From 0a2bdcebca5e567b0388cc95bdb35363b30cd45e Mon Sep 17 00:00:00 2001
From: Henri Herscher <henri@oreka.org>
Date: Fri, 30 May 2008 15:17:52 +0000
Subject: 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
---
 orkaudio/config-linux-template.xml | 85 ++++++++++++++++++++++----------------
 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>
-- 
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