2009/4/26 Rask Ingemann Lambertsen <r...@sygehus.dk>: > On Sat, Apr 18, 2009 at 05:49:05PM +0200, Nicola Mfb wrote: > >> I will be happy to write an AMI gui but now I'm hold having problems >> with the alsa channel. Using the pcm default is not compatible with >> the default shipped /etc/asound.conf, so I just tried to use >> plughw:dnsoop and plughw:dmix, the result is that there freerunner >> does not ring on incoming call (and you cannot hear the other peer), >> while audio transmitting is perfect. Using plughw:0,0 for input/output >> works but I have stuttered audio (from freerunner to peer). > > Why are you not using hw:0,0?
Asterisk has fixed-hardcoded settings for alsa (8000hz, 1 channel etc), and they are incompatible using hw directly, plughw autoconvert sound streams but it uses very short buffer/period size so the stuttered audio (I guess). Nicola _______________________________________________ Openmoko community mailing list community@lists.openmoko.org http://lists.openmoko.org/mailman/listinfo/community