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FOP2 Community Forum

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  3. Can't listen call after redirect to gateway

Can't listen call after redirect to gateway

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  • G
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    gr33tx
    wrote on last edited by
    #1

    Hi! Can anyone help?
    I have internal extensions and VOIP-gateway. I can listen internal calls with FOP2. But when redirect to external number to VOIP-gateway taking place FOP2 can't listen calls. But this call is recording by Asterisk correctly.
    Why Asterisk can record this calls and FOP2 can't listen them?

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    • N
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      nicolas
      wrote on last edited by
      #2

      FOP2 does not listen calls, it just fires the chanspy asterisk command. So your question should be instead:

      Why chanspy cannot monitor a call , but Monitor or Mixmonitor can record it to disk?

      I cannot give you the answer, as in many many scenarios, chanspy works fine, no matter if you are using a gateway or not. Maybe asterisk sends a reinvite to not allow a direct rtp path between endpoints when using monitor/mixmonitor but fails to do that when using chanspy.

      In any case, it is not a fop2 issue, but an asterisk issue.

      Best regards,

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      • G
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        gr33tx
        wrote on last edited by
        #3

        Thank you admin!

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        • G
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          gr33tx
          wrote on last edited by
          #4

          I have assumption.
          FOP2 can't see call activity between external calls and my 3 digits extensions that have custom dial (such as SIP/10000/388501).

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            nicolas
            wrote on last edited by
            #5

            I have assumption.
            FOP2 can't see call activity between external calls and my 3 digits extensions that have custom dial (such as SIP/10000/388501).

            FOP2 monitor devices... in the above SIP string, the device is SIP/10000 . So if you have a button to monitor SIP/10000 everything should work.

            Now, if SIP/10000 is a gateway that dial remote extensions, then you will be able to monitor that as a trunk, not distinct extensions, as they are not distinct channels/devices. Once you dial SIP/10000/whatever or SIP/whatever@10000, ASTERISK will create a channel SIP/10000-randomsession, then you call a differente extension, then a new channel will be created SIP/10000-randomsession. So, you have two calls, maybe for two separate extensions, and both are named by asterisk SIP/10000, and that is all what FOP2 will monitor and work with.

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              gr33tx
              wrote on last edited by
              #6

              Thanks a lot!
              With Channel and External Transfer in Button options for such extensions its works fine. 😄

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