====== Conference room ======
The Internet Gate has built-in conference rooms (requires purchase of [[license:]]).

Besides acting as a conference server for audio conferences, it can detect DTMF tones of the conference participants and react to them with different actions. 

===== Audio files =====
Audio files for the conference unit are stored on the attached USB stick, in the directory /ivr. They have to be in .wav format, 8kHz sampling rate, mono, PCMA or PCMU encoded. You can copy them to this directory on a PC, or record them using the Record button on the conference setup GUI page, where you must specify a file name and the telephone to be called for making the recording.
The audio files are listed on the conference setup GUI page, and the names as shown in this list can be used in the menus below in the field "audio file name".
There is even the possibility to use audio files (in the format as above) which are located on a http server in the network, in that case the full network path for these files has to be specified (e.g. "http://myserver.com/audio/welcome.wav").

===== Menu editor =====
You can create interactive menus which can be used by the conference participant, e.g. for toggling between menus where the micorphone is muted or unmuted (see "conference mode" below).

You can have multiple submenus. For each menu you enter a descriptive menu ID, select what audio file (from the files listed in the **Audio files** section above) to play, if it shall be played one time or in a loop (endless), the conference mode for this menu (no conferencing / conference mode both speaking and listening / listen-only to conference). Also, you specify the duration of the inactivity timeout (time without DTMF key detections, then the action specified below under "Timeout" will be executed). 

Below you specify what will happen after the user presses a touchtone key on his telephone, and a possible action after no-digit-timeout and after the audio file playback is finished. The following actions are available:
  * **Transfer to** the entered SIP address - transfers the call to a human
  * **Go to menu ID** the entered menu ID - implements submenus
  * **Hang up** finishes the call
  * **Replay audio** starts the konfigured audio file for this menu again from the beginning