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    Wednesday, April 17, 2013

    Hanging up active calls in Asterisk PBX

    There are many times when we run out of free channels in your PBX while making calls or in case a phone is not placed properly the calls does not gets disconnected and is shown as busy on the PBX. So at those moments it becomes a job of Administrator to monitor these logs and accordingly hangup unwanted calls to free up the channels.

    Well I had to surf a lot to find the exact command to hangup calls in the latest Free PBX so here I will show you in easy steps for how to hangup active calls on PBX.

    Login to your asterisk CLI console
    asterisk2*CLI> core show channels
    Channel              Location             State   Application(Data)
    SIP/3224-00000a19    s@macro-dial-one:42  Up      Dial(SIP/4027,15,trI)
    IAX2/IAX_Trunk_to_US (None)               Up      AppDial((Outgoing Line))
    SIP/4003-00000a2f    s@macro-dialout-trun Up      Dial(IAX2/IAX_Trunk_to_US/1001
    SIP/4001-0000089e    s-CHANUNAVAIL@macro- Up      VoiceMail(3102@default,u"")
    SIP/3117-00000102    s-NOANSWER@macro-vm: Up      VoiceMail(3106@default,u"")
    SIP/4027-00000a1a    (None)               Up      AppDial((Outgoing Line))
    6 active channels
    4 active calls
    1553 calls processed
    As you can see in my case there are 4 active channels and I want to disconnect user 4003 for example.
    asterisk2*CLI> channel request hangup SIP/4003-00000a2f
    Requested Hangup on channel 'SIP/4003-00000a30'
    -- Executing [h@macro-dialout-trunk:1] Macro("SIP/4003-00000a30", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/4003-00000a30", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] ExecIf("SIP/4003-00000a30", "0? Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:4] Hangup("SIP/4003-00000a30", "") in new stack
    == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/4003-00000a30' in macro hangupcall'
    == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/4003-00000a30'
    -- Hungup 'IAX2/IAX_Trunk_to_US-49'
    Now user 4003 has been disconnected as you can verify below.
    asterisk2*CLI> core show channels
    Channel              Location             State   Application(Data)
    SIP/3224-00000a19    s@macro-dial-one:42  Up      Dial(SIP/4027,15,trI)
    SIP/4001-0000089e    s-CHANUNAVAIL@macro- Up       VoiceMail(3102@default,u"")
    SIP/3117-00000102    s-NOANSWER@macro-vm: Up       VoiceMail(3106@default,u"")
    SIP/4027-00000a1a    (None)               Up      AppDial((Outgoing Line))
    4 active channels
    3 active calls
    1554 calls processed


    Related Articles
    Connecting two IP PBX box using SIP Trunk


    Follow the below links for more tutorials:

    3 comments:

    1. Thanks for the guide! It was a real lifesaver a moment ago when one agent got stuck in a conference call and was unable to leave it.

      ReplyDelete
    2. Thank You! Works like a charm!

      ReplyDelete
    3. I have a big problem.

      We have Elastix here, but some calls no matter what extensions hangs out, and our knowledge in IP telephony is not good enough. So, could anybody help me out to find what is wrong with.
      I really apprecciate your help!

      ReplyDelete