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A Bridge Profile provides the following configuration options:

Option

Values

Description

Notes

type

bridge

Set this to bridge to configure a bridge profile

 

max_members

integer; e.g. 50

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. Admin-level users are exempt from this limit and will still be able to join otherwise-locked, because of limit, conferences.

 

record_conference

yes/no

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 signed linear. By default, this option is disabled. This file will be located in the configured monitoring directory as set in asterisk.conf

 

record_file

path, e.g. /tmp/myfiles

When record_conference is set to yes, the specific name of the recorded file can be set using this option. Note that since multiple conferences may use the same Bridge profile, this can cause issues, depending on the configuration. It is recommended to only use this option dynamically with the CONFBRIDGE() dialplan function. This allows the recorded name to be specified and a unique name to be chosen. By default, the recorded file is stored in Asterisk's spool/monitory directory, with a unique filename starting with the 'confbridge' prefix.

 

internal_sample_rate

auto, 8000, 12000, 16000, 24000, 32000, 44100, 48000, 96000, 192000

Sets the internal native sample rate at which to mix the conference. The "auto" option allows Asterisk to adjust the sample rate to the best quality / performance based on the participant makeup. Numbered values lock the rate to the specified numerical rate. If a defined number does not match an internal sampling rate supported by Asterisk, the nearest sampling rate will be used instead.

 

mixing_interval

10, 20, 40, 80

Sets, in milliseconds, the internal mixing interval. By default, the mixing interval of a bridge is 20ms. This setting reflects how "tight" or "loose" the mixing will be for the conference. Lower intervals provide a "tighter" sound with less delay in the bridge and consume more system resources. Higher intervals provide a "looser" sound with more delay in the bridge and consume less resources

 

video_mode

none, follow_talker, last_marked, first_marked

Configured video (as opposed to audio) distribution method for conference participants. Participants must use the same video codec. Confbridge does not provide MCU functionality. It does not transcode, scale, transrate, or otherwise manipulate the video. Options are "none," where no video source is set by default and a video source may be later set via AMI or DTMF actions; "follow_talker," where video distrubtion follows whomever is talking and providing video; "last_marked," where the last marked user with video capabilities to join the conference will be the single video source distributed to all other participants - when the current video source leaves, the marked user previous to the last-joined will be used as the video source; and "first-marked," where the first marked user with video capabilities to join the conference will be the single video source distributed to all other participants - when the current video source leaves, the marked user that joined next will be used as the video source. Use of video in conjunction with the jitterbuffer results in the audio being slightly out of sync with the video - because the jitterbuffer only operates on the audio stream, not the video stream. Jitterbuffer should be disabled when video is used.

 

sound_join

filename

The sound played to the bridge when a user joins, typically some kind of beep sound

 

sound_leave

filename

The sound played to the bridge when a user leaves, also typically some kind of beep sound

 

sound_has_joined

filename

The sound played as a user intro, e.g. "xxxx has joined the conference."

 

sound_has_left

filename

The sound played as a user parts the conference, e.g. "xxxx has left the conference."

 

sound_kicked

filename

The sound played to a user who has been kicked from the conference.

 

sound_muted

filename

The sound played to a user when the mute option is toggled on.

 

sound_unmuted

filename

The sound played to a user when the mute option is toggled off.

 

sound_only_person

filename

The sound played when a user is the only person in the conference.

 

sound_only_one

filename

The sound played to a user when there is only one other person in the conference.

 

sound_there_are

filename

The sound played when announcing how many users there are in a conference.

 

sound_other_in_party

filename

Used in conjunction with the sound_there_are option, used like "sound_there_are" <number of participants> "sound_other_in_party"

 

sound_place_into_conference

filename

The sound played when someone is placed into a conference, after waiting for a marked user.

 

sound_wait_for_leader

filename

The sound played when a user is placed into a conference that cannot start until a marked user enters.

 

sound_leader_has_left

filename

The sound played when the last marked user leaves the conference.

 

sound_get_pin

filename

The sound played when prompting for a conference PIN

 

sound_invalid_pin

filename

The sound played when an invalid PIN is entered too many (3) times

 

sound_locked

filename

The sound played to a user trying to join a locked conference.

 

sound_locked_now

filename

The sound played to an Admin-level user after toggling the conference to locked mode.

 

sound_unlocked_now

filename

The sound played to an Admin-level user after toggling the conference to unlocked mode.

 

sound_error_menu

filename

The sound played when an invalid menu option is entered.

 

sound_participants_muted

filename

The sound played when all non-admin participants are muted.

New in Asterisk 11

sound_participants_unmuted

filename

The sound played when all non-admin participants are unmuted

New in Asterisk 11

Example
In this example, a Bridge Profile called "fancybridge" will be created. It will be configured to allow up to 20 callers, and will be set to mix at 10ms (tight mixing) at an automatic sampling rate. Additionally, it will be recorded.

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A User Profile provides the following configuration options:

Option

Values

Description

Notes

type

user

Set this to user to configure a user profile

 

admin

yes/no

Sets if the user is an Admin or not. By default, no.

 

marked

yes/no

Sets if the user is Marked or not. By default, no.

 

startmuted

yes/no

sets if the user should start out muted. By default, no.

 

music_on_hold_when_empty

yes/no

Sets whether music on hold should be played when only one person is in the conference or when the user is waiting on a marked user to enter the conference. By default, off.

 

music_on_hold_class

music on hold class

Sets the music on hold class to use for music on hold.

 

quiet

yes/no

When set to "yes," enter/leave prompts and user introductions are not played. By default, no.

 

announce_user_count

yes/no

Sets if the number of users in the conference should be announced to the caller. By default, no.

 

announce_user_count_all

yes/no; or an integer

Sets if the number of users should be announced to all other users in the conference when someone joins. When set to a number, the announcement will only occur once the user count is above the specified number

 

announce_only_user

yes/no

Sets if the only user announcement should be played when someone enters an empty conference. By default, yes.

 

announcement

filename

If set, the sound file specified by filename will be played to the user, and only the user, upon joining the conference bridge.

New in Asterisk 11

wait_marked

yes/no

Sets if the user must wait for another marked user to enter before joining the conference. By default, no.

 

end_marked

yes/no

If enabled, every user with this option in their profile will be removed from the conference when the last marked user exists the conference.

 

dsp_drop_silence

yes/no

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. This option is highly recommended for large conferences, due to its performance improvements.

 

dsp_talking_threshold

integer in milliseconds

The time, in milliseconds, by default 160, of sound above what the DSP has established as base-line silence for a user, before that user is considered to be talking. This value affects several options:

  1. Audio is only mixed out of a user's incoming audio stream if talking is detected. If this value is set too loose, the user will hear themselves briefly each time they begin talking until the DSP has time to establish that they are in fact talking.
  2. When talker detection AMI events are enabled, this value determines when talking has begun, which causes AMI events to fire. If this value is set too tight, AMI events may be falsely triggered by variants in the background noise of the caller.
  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 loose, the beginning of a user's speech will get cut off as they transition from silence to talking.

 

dsp_silence_threshold

integer in milliseconds

The time, in milliseconds, by default 2500, of sound falling within what the DSP has established as the baseline silence, before a user is considered to be silent. 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. This value affects several operations:

  1. When talker 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 that the user has stopped talking may get faslely 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 bridge, after the user stops talking. If this value is set too low, the user's audio stream may sound choppy to other participants.

 

talk_detection_events

yes/no

Sets whether or not notifications of when a user begins and ends talking should be sent out as events over AMI. By default, no.

 

denoise

yes/no

Whether or not a noise reduction filter should be applied to the audio before mixing. By default, off. This requires codec_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 still 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/no

Whether or not to place a jitter buffer on the caller's audio stream before any audio mixing is performed. This option is highly recommended, but will add a slight delay to the audio and will incur a slight performance penalty. This option makes use of the JITTERBUFFER dialplan function's default adaptive jitter buffer. For a more fine-tuned jitter buffer, disable this option and use the JITTERBUFFER dialplan function on the calling channel, before it enters the ConfBridge application.

 

pin

integer

Sets if the user must enter a PIN before joining the conference. The user will be prompted for the PIN.

 

announce_join_leave

yes/no

When enabled, this option prompts the user for their name when entering the conference. After the name is recorded, it will be played as the user enters and exists the conference. By default, no.

 

dtmf_passthrough

yes/no

Whether or not DTMF received from users should pass through the conference to other users. By default, no.

 

Example
In this example, we will create a user profile called "fancyuser" that includes music on hold, user count announcements, join/leave announcements, silence detection, noise reduction and requires a PIN of 456.

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Option

Values

Description

Notes

type

menu

Set this to menu to configure a conference menu

 

playback

(<name of audio file1>&<name of audio file2>&...)

Plays back an audio file, or a string of audio files chained together using the & character, to the user and then immediately returns them to the conference.

 

playback_and_continue

(<name of audio file 1>&<name of audio file 2>&...)

Plays back an audio file, or a series of audio files chained together using the & character, to the user while continuing the collect the DTMF sequence. This is useful when using a menu prompt that describes all of the menu options. Note that any DTMF during this action will terminate the prompt's playback.

 

toggle_mute

 

Toggles mute on and off. When a user is muted, they will not be able to speak to other conference users, but they can still listen to other users. While muted, DTMF keys from the caller will continue to be collected.

 

no_op

 

This action does nothing. Its only real purpose exists for being able to reserve a sequence in the configuration as a menu exit sequence.

 

decrease_listening_volume

 

Decreases the caller's listening volume. Everything they hear will sound quieter.

 

increase_listening_volume

 

Increases the caller's listening volume. Everything they hear will sound louder.

 

reset_listening_volume

 

Resets the caller's listening volume to the default level.

 

decrease_talking_volume

 

Decreases the caller's talking volume. Everything they say will sound quieter to other callers.

 

increase_talking_volume

 

Increases the caller's talking volume. Everything they say will sound louder to other callers.

 

reset_talking_volume

 

Resets the caller's talking volume to the default level.

 

dialplan_exec

(context,exten,priority)

Allows one to escape from the conference and execute commands in the dialplan. Once the dialplan exits, the user will be put back into the conference.

 

leave_conference

 

Allows a user to exit the conference and continue execution in the dialplan.

 

admin_kick_last

 

Allows an Admin to remove the last participant from the conference. This action only works for users whose User Profiles set them as conference Admins.

 

admin_toggle_conference_lock

 

Allows an Admin to toggle locking and unlocking the conference. When the conference is locked, only other Admin users can join. When the conference is unlocked, any user may join up to the limit defined by the max_members Bridge Profile option. This action only works for users whose User Profiles set them as conference Admins.

 

admin_toggle_mute_participants

 

Allows an Admin to mute/unmute all non-admin participants in the conference.

New in Asterisk 11

set_as_single_video_src

 

Allows a user to set themselves as the single video distribution source for all other participants. This overrides the video_mode setting.

 

release_as_single_video_src

 

Allows a user to release themselves as the single video source. Upon release of the video source, and/or if video_mode is set to "none," this action will result in the conference returning to whatever video mode the Bridge Profile is using. This action will have no effect if the user is not currently the video source. The user is also not guaranteed that the use of this action will prevent them from becoming the video source later.

 

participant_count

 

Plays back the current number of participants into the conference.

New in Asterisk 11

Example
In this example, we'll create a menu called "fancymenu." This menu will utilize many of the options listed above. We will construct a features menu that plays when the user enters the * character. Since we will do this using the playback_and_continue option, we will define other menu items as being a "subset" of the * command, e.g. *4, so that once the user presses *, they can listen to the menu options and then press the specific "after-star" option, e.g. 4, to affect the option. Additionally, we will duplicate those same sub-features as non-* features, so that the user does not need to have entered the * menu structure in order to affect the options, they can just press the key, e.g. "4" at any time, regardless of whether or not they're in the *-tree.

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