Asterisk/app_rpt will play MP3 files alright, using the add-on, although, you may need to change the output level. In format_mp3.c #define OUTSCALE 32768 Seems to be about right (I think the original was 8192 but YMMV as they say) I used mp3 for bulletins for a while - I even bought a copy of Adobe Audition CS5.5 (now there is version 6) so I could make "real" MP3 files. MP3's made by Audacity sound *really* weird. MP3 files contain a frequency spectrum that depends on the sampling rate of the recording. This is a separate rate from the Bit Rate of the MP3 playback. If you reduce the playback bitrate of an MP3 file, you may still have frequency components up through perhaps 12 or 15 kHz, you just get more distortion and artifacts. All those frequency components above about 3.6 KHz will be ignored by Asterisk, which runs its sound system at 8 k-samples/sec. This may produce some strange effects since you are playing back stuff that Asterisk can't "hear". The distortion below 3.6 kHz comes through of course... To get a file like ARRL Audio News (44 kHz sample rate, in stereo, with an MP3 bit rate of 192k) to play properly on asterisk, you need some software that can re-sample the original file down to 8 k sampling rate (and 8k bit rate). That way the playback spectrum of the MP3 will have components only up through about 3.4 kHz, which is quite compatible with asterisk. Adobe Audition 5.5 (and later) can do this. Audacity will also re-sample (at the "Project Rate"), but it is best used to make ulaw files, not MP3's; but it's free. If the original recording was frequency-limited then no re-sampling of the recording is needed. For example, my NWS weather podcasts are a computer voice designed for telephone bandwidth, but distributed as MP3's. Those MP3's played through asterisk sound just like the original played on any MP3 player. Lots of fun 73 Ken