From 503ebba846b30a4d9e493a7e8578f9c40cf89bf4 Mon Sep 17 00:00:00 2001 From: lmadsen Date: Wed, 21 Jan 2009 13:19:16 +0000 Subject: Creating tag for asterisk-1.4.23 (in the right location this time too!) git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.23@169668 f38db490-d61c-443f-a65b-d21fe96a405b --- 1.4.23-rc4/configs/alsa.conf.sample | 62 +++++++++++++++++++++++++++++++++++++ 1 file changed, 62 insertions(+) create mode 100644 1.4.23-rc4/configs/alsa.conf.sample (limited to '1.4.23-rc4/configs/alsa.conf.sample') diff --git a/1.4.23-rc4/configs/alsa.conf.sample b/1.4.23-rc4/configs/alsa.conf.sample new file mode 100644 index 000000000..f55030618 --- /dev/null +++ b/1.4.23-rc4/configs/alsa.conf.sample @@ -0,0 +1,62 @@ +; +; Open Sound System Console Driver Configuration File +; +[general] +; +; Automatically answer incoming calls on the console? Choose yes if +; for example you want to use this as an intercom. +; +autoanswer=yes +; +; Default context (is overridden with @context syntax) +; +context=local +; +; Default extension to call +; +extension=s +; +; Default language +; +;language=en +; +; Default Music on Hold class to use when this channel is placed on hold in +; the case that the music class is not set on the channel with +; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel +; putting this one on hold did not suggest a class to use. +; +;mohinterpret=default +; +; Silence suppression can be enabled when sound is over a certain threshold. +; The value for the threshold should probably be between 500 and 2000 or so, +; but your mileage may vary. Use the echo test to evaluate the best setting. +;silencesuppression = yes +;silencethreshold = 1000 +; +; To set which ALSA device to use, change this parameter +;input_device=hw:0,0 +;output_device=hw:0,0 + +;------------------------------ JITTER BUFFER CONFIGURATION -------------------------- +; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an + ; ALSA channel. Defaults to "no". An enabled jitterbuffer will + ; be used only if the sending side can create and the receiving + ; side can not accept jitter. The ALSA channel can't accept jitter, + ; thus an enabled jitterbuffer on the receive ALSA side will always + ; be used if the sending side can create jitter. + +; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. + +; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is + ; resynchronized. Useful to improve the quality of the voice, with + ; big jumps in/broken timestamps, usually sent from exotic devices + ; and programs. Defaults to 1000. + +; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP + ; channel. Two implementations are currently available - "fixed" + ; (with size always equals to jbmax-size) and "adaptive" (with + ; variable size, actually the new jb of IAX2). Defaults to fixed. + +; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". +;----------------------------------------------------------------------------------- + -- cgit v1.2.3