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<config>
	<!-- This is an example configuration file for the Oreka orkaudio capture service on Windows -->
	<!-- Copy this to config.xml and modify according to taste -->

	<AudioOutputPath>c:\oreka\audio</AudioOutputPath>

	<!-- Uncomment the plugin you want to use: -->
	<!-- Use VoIP.dll for SIP, Cisco Skinny and pure RTP -->
	<!-- Use H323voip.dll for Avaya, Nortel Unistim, H.323 and MGCP -->
	<!-- See in <VoIpPlugin> below for more precise protocol tuning -->
	<CapturePlugin>VoIP.dll</CapturePlugin>
	<!--<CapturePlugin>H323voip.dll</CapturePlugin>-->

	<CapturePluginPath>audiocaptureplugins/</CapturePluginPath>	

	<!-- Audio file storage format: choose from: native, gsm, ulaw, alaw, pcmwav -->
	<StorageAudioFormat>gsm</StorageAudioFormat>

	<!-- If you want to keep native audio files as well as compressed, change this to "no" -->
	<DeleteNativeFile>yes</DeleteNativeFile>

	<TrackerHostname>localhost</TrackerHostname>
    <TrackerTcpPort>8080</TrackerTcpPort>
    
	<CapturePortFilters>LiveMonitoring</CapturePortFilters>
	<TapeProcessors>BatchProcessing, Reporting</TapeProcessors>

	<!--<TapeDurationMinimumSec>3</TapeDurationMinimumSec>-->

	<VoIpPlugin>
		<!-- Use this for Nortel proprietary VoIP protocol -->
		<!--<UnistimDetect>yes</UnistimDetect>-->

		<!-- Turn both these on this for Avaya H.323 extensions -->
		<!--<AvayaDetect>yes</AvayaDetect>-->
		<!--<RtcpDetect>yes</RtcpDetect>-->

		<!-- Set the option below to "false" to disable IAX2 support -->
		<!-- the default is that IAX2 support is enabled -->
		<!--<Iax2Support>true</Iax2Support> -->

		<!-- Use this if you want to force capture from a given list of devices. -->
		<!-- All available devices are listed in orkaudio.log when the service is starting -->
		<!--<Devices>\Device\NPF_{E0E496FA-DABF-47C1-97C2-DD914DFD3354}, \Device\NPF_{ADE496FA-DABF-47C1-97C2-DD914DFDAB38}</Devices>-->

		<!--<PcapFilter>net 217.14.0.0/16 or host 10.0.0.1</PcapFilter>-->

		<!-- If AllowedIpRanges is used, only packets with *both* source and destination -->
		<!-- matching the list are retained -->
		<!--<AllowedIpRanges>212.125.143.0/24, 82.150.0.0/16, 82.199.64.133</AllowedIpRanges>-->
		<!-- If BlockedIpRanges is used, packets with *either* source or destination -->
		<!-- matching the list are dropped -->
		<!--<BlockedIpRanges>212.125.143.0/24, 82.150.0.0/16, 82.199.64.133</BlockedIpRanges>-->				

		<!--<SipOverTcpSupport>yes</SipOverTcpSupport>-->
		<!--<SipReportFullAddress>yes</SipReportFullAddress>-->
		<!--<SipUse200OkMediaAddress>yes</SipUse200OkMediaAddress>-->

		<!-- Those two parameters are only needed for call direction detection (one or the other) -->
		<!--<SipDomains>company.com, 65.34.25.87</SipDomains>-->
		<!--<SipDirectionRefenceIpAddresses>65.34.98.56, 65.34.98.57</SipDirectionRefenceIpAddresses>-->

		<!-- Sangoma wanpipe RTP tap for TDM boards -->
		<!--<SangomaRxTcpPortStart>9000</SangomaRxTcpPortStart>-->
		<!--<SangomaTxTcpPortStart>11000</SangomaTxTcpPortStart>-->
	</VoIpPlugin>
</config>