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This option initiates a simple test call from within the 

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 and 

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. The test call uses the specified Call Routing Table to find the outgoing signaling group to call a SIP endpoint or a phone through one of the TDM interfaces - T1/E1, FXS, FXO.

You can test a call for a SIP phone or Analog phone.

Testing a Call

  1. In the WebUI, click the Diagnostics tab.
  2. In the left navigation pane, under Tools, click Test a Call

    Caption
    0Figure
    1Test a Call

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  1. Image Added

  2. Enter

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  1. the Destination Phone Number. This is the destination number of the SIP or analog phone being called (called party number).
  2. Enter the optional Origination/Calling Number.

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  1.  This number appears on the phone receiving the call as the calling party

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  1. number to identify the call as the test call.
  2. From the Call Routing Table drop down list, select a Call Routing Table. Ensure the Call Routing Table you select is configured properly. For information about Call Routing tables, see Managing Call Routing Tables.

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  1. Click OK.

Expected Outcome

Once the call is made, the call goes through the selected Call

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Routing Table. After the

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called party answers, the

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music file is played (the same file that is loaded on the SBC for Music on Hold).

For an unsuccessful call, check the following:

  • Verify the Call Routing Table entry selected. Also check the Translation table for configuration. These tables determine how a call is routed and translated. 
  • Review Local

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  • Log files. Log files record internal and external action on the SBC 2000 (these log files are not available on the SBC1000), including user visible actions (such as placing calls).
  • Verify Music On Hold configuration. If configured to use a music file, but there is no music file, no music is played. If configured to use radio (live source), but there is no valid FXS port, no music is played.

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  • Live source through a FXS port is only applicable to SBC Edge, not SWe Lite.

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