Hi guys. First off, to the folks at Digium: outstanding work. The fact that Asterisk is open source puts you right at the cusp of what will be the most important telecom advance since the transatlantic cable.
Anyway... a couple newbie questions concerning sound quality - I don't see any reason why the system should not use the best possible format for any given connection.
1) Is it possible to store the menu sounds in wav/aiff, and let asterisk compress them to gsm only as necessary? Eg for POTS lines, yes the lines are crap already, but why butcher the sound any further by running it through a speech codec?
2) For my internal SIP phones, I don't care about bandwidth usage. What settings will give the best sound quality? Does the protocol (or for that matter, any particular brand of phones) support uncompressed or very high bit rate audio for intra-pbx calls?
Thanks, Sean
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