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Asterisk - The Open Source Telephony Project
18.5.0
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[general]
; The general section of this config
; is not currently used, but reserved
; for future use.
;
; --- Default Information ---
; The default_user and default_bridge sections are applied
; automatically to all ConfBridge instances invoked without
; a user, or bridge argument. No menu is applied by default.
;
; Note that while properties of the default_user or default_bridge
; profile can be overridden, if removed, they will be automatically
; added and made available to the dialplan upon module load.
;
; --- ConfBridge User Profile Options ---
[default_user]
type=user
;admin=yes ; Sets if the user is an admin or not. Off by default.
;send_events=no ; If events are enabled for this bridge and this option is
; set, users will receive events like join, leave, talking,
; etc. via text messages. For users accessing the bridge
; via chan_pjsip, this means in-dialog MESSAGE messages.
; This is most useful for WebRTC participants where the
; browser application can use the messages to alter the user
; interface.
;echo_events=yes ; If events are enabled for this user and this option is set,
; the user will receive events they trigger, talking, mute, etc.
; If not set, they will not receive their own events.
;marked=yes ; Sets if this is a marked user or not. Off by default.
;startmuted=yes; Sets if all users should start out muted. Off by default
;music_on_hold_when_empty=yes ; Sets whether MOH should be played when only
; one person is in the conference or when the
; the user is waiting on a marked user to enter
; the conference. Off by default.
;music_on_hold_class=default ; The MOH class to use for this user.
;quiet=yes ; When enabled enter/leave prompts and user intros are not played.
; There are some prompts, such as the prompt to enter a PIN number,
; that must be played regardless of what this option is set to.
; Off by default
;announce_user_count=yes ; Sets if the number of users should be announced to the
; caller. Off by default.
;announce_user_count_all=yes ; Sets if the number of users should be announced to
; all the other users in the conference when someone joins.
; This option can be either set to 'yes' or a number.
; When set to a number, the announcement will only occur
; once the user count is above the specified number.
;announce_only_user=yes ; Sets if the only user announcement should be played
; when a channel enters a empty conference. On by default.
;wait_marked=yes ; Sets if the user must wait for a marked user to enter before
; joining the conference. Off by default.
;end_marked=yes ; This option will kick every user with this option set in their
; user profile after the last Marked user exists the conference.
;dsp_drop_silence=yes ; This option drops what Asterisk detects as silence from
; entering into the bridge. Enabling this option will drastically
; improve performance and help remove the buildup of background
; noise from the conference. Highly recommended for large conferences
; due to its performance enhancements.
;dsp_talking_threshold=128 ; Average magnitude threshold to determine talking.
;
; The minimum average magnitude per sample in a frame for the
; DSP to consider talking/noise present. A value below this
; level is considered silence. This value affects several
; operations and should not be changed unless the impact on
; call quality is fully understood.
;
; What this value affects internally:
;
; 1. Audio is only mixed out of a user's incoming audio
; stream if talking is detected. If this value is set too
; high the user will hear himself talking.
;
; 2. When talk detection AMI events are enabled, this value
; determines when talking has begun which results in an
; AMI event to fire. If this value is set too low AMI
; events may be falsely triggered by variants in room
; noise.
;
; 3. The 'drop_silence' option depends on this value to
; determine when the user's audio should be mixed into the
; bridge after periods of silence. If this value is too
; high the user's speech will get discarded as they will
; be considered silent.
;
; Valid values are 1 through 2^15.
; By default this value is 160.
;dsp_silence_threshold=2000 ; The number of milliseconds of silence necessary to declare
; talking stopped.
;
; The time in milliseconds of sound falling below the
; 'dsp_talking_threshold' option when a user is considered to
; stop talking. This value affects several operations and
; should not be changed unless the impact on call quality is
; fully understood.
;
; What this value affects internally:
;
; 1. When talk detection AMI events are enabled, this value
; determines when the user has stopped talking after a
; period of talking. If this value is set too low AMI
; events indicating the user has stopped talking may get
; falsely sent out when the user briefly pauses during mid
; sentence.
;
; 2. The 'drop_silence' option depends on this value to
; determine when the user's audio should begin to be
; dropped from the conference bridge after the user stops
; talking. If this value is set too low the user's audio
; stream may sound choppy to the other participants. This
; is caused by the user transitioning constantly from
; silence to talking during mid sentence.
;
; The best way to approach this option is to set it slightly
; above the maximum amount of milliseconds of silence a user
; may generate during natural speech.
;
; Valid values are 1 through 2^31.
; By default this value is 2500ms.
;talk_detection_events=yes ; This option sets whether or not notifications of when a user
; begins and ends talking should be sent out as events over AMI.
; By default this option is off.
;denoise=yes ; Sets whether or not a denoise filter should be applied
; to the audio before mixing or not. Off by default. Requires
; func_speex to be built and installed. Do not confuse this option
; with drop_silence. Denoise is useful if there is a lot of background
; noise for a user as it attempts to remove the noise while preserving
; the speech. This option does NOT remove silence from being mixed into
; the conference and does come at the cost of a slight performance hit.
;jitterbuffer=yes ; Enabling this option places a jitterbuffer on the user's audio stream
; before audio mixing is performed. This is highly recommended but will
; add a slight delay to the audio. This option is using the JITTERBUFFER
; dialplan function's default adaptive jitterbuffer. For a more fine tuned
; jitterbuffer, disable this option and use the JITTERBUFFER dialplan function
; on the user before entering the ConfBridge application.
;pin=1234 ; Sets if this user must enter a PIN number before entering
; the conference. The PIN will be prompted for.
;announce_join_leave=yes ; When enabled, this option will prompt the user for a
; name when entering the conference. After the name is
; recorded, it will be played as the user enters and exists
; the conference. This option is off by default.
;announce_join_leave_review=yes ; When enabled, implies announce_join_leave, but the user
; will be prompted to review their recording before
; entering the conference. During this phase, the recording
; may be listened to, re-recorded, or accepted as is. This
; option is off by default.
;dtmf_passthrough=yes ; Sets whether or not DTMF should pass through the conference.
; This option is off by default.
;announcement=</path/to/file> ; Play a sound file to the user when they join the conference.
;timeout=3600 ; When set non-zero, this specifies the number of seconds that the participant
; may stay in the conference before being automatically ejected. When the user
; is ejected from the conference, the user's channel will have the CONFBRIDGE_RESULT
; variable set to "TIMEOUT". A value of 0 indicates that there is no timeout.
; Default: 0
;text_messaging=yes ; When set to yes text messages will be sent to this user. Text messages
; may occur as a result of events or can be received from other participants.
; When set to no text messages will not be sent to this user.
;answer_channel=yes ; Sets if the channel should be answered if it hasn't been already.
; On by default.
; --- ConfBridge Bridge Profile Options ---
[default_bridge]
type=bridge
;max_members=50 ; This option limits the number of participants for a single
; conference to a specific number. By default conferences
; have no participant limit. After the limit is reached, the
; conference will be locked until someone leaves. Note however
; that an Admin user will always be alowed to join the conference
; regardless if this limit is reached or not.
;record_conference=yes ; Records the conference call starting when the first user
; enters the room, and ending when the last user exits the room.
; The default recorded filename is
; 'confbridge-<name of conference bridge>-<start time>.wav
; and the default format is 8khz slinear. This file will be
; located in the configured monitoring directory in asterisk.conf.
;record_file=</path/to/file> ; When record_conference is set to yes, the specific name of the
; record file can be set using this option. Note that since multiple
; conferences may use the same bridge profile, this may cause issues
; depending on the configuration. It is recommended to only use this
; option dynamically with the CONFBRIDGE() dialplan function. This
; allows the record name to be specified and a unique name to be chosen.
; By default, the record_file is stored in Asterisk's spool/monitor directory
; with a unique filename starting with the 'confbridge' prefix.
;record_file_append=yes ; Append record file when starting/stopping on same conference recording.
;record_file_timestamp=yes ; Append the start time to the record file name.
;record_options= ; Pass additional options to MixMonitor.
;record_command=</path/to/command> ; Command to execute when recording finishes.
;internal_sample_rate=auto ; Sets the internal native sample rate the
; conference is mixed at. This is set to automatically
; adjust the sample rate to the best quality by default.
; Other values can be anything from 8000-192000. If a
; sample rate is set that Asterisk does not support, the
; closest sample rate Asterisk does support to the one requested
; will be used.
;maximum_sample_rate=none ; Sets the maximum sample rate the conference
; is mixed at. This is set to no maximum by default.
; Values can be anything from 8000-192000.
;mixing_interval=40 ; Sets the internal mixing interval in milliseconds for the bridge. This
; number reflects how tight or loose the mixing will be for the conference.
; In order to improve performance a larger mixing interval such as 40ms may
; be chosen. Using a larger mixing interval comes at the cost of introducing
; larger amounts of delay into the bridge. Valid values here are 10, 20, 40,
; or 80. By default 20ms is used.
;video_mode = follow_talker; Sets how confbridge handles video distribution to the conference participants.
; Note that participants wanting to view and be the source of a video feed
; _MUST_ be sharing the same video codec. Also, using video in conjunction with
; with the jitterbuffer currently results in the audio being slightly out of sync
; with the video. This is a result of the jitterbuffer only working on the audio
; stream. It is recommended to disable the jitterbuffer when video is used.
;
; --- MODES ---
; none: No video sources are set by default in the conference. It is still
; possible for a user to be set as a video source via AMI or DTMF action
; at any time.
;
; follow_talker: The video feed will follow whoever is talking and providing video.
;
; last_marked: The last marked user to join the conference with video capabilities
; will be the single source of video distributed to all participants.
; If multiple marked users are capable of video, the last one to join
; is always the source, when that user leaves it goes to the one who
; joined before them.
;
; first_marked: The first marked user to join the conference with video capabilities
; is the single source of video distribution among all participants. If
; that user leaves, the marked user to join after them becomes the source.
;
; sfu: Selective Forwarding Unit - Sets multi-stream operation
; for a multi-party video conference.
;language=en ; Set the language used for announcements to the conference.
; Default is en (English).
;regcontext=conferences ; The name of the context into which to register conference names as extensions.
;video_update_discard=2000 ; Amount of time (in milliseconds) to discard video update requests after sending a video
; update request. Default is 2000. A video update request is a request for a full video
; intra-frame. Clients can request this if they require a full frame in order to decode
; the video stream. Since a full frame can be large limiting how often they occur can
; reduce bandwidth usage at the cost of increasing how long it may take a newly joined
; channel to receive the video stream.
;remb_send_interval=1000 ; Interval (in milliseconds) at which a combined REMB frame will be sent to sources of video.
; A REMB frame contains receiver estimated maximum bitrate information. By creating a combined
; frame and sending it to the sources of video the sender can be influenced on what bitrate
; they choose allowing a better experience for the receivers. This defaults to 0, or disabled.
;remb_behavior=average ; How the combined REMB report for an SFU video bridge is constructed. If set to "average" then
; the estimated maximum bitrate of each receiver is used to construct an average bitrate. If
; set to "lowest" the lowest maximum bitrate is forwarded to the sender. If set to "highest"
; the highest maximum bitrate is forwarded to the sender. If set to "average_all" a single average
; is generated from every receiver and the same value is sent to every sender. If set to
; "lowest_all" the lowest maximum bitrate of all receivers is sent to every sender. If set to
; "highest_all" the highest maximum bitrate of all receivers is sent to every sender.
; When set to "force", the value set in remb_estimated_bitrate is sent to every sender.
; This defaults to "average".
;remb_estimated_bitrate=0 ; When remb_behavior is set to 'force', this options sets the estimated bitrate
; (in bits per second) sent to each participant in REMB reports.
;enable_events=no ; If enabled, recipients who joined the bridge via a channel driver
; that supports Enhanced Messaging (currently only chan_pjsip) will
; receive in-dialog messages containing a JSON body describing the
; event. The Content-Type header will be
; "text/x-ast-confbridge-event".
; This feature must also be enabled in user profiles.
; All sounds in the conference are customizable using the bridge profile options below.
; Simply state the option followed by the filename or full path of the filename after
; the option. Example: sound_had_joined=conf-hasjoin This will play the conf-hasjoin
; sound file found in the sounds directory when announcing someone's name is joining the
; conference.
;sound_join ; The sound played to everyone when someone enters the conference.
;sound_leave ; The sound played to everyone when someone leaves the conference.
;sound_has_joined ; The sound played before announcing someone's name has
; joined the conference. This is used for user intros.
; Example "_____ has joined the conference"
;sound_has_left ; The sound played when announcing someone's name has
; left the conference. This is used for user intros.
; Example "_____ has left the conference"
;sound_kicked ; The sound played to a user who has been kicked from the conference.
;sound_muted ; The sound played when the mute option is toggled on using DTMF menu.
;sound_unmuted ; The sound played when the mute option is toggled off using DTMF menu.
;sound_only_person ; The sound played when the user is the only person in the conference.
;sound_only_one ; The sound played to a user when there is only one other
; person is in the conference.
;sound_there_are ; The sound played when announcing how many users there
; are in a conference.
;sound_other_in_party; ; This file is used in conjunction with 'sound_there_are"
; when announcing how many users there are in the conference.
; 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
1.8.13