I have recently deployed asterisk 1.2.1 on a 64bit box running gentoo and when i ring a menu the voice prompt plays back at 3 times the original sampling rate and has a slight jitter.
Normal calls between sip phones or over the zap interface sound ok, its only the play back of any recordings, wav, gsm both playing back at a faster sampling rate.
The samples were purchased from digium and therefore I expect have been made to the correct sampling standard.