'
  == Spawn extension (from-did-direct, 5199, 1) exited non-zero on 'SIP/voipms-000057f1'
localhost*CLI>
                            1 Reply Last reply Reply Quote 2
                            • scottalanmillerS
                              scottalanmiller
                              last edited by

                              Crap, that sucks.

                              1 Reply Last reply Reply Quote 0
                              • DashrenderD
                                Dashrender
                                last edited by Dashrender

                                Unless you capture the disconnect button press on a call, I don't know how that would be tracked. But I don't know much about telephony either.

                                scottalanmillerS 1 Reply Last reply Reply Quote 0
                                • scottalanmillerS
                                  scottalanmiller @Dashrender
                                  last edited by

                                  @Dashrender said in Who Ends the Call First ?!:

                                  Unless you capture the disconnect button press on a call, I don't know how that would be tracked. But I don't know much about telephony either.

                                  It's not telephony but the specifics of Asterisk that are in question.

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