How to Improve recording audio quality in Asterisk

Does anybody have idea of improving the sampling rate of recording audio in Asterix FreePBX, default rate is 8k mono. We need to improve the audio quality for a good rate. Anybody have any idea on that.

This aligns with ulaw/alaw (G.711) and has generally done its job since the 70’s

What specific audio issues are you running in to?

Well although you ‘can’t polish a turd’ g711 is never going to be better than g711 and cell phones use gsm which is less than g711, but if everyone can agree on g722 then read for a quick overview

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