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authordvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b>2011-04-21 18:11:40 +0000
committerdvossel <dvossel@f38db490-d61c-443f-a65b-d21fe96a405b>2011-04-21 18:11:40 +0000
commitc7b7b920afe4afccb39c7ea3c8d0bd4e56ad41f9 (patch)
tree3be8de1bbbce5eb12a63028caa39167d69b54e96 /configs
parentc064284e6c7e022570158f7ac3f1b522e411007a (diff)
New HD ConfBridge conferencing application.
Includes a new highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8khz-192khz. Review: https://reviewboard.asterisk.org/r/1147/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314598 f38db490-d61c-443f-a65b-d21fe96a405b
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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.
+;
+
+; --- 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
+ ; 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 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.
+;dtmf_passthrough=yes ; Sets whether or not DTMF should pass through the conference.
+ ; This option is off 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.
+
+;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.
+
+; 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.
+;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.
+
+; --- 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.
+;
+; 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.
+ ; continue to collect the dtmf sequence.
+; 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.
+
+[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=no_op
+8=no_op
+*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