Asterisk, since its early days, has offered a conferencing application called MeetMe (app_meetme.so). MeetMe provides DAHDI-mixed software-based bridges for multi-party audio conferencing. MeetMe is used by nearly all Asterisk implementations - small office, call center, large office, feature-server, third-party application, etc. It has been extremely successful as an audio bridge.
Over time, several significant limitations of MeetMe have been encountered by its users. Among these are two of distinction: MeetMe requires DAHDI for mixing, and is thus limited to 8kHz (PSTN) audio sampling rates; and MeetMe is delivered in a fairly static form, it does not provide extensive configuration options.
To address these limitations, a new conferencing application, based upon the ConfBridge application introduced in Asterisk 1.6.0, is now available with Asterisk 10. This new ConfBridge application replaces the older ConfBridge application. It is not intended to be a direct replacement for MeetMe, it will not provide feature parity with the MeetMe application. Instead, the new ConfBridge application delivers a completely redesigned set of functionality that most users will find more than sufficient, and in many ways better, for their conferencing needs.
; 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.
;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 ; The time in milliseconds of sound above what the dsp has
; established as base line silence for a user before a user
; is considered to be 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. 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 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 tight
; 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 loose
; the beginning of a user's speech will get cut off as they
; transition from silence to talking.
;
; By default this value is 160 ms. Valid values are 1 through 2^31
;dsp_silence_threshold=2000 ; The time in milliseconds of sound falling within the what
; the dsp has established as baseline silence before a user
; is considered be silent. 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 ms of silence a user may generate during
; natural speech.
;
; By default this value is 2500ms. Valid values are 1 through 2^31
;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.
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作者: 角色 時間: 2015-6-7 21:39
本帖最後由 角色 於 2015-6-7 21:42 編輯
confbridge.conf (continued)
;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.
; --- 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.
;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.
;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.
;language=en ; Set the language used for announcements to the conference.
; Default is en (English).
; 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 it toggled on.
;sound_unmuted ; The sound played when the mute option it toggled off.
;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.
複製代碼
作者: 角色 時間: 2015-6-7 21:43
本帖最後由 角色 於 2015-6-7 21:45 編輯
confbridge.conf (continued...2)
; --- 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.
; 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. All
; admin users are unaffected by this option. Note that all
; users, regardless of their admin status, are notified
; that the conference is muted.
; 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
複製代碼
作者: 角色 時間: 2015-6-8 23:28
本帖最後由 角色 於 2015-6-8 23:38 編輯
Example 1: The simplest ConfBridge application function
extensions.conf
exten => 4000,1,ConfBridge(101); join Room 101
複製代碼
Create an extension representing the conference meeting room 101.
The user can make a phone call to the extension of 4000 to enter the conference room. It can accommodate as many as users who dialed the extension number 40000.
