Who Ends the Call First ?!



  • 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 🙂



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



  • @scottalanmiller which logs can view that in asterisk ?



  • /var/log/asterisk/full



  • Look for lines like this:

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


  • @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 🙂



  • @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



  • @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 [[email protected]: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.



  • 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.



  • @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.



  • 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.



  • 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 [[email protected]:1] Macro("SIP/voipms-000057ef", "hangupcall,") in new stack
        -- Executing [[email protected]:1] GotoIf("SIP/voipms-000057ef", "1?endmixmoncheck") in new stack
        -- Goto (macro-hangupcall,s,9)
        -- Executing [[email protected]:9] NoOp("SIP/voipms-000057ef", "End of MIXMON check") in new stack
        -- Executing [[email protected]:10] GotoIf("SIP/voipms-000057ef", "1?nomeetmemon") in new stack
        -- Goto (macro-hangupcall,s,28)
        -- Executing [[email protected]:28] NoOp("SIP/voipms-000057ef", "End of MEETME check") in new stack
        -- Executing [[email protected]:29] GotoIf("SIP/voipms-000057ef", "1?noautomon") in new stack
        -- Goto (macro-hangupcall,s,34)
        -- Executing [[email protected]:34] NoOp("SIP/voipms-000057ef", "TOUCH_MONITOR_OUTPUT=") in new stack
        -- Executing [[email protected]:35] GotoIf("SIP/voipms-000057ef", "1?noautomon2") in new stack
        -- Goto (macro-hangupcall,s,41)
        -- Executing [[email protected]:41] NoOp("SIP/voipms-000057ef", "MONITOR_FILENAME=") in new stack
        -- Executing [[email protected]:42] GotoIf("SIP/voipms-000057ef", "1?skiprg") in new stack
        -- Goto (macro-hangupcall,s,45)
        -- Executing [[email protected]:45] GotoIf("SIP/voipms-000057ef", "1?skipblkvm") in new stack
        -- Goto (macro-hangupcall,s,48)
        -- Executing [[email protected]:48] GotoIf("SIP/voipms-000057ef", "1?theend") in new stack
        -- Goto (macro-hangupcall,s,50)
        -- Executing [[email protected]: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 [[email protected]: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 [[email protected]:1] Macro("SIP/voipms-000057f1", "hangupcall,") in new stack
        -- Executing [[email protected]:1] GotoIf("SIP/voipms-000057f1", "1?endmixmoncheck") in new stack
        -- Goto (macro-hangupcall,s,9)
        -- Executing [[email protected]:9] NoOp("SIP/voipms-000057f1", "End of MIXMON check") in new stack
        -- Executing [[email protected]:10] GotoIf("SIP/voipms-000057f1", "1?nomeetmemon") in new stack
        -- Goto (macro-hangupcall,s,28)
        -- Executing [[email protected]:28] NoOp("SIP/voipms-000057f1", "End of MEETME check") in new stack
        -- Executing [[email protected]:29] GotoIf("SIP/voipms-000057f1", "1?noautomon") in new stack
        -- Goto (macro-hangupcall,s,34)
        -- Executing [[email protected]:34] NoOp("SIP/voipms-000057f1", "TOUCH_MONITOR_OUTPUT=") in new stack
        -- Executing [[email protected]:35] GotoIf("SIP/voipms-000057f1", "1?noautomon2") in new stack
        -- Goto (macro-hangupcall,s,41)
        -- Executing [[email protected]:41] NoOp("SIP/voipms-000057f1", "MONITOR_FILENAME=") in new stack
        -- Executing [[email protected]:42] GotoIf("SIP/voipms-000057f1", "1?skiprg") in new stack
        -- Goto (macro-hangupcall,s,45)
        -- Executing [[email protected]:45] GotoIf("SIP/voipms-000057f1", "1?skipblkvm") in new stack
        -- Goto (macro-hangupcall,s,48)
        -- Executing [[email protected]:48] GotoIf("SIP/voipms-000057f1", "1?theend") in new stack
        -- Goto (macro-hangupcall,s,50)
        -- Executing [[email protected]: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 [[email protected]: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>
    


  • Crap, that sucks.



  • 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.



  • @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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