Hi, I cross compile pjsip for arm board. I have sgtl5000 audio codec as sound card. My project need to use the audio output channels independently. I want: - Play music to left channel (speaker) - Use right channel to sip call (pjsip). This is my asound.conf pcm.dshare { type dmix ipc_key 2048 slave { pcm "hw:0" rate 44100 } bindings { 0 0 1 1 } } pcm.leftx { type route slave { pcm "dshare" channels 2 } ttable.0.0 4 ttable.1.0 4 } pcm.rightx { type route slave { pcm "dshare" channels 2 } ttable.0.1 4 ttable.1.1 4 } pcm.mixin { type dsnoop ipc_key 2049 # must be unique for all dmix plugins!!!! #ipc_key_add_uid yes slave { pcm "hw:0" #channels 2 #period_size 1024 #buffer_size 4096 rate 44100 #periods 0 #period_time 0 } bindings { 0 0 1 1 } } I can play two audio file independently mpg123 -a leftx a.mp3 ---> left channel mpg123 -a rightx b.mp3 ---> right channel I can make a call with pjsip using rightx as playback-dev and mixin as capture-dev. But when i try play music (mpg123 -a leftx a.mp3) to the other speaker during a call, only listen music in left channel and pjsip-call don't play and don't capture nothing (like mute) Pjsip doesn't show any message or error Any idea? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20130820/890c6157/attachment-0001.html>