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i've just installed an asterisk 1.8 IVR using playback() for playing sln files.

i've we have some high quality wav files, that i've converted with sox to sln 16KHz files. (Why aren't higher sampling rates supported?)

when i call the asterisk IVR system the playback quality is really bad. What can i do the enhance the quality so that the callee can actually understand the IVR?

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Higher quality recording doesn't necessarily map to better sound quality in telephony - particularly given the nature and history of voice codecs. You need to make sure you've either recorded- or down-sampled- your file to an 8khz sampling rate and 16-bit samples (i.e. PCM).

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