Testing a Call

This option initiates a simple test call from within the 

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. 
   
   

   Test a Call

   

3. Enter the Destination Phone Number. This is the destination number of the SIP or analog phone being called (called party number).
4. Enter the optional Origination/Calling Number. This is the phone number of the SIP phone or analog phone which should appear on the phone as the calling party number.
5. 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.
6. Click OK.
   
   

Expected Outcome

Once the call is made, the call goes through the selected Call Route. After the SIP phone or Analog phone accepts the call, the user will hear music for about 15 seconds.

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 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.
• If configured to use radio (live source), but there is no valid FXS port, no music is played (applies to SWe Lite only).