<div dir="ltr">Are you using a wideband codec? Most telephone codecs use a 8000 rate so you'll get that drop.</div><div class="gmail_extra"><br><br><div class="gmail_quote">On 17 October 2013 16:15, Bharath Pareek <span dir="ltr"><<a href="mailto:bharathpareek@gmail.com" target="_blank">bharathpareek@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><p>Hi,</p>
<p> </p>
<p>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 ?</p><span class="HOEnZb"><font color="#888888">
<p> </p>
<p>-- <br></p>Regards,<div>Bharath Pareek</div>
</font></span></div>
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