mirror of https://github.com/asterisk/asterisk
ASTERISK-30298 Change-Id: I5c8afb781528afdf55d237e3bffa5e4a862ae8c7pull/30/head
Mike Bradeen
2 years ago
committed by
George Joseph
parent
6b8d3cb89a
commit
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;
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; Open Sound System Console Driver Configuration File
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;
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[general]
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;
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; Automatically answer incoming calls on the console? Choose yes if
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; for example you want to use this as an intercom.
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;
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autoanswer=yes
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;
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; Default context (is overridden with @context syntax)
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;
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context=local
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;
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; Default extension to call
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;
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extension=s
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;
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; Default language
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;
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;language=en
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;
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; Default Music on Hold class to use when this channel is placed on hold in
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; the case that the music class is not set on the channel with
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; Set(CHANNEL(musicclass)=whatever) in the dialplan and the peer channel
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; putting this one on hold did not suggest a class to use.
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;
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;mohinterpret=default
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;
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; Silence suppression can be enabled when sound is over a certain threshold.
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; The value for the threshold should probably be between 500 and 2000 or so,
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; but your mileage may vary. Use the echo test to evaluate the best setting.
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;silencesuppression = yes
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;silencethreshold = 1000
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;
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; To set which ALSA device to use, change this parameter
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;input_device=hw:0,0
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;output_device=hw:0,0
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;
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; Default mute state (can also be toggled via CLI)
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;mute=true
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;
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; If enabled, no audio capture device will be opened. This is useful on
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; systems where there will be no return audio path, such as overhead pagers.
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;noaudiocapture=true
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; ----------------------------- JITTER BUFFER CONFIGURATION --------------------------
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; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of an
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; ALSA channel. Defaults to "no". An enabled jitterbuffer will
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; be used only if the sending side can create and the receiving
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; side can not accept jitter. The ALSA channel can't accept jitter,
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; thus an enabled jitterbuffer on the receive ALSA side will always
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; be used if the sending side can create jitter.
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; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
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; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
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; resynchronized. Useful to improve the quality of the voice, with
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; big jumps in/broken timestamps, usually sent from exotic devices
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; and programs. Defaults to 1000.
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; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
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; channel. Two implementations are currently available - "fixed"
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; (with size always equals to jbmax-size) and "adaptive" (with
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; variable size, actually the new jb of IAX2). Defaults to fixed.
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; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set.
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; The option represents the number of milliseconds by which the new
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; jitter buffer will pad its size. the default is 40, so without
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; modification, the new jitter buffer will set its size to the jitter
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; value plus 40 milliseconds. increasing this value may help if your
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; network normally has low jitter, but occasionally has spikes.
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; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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; ----------------------------------------------------------------------------------
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@ -0,0 +1,6 @@
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Subject: chan_alsa
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Master-Only: True
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This module was deprecated in Asterisk 19
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and is now being removed in accordance with
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the Asterisk Module Deprecation policy.
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