audio for VOIP - filter and resample or viceversa? I am a noob to signal processing, but I need to create some audio files for IVR for Asterisk VOIP.

So my lady friend with a nice voice records her message on a Windows machine and sends me a 44KHz, stereo 32bit wav file.

I currently squash it into a 8KHz, mono, 16bit signed wav file for Linux, and then convert the wav to alaw or ulaw PCM format.

One source of info for this process is this link.

I can do some magic with Audacity:


Stereo to mono
Equalize, using a "voip" or "telephone" filter: discard anything over 3KHz and under 300Hz.
compress volume/levels to -12 to -15 dB
Resample 44 -> 8KHz
Export as 16bit signed PCM wav
Move it to Linux and sox it to alaw or ulaw


The result is acceptable, but I feel it could be improved.
In particular, after resampling, the voice sounds muffled.

Is this order correct? 

Do you have suggestions to get better results?
Your order is fine. The biggest question I have is with,


Equalize, using a "voip" or "telephone" filter: discard anything over 3KHz and under 300Hz.


Since you are resampling to 8 kHz in step 4, you already have less than 4 kHz, I don't see the point in the 3 kHz limit. And knowing nothing about the 3 kHz filtering you're using, I wonder how much intelligibility you're losing in that step. This is easy enough for you to test.

I believe you'll get better quality but less dynamic range with ulaw, compared to alaw. Other than this, the compression (step 3) is probably the next most important aspect to experiment with.