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 -- http://tllts.org/ - The Linux Link Tech Show http://mythtvcast.com/ - MythTVCast http://patdavila.wordpress.com - My blog -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
