Re: [Oreka-user] problems with Orkaudio

[Henri H. <he...@or...>]

Hi Sergio,
 
Yes, the sound device plugin does not build yet on Linux target although it
should do without problems. It's just a matter of writing the right
Makefile.am and a bit of testing I would guess.
 
As for the generator plugin, you will need to use any pcm16 wav sound file,
rename it and it should be picked up by the plugin.
 
However, if you are interested in capturing asterisk audio, there's no need
to worry about either the generator or the sound device plugin, just use the
voip plugin.
 
Cheers
Henri

  _____  

From: ore...@li...
[mailto:ore...@li...] On Behalf Of Sergio Reyes
Sent: 02 January 2007 14:12
To: ore...@li...
Subject: [Oreka-user] problems with Orkaudio


Hello mates,

I have recently installed Orkaudio over Ubuntu, my idea is studying the
security of communications based on an asterisk PBX. I got some problems. 

First of all, I cant use the sound device plugin. In config.xml there are no
tags about it, there are only tags for generator plugin and VoIP. I have
noticed I dont have the library libsounddevice.so, but I think I installed
all the packages needed. Any clue?

Additionally, when using Generator plugin, although libgenerator.so is
correctly installed, it cant find the sin wav file, so it sends an empty
message instead.

sergio@sergio-desktop:/$ sudo orkaudio debug
Password:
2007-01-02 18:56:29,506  INFO root:117 -

OrkAudio service starting

2007-01-02 18:56:29,515  INFO root:93 - Loaded plugin:
/usr/lib/orkaudio/plugins/librtpmixer.so
2007-01-02 18:56:29,516  INFO root:87 - Loaded plugin:
/usr/lib/libgenerator.so
2007-01-02 18:56:29,517  INFO immediateProcessing:53 - thread starting -
queue size:10000 
2007-01-02 18:56:29,517  INFO batchProcessing:129 - thread Th0 starting -
queue size:20000
2007-01-02 18:56:29,518  WARN root:92 - Generator.dll: Could not load audio
test file:sine.8KHz.pcm.wav using empty buffer instead 
2007-01-02 18:56:29,519  WARN port:172 - #port0: received unexpected capture
event:stop
2007-01-02 18:56:29,519  INFO port:166 - #port0: start
2007-01-02 18:56:34,538  INFO port:184 - #port0: stop
2007-01-02 18:56:34,539  INFO tapelist:210 - date=2007-01-02_19-56-29
duration=5 direction=unkn capturePort=port0 localParty= remoteParty=
localEntryPoint= localIp= remoteIp= 
2007-01-02 18:56:34,539  INFO port:166 - #port0: start
2007-01-02 18:56:34,540  INFO batchProcessing:165 - Th0 processing:
20070102_195629_port0
2007-01-02 18:56:34,541  INFO reporting:79 - hostname=sergio-desktop
type=tape recid=20070102_195629_port0 stage=stop captureport=port0
timestamp=1167764189 filename=2007/01/02/19/20070102_195629_port0.wav
localparty= localentrypoint= remoteparty= direction=unkn duration=5
service=orkaudio-sergio-desktop localip= remoteip= 
2007-01-02 18:56:39,559  INFO port:184 - #port0: stop

And a third question, since Im just starting with asterisk, i still havent
implemented the PBX and clients. So I'd like to test the VoIP plugin on real
VoIP traffic. Do applications like skype send RTP traffic? cuz ethereal dont
see any RTP traffic when i start a conversation on Skype. Do you know any
application where I can generate real custom VoIP traffic so that i can test
orkaudio VoIP plugin? 

Thanks a lot in advance!