Asterisk config oss.conf

The oss.conf Asterisk configuration file allows configuration of OSS channels within Asterisk. OSS channels allow calls to be placed to/from OSS devices, using OSS audio input/output devices as telephony devices. This allows a headphone and microphone plugged into a sound card, for example, to be used as a softphone. Calls can be placed and digits can be dialed from the Asterisk CLI, using the "dial", "answer", and "hangup" CLI commands.

The oss.conf file may contain any of the following directives:

  • autoanswer = yes or no
Indicates whether or not calls to OSS channels should be automatically answered as soon as they are rung. This can be used to provide, for exmaple, intercom-like functionality over IP. If autoanswer is disabled, the "answer" command must be entered at the CLI command prompt to answer a call ringing an OSS channel. (For some reason, in the default configuration, the value of autoanswer is set to yes. If you value your privacy, you should disable it.)

  • context = <context name>
Indicates the context in which extensions dialed with the CLI "dial" command are looked up. Ex: "dial 100" will dial extension 100 in the context specified by this directive.

  • extension = <extension name or number>
Indicates the extension which will be dialed (in the context specified by the context= line) if the CLI "dial" command is issued with no extension argument. Ex: just "dial".

With 1.2.x versions of Asterisk, only one OSS channel can be created and only one call can be placed to/from the OSS channel at a given time. If the OSS channel is in use when an incoming call is routed to it, the destination will be treated as busy.

Similar to the OSS channel driver, the ALSA channel driver provides similar functionality, but for ALSA-compatible devices. See Asterisk config alsa.conf.

Only one of either the ALSA or OSS channel drivers may be loaded at a given time. So, if you plan to use an OSS channel, you must enable chan_oss and disable chan_alsa in modules.conf. Ex:

noload => chan_alsa.so
load => chan_oss.so


  • mixer = <mixer_shell_args>
Taken from asterisk 1.6.2.14 source, maybe present on all 1.6.X:
"chan_oss.c:1289"
  • store the mixer argument from the config file, filtering possibly
  • invalid or dangerous values (the string is used as argument for
  • system("mixer %s")


The oss.conf Asterisk configuration file allows configuration of OSS channels within Asterisk. OSS channels allow calls to be placed to/from OSS devices, using OSS audio input/output devices as telephony devices. This allows a headphone and microphone plugged into a sound card, for example, to be used as a softphone. Calls can be placed and digits can be dialed from the Asterisk CLI, using the "dial", "answer", and "hangup" CLI commands.

The oss.conf file may contain any of the following directives:

  • autoanswer = yes or no
Indicates whether or not calls to OSS channels should be automatically answered as soon as they are rung. This can be used to provide, for exmaple, intercom-like functionality over IP. If autoanswer is disabled, the "answer" command must be entered at the CLI command prompt to answer a call ringing an OSS channel. (For some reason, in the default configuration, the value of autoanswer is set to yes. If you value your privacy, you should disable it.)

  • context = <context name>
Indicates the context in which extensions dialed with the CLI "dial" command are looked up. Ex: "dial 100" will dial extension 100 in the context specified by this directive.

  • extension = <extension name or number>
Indicates the extension which will be dialed (in the context specified by the context= line) if the CLI "dial" command is issued with no extension argument. Ex: just "dial".

With 1.2.x versions of Asterisk, only one OSS channel can be created and only one call can be placed to/from the OSS channel at a given time. If the OSS channel is in use when an incoming call is routed to it, the destination will be treated as busy.

Similar to the OSS channel driver, the ALSA channel driver provides similar functionality, but for ALSA-compatible devices. See Asterisk config alsa.conf.

Only one of either the ALSA or OSS channel drivers may be loaded at a given time. So, if you plan to use an OSS channel, you must enable chan_oss and disable chan_alsa in modules.conf. Ex:

noload => chan_alsa.so
load => chan_oss.so


  • mixer = <mixer_shell_args>
Taken from asterisk 1.6.2.14 source, maybe present on all 1.6.X:
"chan_oss.c:1289"
  • store the mixer argument from the config file, filtering possibly
  • invalid or dangerous values (the string is used as argument for
  • system("mixer %s")


Created by: nicolas31, Last modification: Sun 03 of Jun, 2012 (21:00 UTC) by elbriga
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