I have a bunch of mp3 talks that are constantly being updated and replace old talks. The old talk mp3s give this info:

mpg123 -v wed13124.mp3

MPEG 1.0, Layer: III, Freq: 22050, mode: Single-Channel, modext: 0, BPF : 480
Channels: 1, copyright: No, original: Yes, CRC: No, emphasis: 0.
Bitrate: 24 Kbits/s, Extension value: 0
Audio: 1:1 conversion, rate: 22050, encoding: signed 16 bit, channels: 1
Frame# 395 [133577], Time: 00:10.31 [58:09.35], ,
[0:10] Decoding of wed13124.mp3 finished.

These speeches are about an hour and have been compressed to ~ 10 MB. I have the updated speeches in wav files and want to compress them to ~ 10 MB. What lame or sox commands do I need to use?

Well, I would guess you just need to compress them to the same parameters as the original: 22050hz/16bit sample rate, 24kps bitrate, mono-channel. The man files of lame and sox should explain how to set these values.

You don't make it clear exactly, but I hope you aren't taking previously compressed files and recompressing them again. Taking an already-compressed mp3, converting to .wav, and then compressing it back again, especially at such low bitrates, will quickly make the files unlistenable, and every recompression will degrade the sound even more. If you're going to modify the files often, you should keep the originals in a lossless format such as .wav or .flac. The file size will be much larger, but the quality will remain pristine. Then if you need to distribute a smaller file you can create a compressed .mp3 file from the original.