I'd suggest using the currently recommended version of Lame - 3.97b2.  I've used in with VirtualDub a few times and had no problems.  You need to install the ACM in order to select it under Audio / Compression (I assume you already know how to do this).

For best results it's recommended that you keep the same sample rate as the source audio, e.g. if the source audio is at 48000Hz then compress to Lame MP3 @ 48000Hz @ the bitrate of your chosing.  You can view the sample rate of your source audio in vdub under File/File Information... The "view all audio formats" checkbox is there to keep users from making bad resampling choices ("good" choices being those you can see before checking the box). 

If your looking for highest audio quality, select the highest MP3 bitrate.  The ACM allows you to choose either CBR or ABR, though as I hear that VBR audio in AVI is non-standard, the safe bet is to choose CBR.  It's up to you to determine what's overkill and what isn't - 128kbps should be acceptable but you could go up to 320kbps if you really wanted...

I'm not goint to pretend to be an expert on digital video (hardly one on audio either ), but I'm pretty sure as long as it's MP3-standard (which Lame produces), most any computer today should be able to play the audio content.  Unfortunately I know next-to-nothing about which audio sample rates are compatable with which types of video/codecs.  If you're still having trouble you may want to post a question in the digital a/v subforum.

One last thing I can offer is that 32000 bits/sample might not be a valid sample rate for mpeg 1 layer 3, though IIRC Lame forces mpeg 2 (or something like that) for these sample rates.  I remember seeing a table somewhere here around HA that had this information - I just searched around for a while w/ no luck.  Perhaps one of the mp3 experts here can shine a little light on the subject. 

Also, make sure that your not confusing bit depth (e.g., 16-bits per sample, very common w/ digital audio) with bit rate (e.g., 16 kilobits per second, which would be very low quality audio).  For uncompressed audio:

bitrate (kbps) = sample rate (Hz)  x  bit depth (bits/sample)  x  # of channels  /  1000

Good luck!

If the audio is 32khz then leave it at 32khz (unless there is some reason why you need to change it), and only use CBR with AVI (even though ABR/VBR can work, don't use it)

As far as I can tell, you seem to be doing everything correctly in vdub.
What video codec are you using?
What decoders are you using for playback? (if you are using a directshow based player, what filters)

My guess on the other computer's playback problem would be that it is missing a decoder.

But now that the new VDub as audio compression, I realized that my vids were so big cuz of the AUDIO not being compressed. And since I can make the video a much higher quality at a good size if I also compress the audio, I am trying to figure it out.

Intermittent playback on other PC's could be due to the A/V interleaving in VDub.  I've found that if I compress with XVid and interleave MP3 audio, I get A/V sync problems on playback.  You can disable interleaving by going into the (surprise, surprise) "Interleaving" option under the "Audio" menu (if you see "Stream list" under "Audio", select that, then right-click on the audio stream and select "Interleaving").  Disabling interleaving shouldn't cause any problems when playing back from a hard drive (note that I've only tested this on three different PC's), but it won't work if you're trying to stream the files over the Internet.

It's hard to say why you're getting an error on playback with LAME 3.97b2 encoding, but I can tell you that if you're trying to encode at a different sample rate than your source audio, you need to go into the "Conversion" option in VDub's "Audio" menu and change the sample rate there (and be sure to enable the "high quality" option).  The LAME ACM codec evidently can't do sample rate conversion, so if your source audio is at 32 kHz and you try and encode at, say, 48 kHz, it will give you an error.

However, there's really no reason to do that as 32 kHz is most certainly a standard MP3 sampling rate (at bitrates of 32 kb/s and higher) and should be played properly by any MP3 decoder.

Edit: It turns out that the interleaving/sync issue may have been isolated to one source clip, or, more specifically, to one source clip using an external WAV file as the audio source - I haven't been able to reproduce the issue with a couple other clips, so YMMV.