Tech Blog

Seem to be on a roll with Sox tonight. Also figured out how to normalise audio files nicely.

The first thing to do with sox is to get it to calculate the max volume adjustment possible:

This will return a number like: 4.234

You can then call sox again using this number:

And if you're feeling particularly good you can put the whole command on a single line as follows:

Easy when you know how.

However, this time I finally managed to get past all the hurdles and jump through all the hoops to make this work.

Scenario is a follows. Audio file is comes in in alaw or ulaw format and we want to trim any silence before and after it.

First step is to convert it from alaw to pcm encoded wav file as SOX has issues filtering silence in alaw files

Then we do the trimming in one fell swoop.

A huge thanks to this article for giving such a thorough overview of the how silence detection works in sox.

Too easy, when you know how :-)

There is a bit of setup to be done but when you get the whole picture is it's actually very easy. To do it you need at least 2 asterisk boxes - the one you are load testing (lets call it 'stressbox' for this example) and a another one to drive load to it which we'll call 'driver'.

We need to do the following steps:

• Create an sip account on stressbox for driver to connect to
• Register the account from the driver machine
• Create a dial plan which simulates a call
• Create a call file for the call
• Start you system monitoring
• Create lots of call files

So here is the step by step:

Create an sip account on stressbox for driver to connect to

In the sip.conf file on stressbox (the one we are testing) add the following:

Register the account from the driver machine

Your driver machine should now be able to make calls to the stressbox machine and you should see a line that looks like this if you were watching the console:

Create a dial plan which simulates a call

Create a call file for the call

A bit of explanation of these lines: Channel: SIP/6@xxx.xxx.xxx.xxx - the 6 is the extension to dial on the stressbox server - change as necessary. The ip adress should be the stressbox one. Context: dialplan-loadtest1 - this is the dial plan that will be run for this call.

Start you system monitoring

For this I use a vmstat on linux via a little script I wrote which makes it easy to collect stats from multiple machines

It would be used as follows:

Create lots of call files

You are not ready to test you system. To do this you need to create copies the test.call file and move them into /var/spool/asterisk/outgoing on the driver machine.

I use a little script to do this:

This is used as follows:

This will make 5 calls with a 2 second delay between the start of each call.

Note: It needs to be run as sudo so it can change the ownership correctly.

Credits: Thanks to Stuart Elvish for his help with getting this working so nicely.

Happy load testing. Cheers, Mark

It works great, but when I tried it in my new install the sound was all choppy, i.e. I would get every half of each second of the audio.

A bit of fiddling about with the settings and I got it solved. Posting it here so when I forget about it and run into the same problem again I'll know how to fix it.

The solutions is as simple as:

• From the menu select Edit -> System Settings.
• Select Audio on the left
• Change the audio devices to use the most approriate device that is NOT "ALSA: default: Default Device" - in my case it was "ALSA: plughw:0,0: HDA Intel (AD193x Analog)"

I do like the very friendly naming scheme for devices :-) Hope it helps, Mark

For this example I'm going to replace the standard digits of 0 to 10 with files that I've had recorded.

Create files in the correct format:

We can convert them to alaw, ulaw and gsm formats which reduces asterisk transcoding with the following script: (requires sox to be installed)

Move the files to the correct place on the server:

I'm going to use the name 'learnosity' for the language as these are the files that we've recorded. So we need to create the following directory on the asterisk machine:

If you wanted to do more audio files, you can create learnosity directories for the other audio files.

Once this is done, copy the alaw, ulaw and gsm files in but leave the original wav files out.

Tell asterisk to use these files:

The easiest way to use these language files is to set the channel language for the sections you want to use it:

Here's an example from extensions.conf:

If you dial extension 4 you get the digits 1 to 9 in the default language and if you've done everything correctly you will get your newly recorded digits the second time.

Cheers, Mark

If you are looking to work with the latest technogies and are a motivated self starter with a positive "can do" attitude this is the job for you. The ability to take problems and deliver solutions is a must.

There will be significant on the job learning as we are always pushing the boundaries and using the latest and greatest technologies. We are currently working on projects using Actionscript/Flex, ColdFusion, PHP, and Javascript, and are leveraging technologies such as VOIP, SMS, Instant Messenger (Jabber XMPP) to deliver cutting edge solutions.

Suitable applicants will have a Bachelors Degree in Computer Science or Engineering, or similar tertiary qualification. Remuneration commensurate with experience.

Responsibilities would include:

• Developing new functionality for Asterisk and VOIP applications
• Creating Rich Internet Applications with Adobe Flex
• Creation of Desktop applications with Adobe AIR
• Design and development of new Web applications

Must haves:

• A robust foundation in Object Oriented programming
• Demonstrated experience in at least one OO language
• Motivation to learn and push the boundaries
• Understanding of XHTML and CSS

Preferable:

• Experience with PHP, ColdFusion, Farcry CMS, XML, Web Standards etc
• Competency using linux-based tools (SSH, bash, etc)
• Familiar with using source control tools (Subversion, etc)
• Familiarity with Actionscript or Flash

This role may be full time, part-time or on a contract basis depending on the candidates skills and experience. You will be working in a casual workplace with flexible hours in the centre of Sydney.

If this sounds like the job for you email your resume to mark@lynchconsulting.com.au if interested - no agencies please.