SIP Usage - Dial Out

If the SIP module is available then a link to "Dial Out" will be displayed on the front page. This service allows users to include SIP phones as well as any standard and mobile telephone into an e-meeting.

This means that you can bring in participants that only have access to a telephone, in their home, office or on their cell phone. These can participate in the e-meeting using audio.

From the Current e-meeting room page in the Viewer, the Dial Out function requires two parameters:

Phone number: The number to call. Example: peter@195.196.42.45 or 15555551212. No spaces or hyphens are allowed.

Line: Which SIP line should be used for this call? This list is dynamically created each time the invite page is shown and it includes a list of currently available SIP lines that can be used for this call. This is based on the current user credentials (i.e. if the user is logged into the Manager or not and by which user) and how access to the SIP line is configured by the administrator. A line might not be available if there are other currently active calls using the line. Not available lines will not be shown in the line list.

To make call, please press Call. Progress about the call progress is now shown with the following attributes:

  • Dialing: The call is being set up.
  • Ringing: It is ringing at the callee.
  • Busy: The callee was busy.
  • Failed: The call could not be completed. This can be due to a number of reasons. The SIP line might not be configured correctly, a network error might have occurred or the SIP server might not have been able to complete the call.
  • Not found: This means that a call could not be made to this callee. The most common reasons include the number does not exist or there was no such callee at SIP server or SIP destination.
  • Connected: Means that the call was completed.
  • Disconnected: Means that the call was disconnected. E.g. the callee hang up

To cancel the calling process, please press the Hangup button

When the call is in connected state, audio data is flowing between the callee and the Manager and into the e-meeting room.

Using the SIP Status link in the menu a list over available active calls is presented. After each active call in the list two buttons are present, Hangup and Send dial tones . The former cancels the call and the latter shows the DTMF Send page (see below). Note, that only calls that the user has made or has access to are shown. To see a list of all active calls please refer to the SIP Line Call Status page in the administrator interface.

The Send Dial Tones page is used to send DTMF tones (i.e. key beeps normally used to interact with automated phone service. E.g. selecting menu items). DTMF tones can be sent to a SIP callee by either entering a string to be sent or by pressing the individual phone keys displayed. The DTMF tones will not be heard by others in the E-meeting room or other SIP callees. Example: 123ab*90*42#da1243*

When using DTMF tones, ask your colleagues to keep silent. It will help the DTMF tones to get through.


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