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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