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

*CLI> module load
Unable to load module
Command 'module load ' failed.

3. for some reason module “chan_alsa” is not loaded automatically, after restart.

*CLI> module load
*CLI> module show like alsa
Module                         Description                              Use Count                   ALSA Console Channel Driver              0
1 modules loaded

and now you should have the console command.

; Open Sound System Console Driver Configuration File
; Automatically answer incoming calls on the console?  Choose yes if
; for example you want to use this as an intercom.
; Default context (is overridden with @context syntax)
; Default extension to call
; Default language
; 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.
; 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

; Default mute state (can also be toggled via CLI)

; 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.

;------------------------------ 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".

Файлы конфигурации Asterisk

  • asterisk/cf/alsa.conf.txt
  • Последние изменения: 2018/03/12