У меня немного другая задача, нужно писать телефонные переговоры с четырех аналоговых линий. Но похоже что oreka под linux не собирает sounddevice plugin.
Кстати под Windows oreka просто падает, как только включаешь sounddevice plugin. Собирать из исходников под Windows не пробовал.

Отдельное Спасибо Павлу Погодину, многое из его советов я использовал. Немного добавлю от себя вопросов. Копию письма отправил сегодня разработчикам, вот цитата (простите за ломанный инглиш):

Plugin sounddevice cannot be compiled from source in unix system ?

My installation procedure:

apt-get install subversion
apt-get install portaudio19-dev sox g++ libtool libxml2-dev liblog4cxx9-dev libace-dev libboost-dev libpcap0.8-dev libxerces-c2-dev libsndfile1-dev automake make

svn co https://oreka.svn.sourceforge.net/svnroot/oreka/trunk oreka

cd oreka/orkbasecxx/
cat /usr/share/aclocal/libtool.m4 >>aclocal.m4

automake -a
libtoolize --force
make -f Makefile.cvs
automake -a
make -f Makefile.cvs

./configure
make
make install
cd ../orkaudio/
cat /usr/share/aclocal/libtool.m4 >>aclocal.m4
libtoolize --force
automake -a
make -f Makefile.cvs
automake -a
make -f Makefile.cvs
./configure
make
make install

With these compilation procedure libsounddevice.la cannot be compiled. May be it is because /../orkaudio/audiocaptureplugins/Makefile.am
does not have directive like this:

METASOURCES=AUTO
SUBDIRS=sounddevice generator voip

but original Makefile.am:
METASOURCES=AUTO
SUBDIRS=generator voip

And last, /sounddevice directory in trunk does not have Makefile.am. I know that plugin depends from portaudio, ace, log4cxx and orkbase. But how can I compile it ?

P.S. my config.xml file:

<config>

<AudioOutputPath>./AudioRecordings</AudioOutputPath>
<!--<CapturePlugin>libvoip.so</CapturePlugin> -->

<!-- HOW TO COMPILE IT?????? -->
<CapturePlugin>libsounddevice.so</CapturePlugin>
<!--Why sounddevice.dll is present only in Windows precompiled package ? -->

<CapturePluginPath>audiocaptureplugins/</CapturePluginPath>

<StorageAudioFormat>gsm</StorageAudioFormat>

<!--deleting raw files-->
<DeleteNativeFile>yes</DeleteNativeFile>

<TrackerHostname>localhost</TrackerHostname>

<EnableReporting>false</EnableReporting>

<ClientTimeout>1000</ClientTimeout>

<NumBatchThreads></NumBatchThreads>

<AudioSegmentation>false</AudioSegmentation>
<AudioSegmentDuration>10</AudioSegmentDuration>

<SoundDevicePlugin>
<AudioChunkSize>8000</AudioChunkSize>

<SampleRate>8000</SampleRate>
  <SampleRate>8000</SampleRate>

<!-- May be it my mistake,but param "ports" from source, may be it is NumConcurentPorts? -->
<!--     <ports></ports>   -->

  <LogRms>false</LogRms>
<!--  Voice Activity Detection by default FALSE (from sources) -->
  <VAD>true</VAD>
  <!--by default is -12.5 (not better choice default leveling)-->
  <VadHighThresholdDb>-40</VadHighThresholdDb>
  <!--by default is -12.2 (why so small difference between params by default ?) -->
  <VadLowThresholdDb>-4</VadLowThresholdDb>
  
  <!--Hold the record after some timeout when VadLowThresholdDb lower than param -->
  <VadHoldOnSec>4</VadHoldOnSec>

</SoundDevicePlugin>

<VoIpPlugin>
        <!-- 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>-->
        
        <!-- 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>-->
        
        <!-- The following is a csv list of your PBX, PSTN gateway, conferencing server or such "gateway" devices -->
        <!--<MediaGateways>10.0.0.102</MediaGateways>-->
</VoIpPlugin>

<GeneratorPlugin>
<NumConcurrentPorts>1</NumConcurrentPorts>
<AudioDuration>5</AudioDuration>
<AudioFilename>sine.8KHz.pcm.wav</AudioFilename>
</GeneratorPlugin>

</config>

Есть у кого нибудь идеи ?