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<config>
<!-- This is an example configuration file for the Oreka orkaudio capture service on Linux -->
<!-- Copy this to config.xml and modify according to taste -->
<AudioOutputPath>/var/log/orkaudio/audio</AudioOutputPath>
<!-- Uncomment the plugin you want to use: -->
<!-- Use libvoip.so for SIP, Cisco Skinny and pure RTP -->
<!-- Use libh323voip.so for Avaya, Nortel Unistim, H.323 and MGCP -->
<!-- See in <VoIpPlugin> below for more precise protocol tuning -->
<CapturePlugin>libvoip.so</CapturePlugin>
<!--<CapturePlugin>libh323voip.so</CapturePlugin>-->
<CapturePluginPath>/usr/lib</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>
<BatchProcessingEnhancePriority>true</BatchProcessingEnhancePriority>
<!--<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>eth1, eth2</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>
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