[asterisk-users] Improving audio bitrate for all callers in a conference room for a podcast

[Patrick Davila]

Hello,
As a podcaster I use Asterisk extensively and often have several people in
a conference room. We'll record the calls via a SIP phone connected to a
sound mixer. Is there an easy way to bump up the audio bitrate for all
callers connected to the Asterisk server and improve the general sound
quality? The server is not used much outside of recording the podcast.
We're not opposed to compiling Asterisk ourselves to get the results we'd
like.

Any help is appreciated.
Thanks

Pat Davila

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