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Add volume, gain and energy controls.
DTMF: 1 = volume down 2 = volume default 3 = volume up 4 = gain down 5 = gain default 6 = gain up 7 = energy level down 8 = energy level default 9 = energy level up * = toggle mute + deaf 0 = toggle mute # = quit new api commands to adj volume, gain and energy git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@1787 d0543943-73ff-0310-b7d9-9358b9ac24b2
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@@ -273,6 +273,7 @@
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you need to really set them up -->
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<configuration name="conference.conf" description="Audio Conference">
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<!-- Profiles are collections of settings you can reference by name. -->
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<profiles>
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<profile name="default">
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<!-- Sample Rate-->
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@@ -281,6 +282,14 @@
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<param name="interval" value="20"/>
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<!-- Energy level required for audio to be sent to the other users -->
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<param name="energy_level" value="300"/>
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<!-- TTS Engine to use -->
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<!--<param name="tts_engine" value="cepstral"/>-->
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<!-- TTS Voice to use -->
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<!--<param name="tts_voice" value="david"/>-->
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<!-- If TTS is enabled all audio-file params not beginning with '/'
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will be considered text to say with TTS -->
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<!-- File to play to acknowledge succees -->
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<!--<param name="ack_sound" value="/soundfiles/beep.wav"/>-->
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<!-- File to play to acknowledge failure -->
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@@ -305,6 +314,10 @@
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<!--<param name="bad_pin_sound" value="/soundfiles/invalid-pin.wav"/>-->
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<!-- Conference pin -->
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<!--<param name="pin" value="12345"/>-->
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<!-- Default Caller ID Name for outbound calls -->
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<param name="caller_id_name" value="FreeSWITCH"/>
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<!-- Default Caller ID Number for outbound calls -->
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<param name="caller_id_number" value="8777423583"/>
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</profile>
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</profiles>
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<rooms>
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