diff options
-rw-r--r-- | orkaudio/config-template.xml | 72 |
1 files changed, 36 insertions, 36 deletions
diff --git a/orkaudio/config-template.xml b/orkaudio/config-template.xml index d74595b..866b94e 100644 --- a/orkaudio/config-template.xml +++ b/orkaudio/config-template.xml @@ -2,68 +2,68 @@ <!-- This is an example configuration file for the Oreka orkaudio capture service on Windows --> <!-- Copy this to config.xml and modify according to taste --> - <!-- Change this to point to Tomcat if you run the OrkWeb user interface --> + <!-- Change this to point to Tomcat if you run the OrkWeb user interface --> <AudioOutputPath>./AudioRecordings</AudioOutputPath> - <!--<AudioOutputPath>C:\Program Files\Apache Software Foundation\Tomcat 5.5\webapps\ROOT</AudioOutputPath>--> + <!--<AudioOutputPath>C:\Program Files\Apache Software Foundation\Tomcat 5.5\webapps\ROOT</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 --> + <!-- 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>--> - + <!--<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> <EnableReporting>true</EnableReporting> <CapturePortFilters>LiveMonitoring</CapturePortFilters> <TapeProcessors>BatchProcessing, Reporting</TapeProcessors> - - <!--<TapeDurationMinimumSec>3</TapeDurationMinimumSec>--> - + + <!--<TapeDurationMinimumSec>3</TapeDurationMinimumSec>--> + <VoIpPlugin> - <!-- Use this for Nortel proprietary VoIP protocol --> - <!--<UnistimDetect>yes</UnistimDetect>--> + <!-- 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>--> + <!-- 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> --> - <!-- 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>--> - + + <!--<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 RTP tap for TDM boards --> - <!--<SangomaRxTcpPortStart>9000</SangomaRxTcpPortStart>--> - <!--<SangomaTxTcpPortStart>11000</SangomaTxTcpPortStart>--> + <!--<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 RTP tap for TDM boards --> + <!--<SangomaRxTcpPortStart>9000</SangomaRxTcpPortStart>--> + <!--<SangomaTxTcpPortStart>11000</SangomaTxTcpPortStart>--> </VoIpPlugin> </config> |