462 lines
30 KiB
Plaintext
462 lines
30 KiB
Plaintext
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[general]
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; The general section of this config
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; is not currently used, but reserved
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; for future use.
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;
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; --- Default Information ---
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; The default_user and default_bridge sections are applied
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; automatically to all ConfBridge instances invoked without
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; a user, or bridge argument. No menu is applied by default.
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;
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; Note that while properties of the default_user or default_bridge
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; profile can be overridden, if removed, they will be automatically
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; added and made available to the dialplan upon module load.
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;
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; --- ConfBridge User Profile Options ---
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[default_user]
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type=user
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;admin=yes ; Sets if the user is an admin or not. Off by default.
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;send_events=no ; If events are enabled for this bridge and this option is
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; set, users will receive events like join, leave, talking,
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; etc. via text messages. For users accessing the bridge
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; via chan_pjsip, this means in-dialog MESSAGE messages.
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; This is most useful for WebRTC participants where the
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; browser application can use the messages to alter the user
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; interface.
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;echo_events=yes ; If events are enabled for this user and this option is set,
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; the user will receive events they trigger, talking, mute, etc.
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; If not set, they will not receive their own events.
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;marked=yes ; Sets if this is a marked user or not. Off by default.
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;startmuted=yes; Sets if all users should start out muted. Off by default
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;music_on_hold_when_empty=yes ; Sets whether MOH should be played when only
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; one person is in the conference or when the
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; the user is waiting on a marked user to enter
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; the conference. Off by default.
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;music_on_hold_class=default ; The MOH class to use for this user.
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;quiet=yes ; When enabled enter/leave prompts and user intros are not played.
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; There are some prompts, such as the prompt to enter a PIN number,
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; that must be played regardless of what this option is set to.
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; Off by default
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;hear_own_join_sound=yes ; Sets if a user joining the conference should hear the sound_join
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; audio sound when they enter the conference. If set to 'no' the
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; user will not hear the sound_join audio but the other participants
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; in the conference will still hear the audio. If set to 'yes'
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; everyone hears the sound_join audio when this user enters the conference.
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; On by default
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;announce_user_count=yes ; Sets if the number of users should be announced to the
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; caller. Off by default.
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;announce_user_count_all=yes ; Sets if the number of users should be announced to
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; all the other users in the conference when someone joins.
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; This option can be either set to 'yes' or a number.
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; When set to a number, the announcement will only occur
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; once the user count is above the specified number.
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;announce_only_user=yes ; Sets if the only user announcement should be played
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; when a channel enters a empty conference. On by default.
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;wait_marked=yes ; Sets if the user must wait for a marked user to enter before
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; joining the conference. Off by default.
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;end_marked=yes ; This option will kick every non-marked user with this option set in their
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; user profile after the last marked user exits the conference.
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;end_marked_any=no ; This option will kick every user with this option set in
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; their user profile after any marked user exits the conference.
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; Additionally, note that unlike end_marked, this includes marked users.
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;dsp_drop_silence=yes ; This option drops what Asterisk detects as silence from
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; entering into the bridge. Enabling this option will drastically
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; improve performance and help remove the buildup of background
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; noise from the conference. Highly recommended for large conferences
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; due to its performance enhancements.
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;dsp_talking_threshold=128 ; Average magnitude threshold to determine talking.
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;
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; The minimum average magnitude per sample in a frame for the
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; DSP to consider talking/noise present. A value below this
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; level is considered silence. This value affects several
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; operations and should not be changed unless the impact on
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; call quality is fully understood.
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;
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; What this value affects internally:
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;
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; 1. Audio is only mixed out of a user's incoming audio
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; stream if talking is detected. If this value is set too
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; high the user will hear himself talking.
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;
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; 2. When talk detection AMI events are enabled, this value
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; determines when talking has begun which results in an
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; AMI event to fire. If this value is set too low AMI
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; events may be falsely triggered by variants in room
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; noise.
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;
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; 3. The 'drop_silence' option depends on this value to
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; determine when the user's audio should be mixed into the
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; bridge after periods of silence. If this value is too
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; high the user's speech will get discarded as they will
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; be considered silent.
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;
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; Valid values are 1 through 2^15.
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; By default this value is 160.
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;dsp_silence_threshold=2000 ; The number of milliseconds of silence necessary to declare
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; talking stopped.
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;
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; The time in milliseconds of sound falling below the
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; 'dsp_talking_threshold' option when a user is considered to
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; stop talking. This value affects several operations and
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; should not be changed unless the impact on call quality is
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; fully understood.
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;
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; What this value affects internally:
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;
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; 1. When talk detection AMI events are enabled, this value
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; determines when the user has stopped talking after a
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; period of talking. If this value is set too low AMI
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; events indicating the user has stopped talking may get
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; falsely sent out when the user briefly pauses during mid
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; sentence.
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;
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; 2. The 'drop_silence' option depends on this value to
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; determine when the user's audio should begin to be
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; dropped from the conference bridge after the user stops
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; talking. If this value is set too low the user's audio
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; stream may sound choppy to the other participants. This
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; is caused by the user transitioning constantly from
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; silence to talking during mid sentence.
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;
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; The best way to approach this option is to set it slightly
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; above the maximum amount of milliseconds of silence a user
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; may generate during natural speech.
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;
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; Valid values are 1 through 2^31.
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; By default this value is 2500ms.
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;talk_detection_events=yes ; This option sets whether or not notifications of when a user
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; begins and ends talking should be sent out as events over AMI.
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; By default this option is off.
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;denoise=yes ; Sets whether or not a denoise filter should be applied
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; to the audio before mixing or not. Off by default. Requires
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; func_speex to be built and installed. Do not confuse this option
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; with drop_silence. Denoise is useful if there is a lot of background
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; noise for a user as it attempts to remove the noise while preserving
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; the speech. This option does NOT remove silence from being mixed into
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; the conference and does come at the cost of a slight performance hit.
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;jitterbuffer=yes ; Enabling this option places a jitterbuffer on the user's audio stream
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; before audio mixing is performed. This is highly recommended but will
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; add a slight delay to the audio. This option is using the JITTERBUFFER
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; dialplan function's default adaptive jitterbuffer. For a more fine tuned
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; jitterbuffer, disable this option and use the JITTERBUFFER dialplan function
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; on the user before entering the ConfBridge application.
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;pin=1234 ; Sets if this user must enter a PIN number before entering
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; the conference. The PIN will be prompted for.
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;announce_join_leave=yes ; When enabled, this option will prompt the user for a
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; name when entering the conference. After the name is
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; recorded, it will be played as the user enters and exists
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; the conference. This option is off by default.
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;announce_join_leave_review=yes ; When enabled, implies announce_join_leave, but the user
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; will be prompted to review their recording before
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; entering the conference. During this phase, the recording
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; may be listened to, re-recorded, or accepted as is. This
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; option is off by default.
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;dtmf_passthrough=yes ; Sets whether or not DTMF should pass through the conference.
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; This option is off by default.
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;announcement=</path/to/file> ; Play a sound file to the user when they join the conference.
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;timeout=3600 ; When set non-zero, this specifies the number of seconds that the participant
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; may stay in the conference before being automatically ejected. When the user
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; is ejected from the conference, the user's channel will have the CONFBRIDGE_RESULT
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; variable set to "TIMEOUT". A value of 0 indicates that there is no timeout.
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; Default: 0
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;text_messaging=yes ; When set to yes text messages will be sent to this user. Text messages
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; may occur as a result of events or can be received from other participants.
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; When set to no text messages will not be sent to this user.
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;answer_channel=yes ; Sets if the channel should be answered if it hasn't been already.
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; On by default.
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; --- ConfBridge Bridge Profile Options ---
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[default_bridge]
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type=bridge
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;max_members=50 ; This option limits the number of participants for a single
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; conference to a specific number. By default conferences
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; have no participant limit. After the limit is reached, the
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; conference will be locked until someone leaves. Note however
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; that an Admin user will always be alowed to join the conference
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; regardless if this limit is reached or not.
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;record_conference=yes ; Records the conference call starting when the first user
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; enters the room, and ending when the last user exits the room.
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; The default recorded filename is
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; 'confbridge-<name of conference bridge>-<start time>.wav
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; and the default format is 8khz slinear. This file will be
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; located in the configured monitoring directory in asterisk.conf.
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;record_file=</path/to/file> ; When record_conference is set to yes, the specific name of the
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; record file can be set using this option. Note that since multiple
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; conferences may use the same bridge profile, this may cause issues
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; depending on the configuration. It is recommended to only use this
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; option dynamically with the CONFBRIDGE() dialplan function. This
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; allows the record name to be specified and a unique name to be chosen.
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; By default, the record_file is stored in Asterisk's spool/monitor directory
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; with a unique filename starting with the 'confbridge' prefix.
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;record_file_append=yes ; Append record file when starting/stopping on same conference recording.
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;record_file_timestamp=yes ; Append the start time to the record file name.
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;record_options= ; Pass additional options to MixMonitor.
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;record_command=</path/to/command> ; Command to execute when recording finishes.
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;internal_sample_rate=auto ; Sets the internal native sample rate the
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; conference is mixed at. This is set to automatically
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; adjust the sample rate to the best quality by default.
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; Other values can be anything from 8000-192000. If a
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; sample rate is set that Asterisk does not support, the
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; closest sample rate Asterisk does support to the one requested
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; will be used.
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;maximum_sample_rate=none ; Sets the maximum sample rate the conference
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; is mixed at. This is set to no maximum by default.
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; Values can be anything from 8000-192000.
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;mixing_interval=40 ; Sets the internal mixing interval in milliseconds for the bridge. This
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; number reflects how tight or loose the mixing will be for the conference.
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; In order to improve performance a larger mixing interval such as 40ms may
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; be chosen. Using a larger mixing interval comes at the cost of introducing
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; larger amounts of delay into the bridge. Valid values here are 10, 20, 40,
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; or 80. By default 20ms is used.
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;video_mode = follow_talker; Sets how confbridge handles video distribution to the conference participants.
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; Note that participants wanting to view and be the source of a video feed
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; _MUST_ be sharing the same video codec. Also, using video in conjunction with
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; with the jitterbuffer currently results in the audio being slightly out of sync
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; with the video. This is a result of the jitterbuffer only working on the audio
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; stream. It is recommended to disable the jitterbuffer when video is used.
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;
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; --- MODES ---
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; none: No video sources are set by default in the conference. It is still
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; possible for a user to be set as a video source via AMI or DTMF action
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; at any time.
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;
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; follow_talker: The video feed will follow whoever is talking and providing video.
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;
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; last_marked: The last marked user to join the conference with video capabilities
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; will be the single source of video distributed to all participants.
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; If multiple marked users are capable of video, the last one to join
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; is always the source, when that user leaves it goes to the one who
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; joined before them.
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;
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; first_marked: The first marked user to join the conference with video capabilities
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; is the single source of video distribution among all participants. If
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; that user leaves, the marked user to join after them becomes the source.
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;
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; sfu: Selective Forwarding Unit - Sets multi-stream operation
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; for a multi-party video conference.
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;language=en ; Set the language used for announcements to the conference.
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; Default is en (English).
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;regcontext=conferences ; The name of the context into which to register conference names as extensions.
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;video_update_discard=2000 ; Amount of time (in milliseconds) to discard video update requests after sending a video
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; update request. Default is 2000. A video update request is a request for a full video
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; intra-frame. Clients can request this if they require a full frame in order to decode
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; the video stream. Since a full frame can be large limiting how often they occur can
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; reduce bandwidth usage at the cost of increasing how long it may take a newly joined
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; channel to receive the video stream.
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;remb_send_interval=1000 ; Interval (in milliseconds) at which a combined REMB frame will be sent to sources of video.
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; A REMB frame contains receiver estimated maximum bitrate information. By creating a combined
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; frame and sending it to the sources of video the sender can be influenced on what bitrate
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; they choose allowing a better experience for the receivers. This defaults to 0, or disabled.
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;remb_behavior=average ; How the combined REMB report for an SFU video bridge is constructed. If set to "average" then
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; the estimated maximum bitrate of each receiver is used to construct an average bitrate. If
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; set to "lowest" the lowest maximum bitrate is forwarded to the sender. If set to "highest"
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; the highest maximum bitrate is forwarded to the sender. If set to "average_all" a single average
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; is generated from every receiver and the same value is sent to every sender. If set to
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; "lowest_all" the lowest maximum bitrate of all receivers is sent to every sender. If set to
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; "highest_all" the highest maximum bitrate of all receivers is sent to every sender.
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; When set to "force", the value set in remb_estimated_bitrate is sent to every sender.
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; This defaults to "average".
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;remb_estimated_bitrate=0 ; When remb_behavior is set to 'force', this options sets the estimated bitrate
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; (in bits per second) sent to each participant in REMB reports.
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;enable_events=no ; If enabled, recipients who joined the bridge via a channel driver
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; that supports Enhanced Messaging (currently only chan_pjsip) will
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; receive in-dialog messages containing a JSON body describing the
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; event. The Content-Type header will be
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; "text/x-ast-confbridge-event".
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; This feature must also be enabled in user profiles.
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; All sounds in the conference are customizable using the bridge profile options below.
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; Simply state the option followed by the filename or full path of the filename after
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; the option. Example: sound_had_joined=conf-hasjoin This will play the conf-hasjoin
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; sound file found in the sounds directory when announcing someone's name is joining the
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; conference.
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;sound_join ; The sound played to everyone when someone enters the conference.
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;sound_leave ; The sound played to everyone when someone leaves the conference.
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;sound_has_joined ; The sound played before announcing someone's name has
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; joined the conference. This is used for user intros.
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; Example "_____ has joined the conference"
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;sound_has_left ; The sound played when announcing someone's name has
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; left the conference. This is used for user intros.
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; Example "_____ has left the conference"
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;sound_kicked ; The sound played to a user who has been kicked from the conference.
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;sound_muted ; The sound played when the mute option is toggled on using DTMF menu.
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;sound_unmuted ; The sound played when the mute option is toggled off using DTMF menu.
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;sound_only_person ; The sound played when the user is the only person in the conference.
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;sound_only_one ; The sound played to a user when there is only one other
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; person is in the conference.
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;sound_there_are ; The sound played when announcing how many users there
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; are in a conference.
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;sound_other_in_party; ; This file is used in conjunction with 'sound_there_are"
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; when announcing how many users there are in the conference.
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|
|
; The sounds are stringed together like this.
|
||
|
|
; "sound_there_are" <number of participants> "sound_other_in_party"
|
||
|
|
;sound_place_into_conference ; The sound played when someone is placed into the conference
|
||
|
|
; after waiting for a marked user. This sound is now deprecated
|
||
|
|
; since it was only ever used improperly and correcting that bug
|
||
|
|
; made it completely unused.
|
||
|
|
;sound_wait_for_leader ; The sound played when a user is placed into a conference that
|
||
|
|
; can not start until a marked user enters.
|
||
|
|
;sound_leader_has_left ; The sound played when the last marked user leaves the conference.
|
||
|
|
;sound_get_pin ; The sound played when prompting for a conference pin number.
|
||
|
|
;sound_invalid_pin ; The sound played when an invalid pin is entered too many times.
|
||
|
|
;sound_locked ; The sound played to a user trying to join a locked conference.
|
||
|
|
;sound_locked_now ; The sound played to an admin after toggling the conference to locked mode.
|
||
|
|
;sound_unlocked_now; The sound played to an admin after toggling the conference to unlocked mode.
|
||
|
|
;sound_error_menu ; The sound played when an invalid menu option is entered.
|
||
|
|
;sound_begin ; The sound played to the conference when the first marked user enters the conference.
|
||
|
|
;sound_binaural_on ; The sound played when binaural audio is turned on
|
||
|
|
;sound_binaural_off ; The sound played when binaural audio is turned off
|
||
|
|
|
||
|
|
; --- ConfBridge Menu Options ---
|
||
|
|
; The ConfBridge application also has the ability to
|
||
|
|
; apply custom DTMF menus to each channel using the
|
||
|
|
; application. Like the User and Bridge profiles
|
||
|
|
; a menu is passed in to ConfBridge as an argument in
|
||
|
|
; the dialplan.
|
||
|
|
;
|
||
|
|
; Below is a list of menu actions that can be assigned
|
||
|
|
; to a DTMF sequence.
|
||
|
|
;
|
||
|
|
; To have the first DTMF digit in a sequence be the '#' character, you need to
|
||
|
|
; escape it. If it is not escaped then normal config file processing will
|
||
|
|
; think it is a directive like #include. For example:
|
||
|
|
; \#1=toggle_mute ; Pressing #1 will toggle the mute setting.
|
||
|
|
;
|
||
|
|
; A single DTMF sequence can have multiple actions associated with it. This is
|
||
|
|
; accomplished by stringing the actions together and using a ',' as the delimiter.
|
||
|
|
; Example: Both listening and talking volume is reset when '5' is pressed.
|
||
|
|
; 5=reset_talking_volume, reset_listening_volume
|
||
|
|
;
|
||
|
|
; playback(<name of audio file>&<name of audio file>)
|
||
|
|
; Playback will play back an audio file to a channel
|
||
|
|
; and then immediately return to the conference.
|
||
|
|
; This file can not be interupted by DTMF.
|
||
|
|
; Mutliple files can be chained together using the
|
||
|
|
; '&' character.
|
||
|
|
; playback_and_continue(<name of playback prompt>&<name of playback prompt>)
|
||
|
|
; playback_and_continue will
|
||
|
|
; play back a prompt while continuing to
|
||
|
|
; collect the dtmf sequence. This is useful
|
||
|
|
; when using a menu prompt that describes all
|
||
|
|
; the menu options. Note however that any DTMF
|
||
|
|
; during this action will terminate the prompts
|
||
|
|
; playback. Prompt files can be chained together
|
||
|
|
; using the '&' character as a delimiter.
|
||
|
|
; toggle_mute ; Toggle turning on and off mute. Mute will make the user silent
|
||
|
|
; to everyone else, but the user will still be able to listen in.
|
||
|
|
; toggle_binaural ; Toggle on or off binaural audio processing.
|
||
|
|
|
||
|
|
; no_op ; This action does nothing (No Operation). Its only real purpose exists for
|
||
|
|
; being able to reserve a sequence in the config as a menu exit sequence.
|
||
|
|
; decrease_listening_volume ; Decreases the channel's listening volume.
|
||
|
|
; increase_listening_volume ; Increases the channel's listening volume.
|
||
|
|
; reset_listening_volume ; Reset channel's listening volume to default level.
|
||
|
|
|
||
|
|
; decrease_talking_volume ; Decreases the channel's talking volume.
|
||
|
|
; increase_talking_volume ; Icreases the channel's talking volume.
|
||
|
|
; reset_talking_volume ; Reset channel's talking volume to default level.
|
||
|
|
;
|
||
|
|
; dialplan_exec(context,exten,priority) ; The dialplan_exec action allows a user
|
||
|
|
; to escape from the conference and execute
|
||
|
|
; commands in the dialplan. Once the dialplan
|
||
|
|
; exits the user will be put back into the
|
||
|
|
; conference. The possibilities are endless!
|
||
|
|
; leave_conference ; This action allows a user to exit the conference and continue
|
||
|
|
; execution in the dialplan.
|
||
|
|
;
|
||
|
|
; admin_kick_last ; This action allows an Admin to kick the last participant from the
|
||
|
|
; conference. This action will only work for admins which allows
|
||
|
|
; a single menu to be used for both users and admins.
|
||
|
|
;
|
||
|
|
; admin_toggle_conference_lock ; This action allows an Admin to toggle locking and
|
||
|
|
; unlocking the conference. Non admins can not use
|
||
|
|
; this action even if it is in their menu.
|
||
|
|
|
||
|
|
; set_as_single_video_src ; This action allows any user to set themselves as the
|
||
|
|
; single video source distributed to all participants.
|
||
|
|
; This will make the video feed stick to them regardless
|
||
|
|
; of what the video_mode is set to.
|
||
|
|
|
||
|
|
; release_as_single_video_src ; This action allows a user to release themselves as
|
||
|
|
; the video source. If video_mode is not set to "none"
|
||
|
|
; this action will result in the conference returning to
|
||
|
|
; whatever video mode the bridge profile is using.
|
||
|
|
;
|
||
|
|
; Note that this action will have no effect if the user
|
||
|
|
; is not currently the video source. Also, the user is
|
||
|
|
; not guaranteed by using this action that they will not
|
||
|
|
; become the video source again. The bridge will return
|
||
|
|
; to whatever operation the video_mode option is set to
|
||
|
|
; upon release of the video src.
|
||
|
|
|
||
|
|
; admin_toggle_mute_participants ; This action allows an administrator to toggle the mute
|
||
|
|
; state for all non-admins within a conference.
|
||
|
|
; Subsequent non-admins joining a muted conference will
|
||
|
|
; start muted. All admin users are unaffected by this
|
||
|
|
; option. Note that all users, regardless of their admin
|
||
|
|
; status, are notified that the conference is muted when
|
||
|
|
; the state is toggled.
|
||
|
|
|
||
|
|
; participant_count ; This action plays back the number of participants currently
|
||
|
|
; in a conference
|
||
|
|
|
||
|
|
[sample_user_menu]
|
||
|
|
type=menu
|
||
|
|
*=playback_and_continue(conf-usermenu)
|
||
|
|
*1=toggle_mute
|
||
|
|
1=toggle_mute
|
||
|
|
*4=decrease_listening_volume
|
||
|
|
4=decrease_listening_volume
|
||
|
|
*6=increase_listening_volume
|
||
|
|
6=increase_listening_volume
|
||
|
|
*7=decrease_talking_volume
|
||
|
|
7=decrease_talking_volume
|
||
|
|
*8=leave_conference
|
||
|
|
8=leave_conference
|
||
|
|
*9=increase_talking_volume
|
||
|
|
9=increase_talking_volume
|
||
|
|
|
||
|
|
[sample_admin_menu]
|
||
|
|
type=menu
|
||
|
|
*=playback_and_continue(conf-adminmenu)
|
||
|
|
*1=toggle_mute
|
||
|
|
1=toggle_mute
|
||
|
|
*2=admin_toggle_conference_lock ; only applied to admin users
|
||
|
|
2=admin_toggle_conference_lock ; only applied to admin users
|
||
|
|
*3=admin_kick_last ; only applied to admin users
|
||
|
|
3=admin_kick_last ; only applied to admin users
|
||
|
|
*4=decrease_listening_volume
|
||
|
|
4=decrease_listening_volume
|
||
|
|
*6=increase_listening_volume
|
||
|
|
6=increase_listening_volume
|
||
|
|
*7=decrease_talking_volume
|
||
|
|
7=decrease_talking_volume
|
||
|
|
*8=no_op
|
||
|
|
8=no_op
|
||
|
|
*9=increase_talking_volume
|
||
|
|
9=increase_talking_volume
|