asterisk:alsa.conf
1. make sure that “chan_oss” and “chan_alsa” are compiled while installing asterisk.
2. ensure that “alsa.conf” exists in /etc/asterisk/alsa.conf.
I’m simply using the alsa.conf created from the samples file.
In case, alsa.conf doesn’t exist, you’ll get the following error, while loading chan_alsa.so.
*CLI> module load chan_alsa.so Unable to load module chan_alsa.so Command 'module load chan_alsa.so ' failed.
3. for some reason module “chan_alsa” is not loaded automatically, after restart.
*CLI> module load chan_alsa.so Loaded chan_alsa.so
*CLI> module show like alsa Module Description Use Count chan_alsa.so ALSA Console Channel Driver 0 1 modules loaded
and now you should have the console command.
; ; 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 ; ; Default mute state (can also be toggled via CLI) ;mute=true ; ; If enabled, no audio capture device will be opened. This is useful on ; systems where there will be no return audio path, such as overhead pagers. ;noaudiocapture=true ;------------------------------ 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. ; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set. ; The option represents the number of milliseconds by which the new ; jitter buffer will pad its size. the default is 40, so without ; modification, the new jitter buffer will set its size to the jitter ; value plus 40 milliseconds. increasing this value may help if your ; network normally has low jitter, but occasionally has spikes. ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". ;-----------------------------------------------------------------------------------