Hi,
I have been trying to play mp3 and wav files using freeswitch but noticed that the audio quality of these files are degraded. After looking for a while on logs, I think the sampling rate of these files goes down to 8K as I see “Codec Activated L16@8000 1Channels 20ms” in the fs_cli log… How can I retain the audio quality (sampling rate) of these files ?
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