> > You could trying changing this in sip.cfg > <AES voice.aes.hs.enable="0" > To > <AES voice.aes.hs.enable="1" > >
Just tried that - rebooted my polycom and still half audio. Thanks, Jerry _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
