[asterisk-users] Using Asterisk to measure call quality: Introducing Recqual
Kristian Kielhofner
kristian.kielhofner at gmail.com
Tue Dec 23 15:17:00 CST 2008
On Tue, Dec 23, 2008 at 4:02 PM, Atis Lezdins <atis at iq-labs.net> wrote:
>
> Hi,
>
> This is good idea, and i will probably try it out someday next year
> (too busy completing my business requirements :)
Luckily next year is just over a week away. We won't have to wait
that long ;).
> I took a look at asterisk patch, and it seems quite simple. I just
> don't see the point of removing "if(debug)". You could easily get this
> additional logging into Asterisk trunk (if preserving RTP info in
> debug level), and starting asterisk with "debug 1". So, then it would
> be easier to install recqual. Also, being able to run on unmodified
> version of Asterisk, it would be good to allow keeping current
> dialplan and just route test calls trough it. So, people would be able
> to keep track of their billing, etc for those test calls.
This is true, however, I wasn't very excited about any other debug
messages that might get printed with "debug 1". I knew I only needed
the endpoint RTP address, so I just removed the if. Of course you
could always just run with "debug 1" instead of the patch too.
Again, this modification isn't strictly required. I just did if for
SIP providers that give unpredictable media endpoint IP addresses...
:)
> Also, thanks for showing us magics of ecasound. I have similar project
> (pbx-test-framework) that allows IVR/Queue/etc testing in automated
> mode. Recording everything and checking voice quuailty would be great
> addition :)
Ecasound is very, very cool. Recqual is only scratching the surface
of what it can do!
--
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com
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