diff options
author | Henri Herscher <henri@oreka.org> | 2008-05-30 15:17:52 +0000 |
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committer | Henri Herscher <henri@oreka.org> | 2008-05-30 15:17:52 +0000 |
commit | 0a2bdcebca5e567b0388cc95bdb35363b30cd45e (patch) | |
tree | ff8531b5953d216930ee8d597e1448a4ba6c75c7 | |
parent | c77ed8ceb4a6576253be4604830399e1280593e9 (diff) |
Brought orkaudio config.xml templates up to date.
git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@542 09dcff7a-b715-0410-9601-b79a96267cd0
-rw-r--r-- | orkaudio/config-linux-template.xml | 85 | ||||
-rw-r--r-- | orkaudio/config-template.xml | 55 |
2 files changed, 83 insertions, 57 deletions
diff --git a/orkaudio/config-linux-template.xml b/orkaudio/config-linux-template.xml index 7cd689e..71f4725 100644 --- a/orkaudio/config-linux-template.xml +++ b/orkaudio/config-linux-template.xml @@ -1,58 +1,71 @@ <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 --> + <!-- 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 --> <AudioOutputPath>/var/log/orkaudio</AudioOutputPath> + <!--<AudioOutputPath>/opt/tomcat5/webapps/ROOT</AudioOutputPath>--> - <!-- Use the following if oreka has been installed (having run "make install") --> - <CapturePluginPath>/usr/lib</CapturePluginPath> - <CapturePlugin>libvoip.so</CapturePlugin> - <!-- Use the following if you run oreka uninstalled (without having run "make install") --> - <!--<CapturePluginPath>audiocaptureplugins/voip/.libs</CapturePluginPath>--> - <!--<CapturePlugin>libvoip.so</CapturePlugin>--> - + <!-- 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>voip.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> - + <DeleteNativeFile>yes</DeleteNativeFile> + <TrackerHostname>localhost</TrackerHostname> <EnableReporting>true</EnableReporting> - - <AudioSegmentation>false</AudioSegmentation> - <AudioSegmentDuration>10</AudioSegmentDuration> - - <BatchProcessingEnhancePriority>true</BatchProcessingEnhancePriority> - <CapturePortFilters>LiveMonitoring</CapturePortFilters> - <TapeProcessors>BatchProcessing, Reporting</TapeProcessors> + <CapturePortFilters>LiveMonitoring</CapturePortFilters> + <TapeProcessors>BatchProcessing, Reporting</TapeProcessors> + <BatchProcessingEnhancePriority>true</BatchProcessingEnhancePriority> + + <!--<TapeDurationMinimumSec>3</TapeDurationMinimumSec>--> + <VoIpPlugin> - <!-- Set the option below to "false" to disable IAX2 support --> - <!-- the default is that IAX2 support is enabled --> - <!--<Iax2Support>true</Iax2Support> --> + <!-- 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 /etc/orkaudio/orkaudio.log when the service is starting --> - <!--<Devices>eth1, eth2</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>--> - - <!-- LanMasks can be ignored if you have standard LAN addresses (192.168.x.x or 10.x.x.x) --> - <!-- LanMasks might be used to determine the direction of a call (incoming or outgoing into or from the LAN) --> - <!--<LanMasks>10.4.3.4, 1.2.3.4</LanMasks>--> + <!--<BlockedIpRanges>212.125.143.0/24, 82.150.0.0/16, 82.199.64.133</BlockedIpRanges>--> - <!-- The following is a csv list of your PBX, PSTN gateway, conferencing server or such "gateway" devices --> - <!--<MediaGateways>10.0.0.102</MediaGateways>--> + <!--<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>--> </VoIpPlugin> - <GeneratorPlugin> - <NumConcurrentPorts>1</NumConcurrentPorts> - <AudioDuration>5</AudioDuration> - <AudioFilename>sine.8KHz.pcm.wav</AudioFilename> - </GeneratorPlugin> </config> diff --git a/orkaudio/config-template.xml b/orkaudio/config-template.xml index a606223..d74595b 100644 --- a/orkaudio/config-template.xml +++ b/orkaudio/config-template.xml @@ -2,12 +2,17 @@ <!-- 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 --> <AudioOutputPath>./AudioRecordings</AudioOutputPath> + <!--<AudioOutputPath>C:\Program Files\Apache Software Foundation\Tomcat 5.5\webapps\ROOT</AudioOutputPath>--> - <!-- Uncomment the plugin you want to use --> + <!-- 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>Generator.dll</CapturePlugin>--> - <!--<CapturePlugin>SoundDevice.dll</CapturePlugin>--> + <!--<CapturePlugin>H323voip.dll</CapturePlugin>--> + <CapturePluginPath>audiocaptureplugins/</CapturePluginPath> <!-- Audio file storage format: choose from: native, gsm, ulaw, alaw, pcmwav --> @@ -18,21 +23,30 @@ <TrackerHostname>localhost</TrackerHostname> <EnableReporting>true</EnableReporting> - - <AudioSegmentation>false</AudioSegmentation> - <AudioSegmentDuration>10</AudioSegmentDuration> <CapturePortFilters>LiveMonitoring</CapturePortFilters> <TapeProcessors>BatchProcessing, Reporting</TapeProcessors> + <!--<TapeDurationMinimumSec>3</TapeDurationMinimumSec>--> + <VoIpPlugin> - <!-- Set the option below to "false" to disable IAX2 support --> - <!-- the default is that IAX2 support is enabled --> - <!--<Iax2Support>true</Iax2Support> --> + <!-- 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>--> @@ -40,17 +54,16 @@ <!-- matching the list are dropped --> <!--<BlockedIpRanges>212.125.143.0/24, 82.150.0.0/16, 82.199.64.133</BlockedIpRanges>--> - <!-- LanMasks can be ignored if you have standard LAN addresses (192.168.x.x or 10.x.x.x) --> - <!-- LanMasks might be used to determine the direction of a call (incoming or outgoing into or from the LAN) --> - <!--<LanMasks>10.4.3.4, 1.2.3.4</LanMasks>--> - - <!-- The following is a csv list of your PBX, PSTN gateway, conferencing server or such "gateway" devices --> - <!-- It is needed to properly detect call direction --> - <!--<MediaGateways>10.0.0.102</MediaGateways>--> + <!--<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>--> </VoIpPlugin> - <GeneratorPlugin> - <NumConcurrentPorts>1</NumConcurrentPorts> - <AudioDuration>5</AudioDuration> - <AudioFilename>sine.8KHz.pcm.wav</AudioFilename> - </GeneratorPlugin> </config> |