diff options
| author | Henri Herscher <henri@oreka.org> | 2008-06-05 20:00:49 +0000 |
|---|---|---|
| committer | Henri Herscher <henri@oreka.org> | 2008-06-05 20:00:49 +0000 |
| commit | 5c045efa07c404945b95269e92c3a9c2eb2c2758 (patch) | |
| tree | e2794f9ed9d07d543fded3bfcf53f1c6269d2e17 | |
| parent | 0a2bdcebca5e567b0388cc95bdb35363b30cd45e (diff) | |
Corrected linux voip plugin name in config.xml template to libvoip.so.
git-svn-id: https://oreka.svn.sourceforge.net/svnroot/oreka/trunk@543 09dcff7a-b715-0410-9601-b79a96267cd0
| -rw-r--r-- | orkaudio/config-linux-template.xml | 50 |
1 files changed, 25 insertions, 25 deletions
diff --git a/orkaudio/config-linux-template.xml b/orkaudio/config-linux-template.xml index 71f4725..f95f31d 100644 --- a/orkaudio/config-linux-template.xml +++ b/orkaudio/config-linux-template.xml @@ -2,16 +2,16 @@ <!-- 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>/var/log/orkaudio</AudioOutputPath> - <!--<AudioOutputPath>/opt/tomcat5/webapps/ROOT</AudioOutputPath>--> + <!--<AudioOutputPath>/opt/tomcat5/webapps/ROOT</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>voip.so</CapturePlugin> - <!--<CapturePlugin>libh323voip.so</CapturePlugin>--> + <!-- 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> @@ -29,25 +29,25 @@ <BatchProcessingEnhancePriority>true</BatchProcessingEnhancePriority> - <!--<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 --> @@ -57,15 +57,15 @@ <!--<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>--> + <!--<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>--> + <!-- 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>--> + <!-- Sangoma RTP tap for TDM boards --> + <!--<SangomaRxTcpPortStart>9000</SangomaRxTcpPortStart>--> + <!--<SangomaTxTcpPortStart>11000</SangomaTxTcpPortStart>--> </VoIpPlugin> </config> |