mirror of https://github.com/asterisk/asterisk
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: https://origsvn.digium.com/svn/asterisk/trunk@314598 65c4cc65-6c06-0410-ace0-fbb531ad65f310-digiumphones
David Vossel
14 years ago
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b8f253161b
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2011, Digium, Inc.
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*
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* David Vossel <dvossel@digium.com>
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* Joshua Colp <jcolp@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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#ifndef _CONFBRIDGE_H
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#define _CONFBRIDGE_H
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#include "asterisk.h"
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#include "asterisk/app.h"
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#include "asterisk/logger.h"
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#include "asterisk/linkedlists.h"
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#include "asterisk/channel.h"
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#include "asterisk/bridging.h"
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#include "asterisk/bridging_features.h"
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/* Maximum length of a conference bridge name */
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#define MAX_CONF_NAME 32
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/* Maximum length of a conference pin */
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#define MAX_PIN 80
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#define DEFAULT_USER_PROFILE "default_user"
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#define DEFAULT_BRIDGE_PROFILE "default_bridge"
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#define DEFAULT_TALKING_THRESHOLD 160
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#define DEFAULT_SILENCE_THRESHOLD 2500
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enum user_profile_flags {
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USER_OPT_ADMIN = (1 << 0), /*!< Set if the caller is an administrator */
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USER_OPT_NOONLYPERSON = (1 << 1), /*!< Set if the "you are currently the only person in this conference" sound file should not be played */
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USER_OPT_MARKEDUSER = (1 << 2), /*!< Set if the caller is a marked user */
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USER_OPT_STARTMUTED = (1 << 3), /*!< Set if the caller should be initially set muted */
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USER_OPT_MUSICONHOLD = (1 << 4), /*!< Set if music on hold should be played if nobody else is in the conference bridge */
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USER_OPT_QUIET = (1 << 5), /*!< Set if no audio prompts should be played */
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USER_OPT_ANNOUNCEUSERCOUNT = (1 << 6), /*!< Set if the number of users should be announced to the caller */
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USER_OPT_WAITMARKED = (1 << 7), /*!< Set if the user must wait for a marked user before starting */
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USER_OPT_ENDMARKED = (1 << 8), /*!< Set if the user should be kicked after the last Marked user exits */
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USER_OPT_DENOISE = (1 << 9), /*!< Sets if denoise filter should be used on audio before mixing. */
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USER_OPT_ANNOUNCE_JOIN_LEAVE = (1 << 10), /*!< Sets if the user's name should be recorded and announced on join and leave. */
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USER_OPT_TALKER_DETECT = (1 << 11), /*!< Sets if start and stop talking events should generated for this user over AMI. */
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USER_OPT_DROP_SILENCE = (1 << 12), /*!< Sets if silence should be dropped from the mix or not. */
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USER_OPT_DTMF_PASS = (1 << 13), /*!< Sets if dtmf should be passed into the conference or not */
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USER_OPT_ANNOUNCEUSERCOUNTALL = (1 << 14), /*!< Sets if the number of users should be announced to everyone. */
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USER_OPT_JITTERBUFFER = (1 << 15), /*!< Places a jitterbuffer on the user. */
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};
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enum bridge_profile_flags {
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BRIDGE_OPT_RECORD_CONFERENCE = (1 << 0), /*!< Set if the conference should be recorded */
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};
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enum conf_menu_action_id {
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MENU_ACTION_TOGGLE_MUTE = 1,
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MENU_ACTION_PLAYBACK,
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MENU_ACTION_PLAYBACK_AND_CONTINUE,
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MENU_ACTION_INCREASE_LISTENING,
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MENU_ACTION_DECREASE_LISTENING,
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MENU_ACTION_RESET_LISTENING,
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MENU_ACTION_RESET_TALKING,
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MENU_ACTION_INCREASE_TALKING,
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MENU_ACTION_DECREASE_TALKING,
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MENU_ACTION_DIALPLAN_EXEC,
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MENU_ACTION_ADMIN_TOGGLE_LOCK,
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MENU_ACTION_ADMIN_KICK_LAST,
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MENU_ACTION_LEAVE,
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MENU_ACTION_NOOP,
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};
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/*! The conference menu action contains both
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* the action id that represents the action that
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* must take place, along with any data associated
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* with that action. */
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struct conf_menu_action {
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enum conf_menu_action_id id;
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union {
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char playback_file[PATH_MAX];
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struct {
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char context[AST_MAX_CONTEXT];
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char exten[AST_MAX_EXTENSION];
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int priority;
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} dialplan_args;
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} data;
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AST_LIST_ENTRY(conf_menu_action) action;
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};
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/*! Conference menu entries contain the DTMF sequence
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* and the list of actions that are associated with that
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* sequence. */
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struct conf_menu_entry {
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/*! the DTMF sequence that triggers the actions */
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char dtmf[MAXIMUM_DTMF_FEATURE_STRING];
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/*! The actions associated with this menu entry. */
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AST_LIST_HEAD_NOLOCK(, conf_menu_action) actions;
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AST_LIST_ENTRY(conf_menu_entry) entry;
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};
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/*! Conference menu structure. Contains a list
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* of DTMF sequences coupled with the actions those
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* sequences invoke.*/
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struct conf_menu {
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char name[128];
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int delme;
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AST_LIST_HEAD_NOLOCK(, conf_menu_entry) entries;
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};
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struct user_profile {
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char name[128];
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char pin[MAX_PIN];
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char moh_class[128];
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unsigned int flags;
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unsigned int announce_user_count_all_after;
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/*! The time in ms of talking before a user is considered to be talking by the dsp. */
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unsigned int talking_threshold;
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/*! The time in ms of silence before a user is considered to be silent by the dsp. */
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unsigned int silence_threshold;
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int delme;
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};
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enum conf_sounds {
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CONF_SOUND_HAS_JOINED,
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CONF_SOUND_HAS_LEFT,
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CONF_SOUND_KICKED,
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CONF_SOUND_MUTED,
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CONF_SOUND_UNMUTED,
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CONF_SOUND_ONLY_ONE,
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CONF_SOUND_THERE_ARE,
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CONF_SOUND_OTHER_IN_PARTY,
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CONF_SOUND_PLACE_IN_CONF,
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CONF_SOUND_WAIT_FOR_LEADER,
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CONF_SOUND_LEADER_HAS_LEFT,
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CONF_SOUND_GET_PIN,
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CONF_SOUND_INVALID_PIN,
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CONF_SOUND_ONLY_PERSON,
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CONF_SOUND_LOCKED,
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CONF_SOUND_LOCKED_NOW,
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CONF_SOUND_UNLOCKED_NOW,
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CONF_SOUND_ERROR_MENU,
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CONF_SOUND_JOIN,
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CONF_SOUND_LEAVE,
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};
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struct bridge_profile_sounds {
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AST_DECLARE_STRING_FIELDS(
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AST_STRING_FIELD(hasjoin);
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AST_STRING_FIELD(hasleft);
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AST_STRING_FIELD(kicked);
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AST_STRING_FIELD(muted);
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AST_STRING_FIELD(unmuted);
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AST_STRING_FIELD(onlyone);
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AST_STRING_FIELD(thereare);
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AST_STRING_FIELD(otherinparty);
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AST_STRING_FIELD(placeintoconf);
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AST_STRING_FIELD(waitforleader);
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AST_STRING_FIELD(leaderhasleft);
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AST_STRING_FIELD(getpin);
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AST_STRING_FIELD(invalidpin);
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AST_STRING_FIELD(onlyperson);
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AST_STRING_FIELD(locked);
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AST_STRING_FIELD(lockednow);
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AST_STRING_FIELD(unlockednow);
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AST_STRING_FIELD(errormenu);
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AST_STRING_FIELD(leave);
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AST_STRING_FIELD(join);
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);
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};
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struct bridge_profile {
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char name[64];
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char rec_file[PATH_MAX];
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unsigned int flags;
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unsigned int max_members; /*!< The maximum number of participants allowed in the conference */
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unsigned int internal_sample_rate; /*!< The internal sample rate of the bridge. 0 when set to auto adjust mode. */
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unsigned int mix_interval; /*!< The internal mixing interval used by the bridge. When set to 0 the bridgewill use a default interval. */
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struct bridge_profile_sounds *sounds;
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int delme;
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};
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/*! \brief The structure that represents a conference bridge */
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struct conference_bridge {
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char name[MAX_CONF_NAME]; /*!< Name of the conference bridge */
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struct ast_bridge *bridge; /*!< Bridge structure doing the mixing */
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struct bridge_profile b_profile; /*!< The Bridge Configuration Profile */
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unsigned int users; /*!< Number of users present */
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unsigned int markedusers; /*!< Number of marked users present */
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unsigned int locked:1; /*!< Is this conference bridge locked? */
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struct ast_channel *playback_chan; /*!< Channel used for playback into the conference bridge */
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struct ast_channel *record_chan; /*!< Channel used for recording the conference */
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pthread_t record_thread; /*!< The thread the recording chan lives in */
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ast_mutex_t playback_lock; /*!< Lock used for playback channel */
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AST_LIST_HEAD_NOLOCK(, conference_bridge_user) users_list; /*!< List of users participating in the conference bridge */
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};
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/*! \brief The structure that represents a conference bridge user */
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struct conference_bridge_user {
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struct conference_bridge *conference_bridge; /*!< Conference bridge they are participating in */
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struct bridge_profile b_profile; /*!< The Bridge Configuration Profile */
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struct user_profile u_profile; /*!< The User Configuration Profile */
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char menu_name[64]; /*!< The name of the DTMF menu assigned to this user */
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char name_rec_location[PATH_MAX]; /*!< Location of the User's name recorded file if it exists */
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struct ast_channel *chan; /*!< Asterisk channel participating */
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struct ast_bridge_features features; /*!< Bridge features structure */
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struct ast_bridge_tech_optimizations tech_args; /*!< Bridge technology optimizations for talk detection */
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unsigned int kicked:1; /*!< User has been kicked from the conference */
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AST_LIST_ENTRY(conference_bridge_user) list; /*!< Linked list information */
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};
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/*! \brief load confbridge.conf file */
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int conf_load_config(int reload);
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/*! \brief destroy the information loaded from the confbridge.conf file*/
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void conf_destroy_config(void);
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/*!
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* \brief find a user profile given a user profile's name and store
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* that profile in result structure.
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*
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* \details This function first attempts to find any custom user
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* profile that might exist on a channel datastore, if that doesn't
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* exist it looks up the provided user profile name, if that doesn't
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* exist either the default_user profile is used.
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* \retval user profile on success
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* \retval NULL on failure
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*/
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const struct user_profile *conf_find_user_profile(struct ast_channel *chan, const char *user_profile_name, struct user_profile *result);
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/*!
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* \brief Find a bridge profile
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*
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* \details Any bridge profile found using this function must be
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* destroyed using conf_bridge_profile_destroy. This function first
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* attempts to find any custom bridge profile that might exist on
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* a channel datastore, if that doesn't exist it looks up the
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* provided bridge profile name, if that doesn't exist either
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* the default_bridge profile is used.
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*
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* \retval Bridge profile on success
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* \retval NULL on failure
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*/
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const struct bridge_profile *conf_find_bridge_profile(struct ast_channel *chan, const char *bridge_profile_name, struct bridge_profile *result);
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/*!
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* \brief Destroy a bridge profile found by 'conf_find_bridge_profile'
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*/
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void conf_bridge_profile_destroy(struct bridge_profile *b_profile);
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/*!
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* \brief copies a bridge profile
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* \note conf_bridge_profile_destroy must be called on the dst structure
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*/
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void conf_bridge_profile_copy(struct bridge_profile *dst, struct bridge_profile *src);
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/*!
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* \brief Set a DTMF menu to a conference user by menu name.
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*
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* \retval 0 on success, menu was found and set
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* \retval -1 on error, menu was not found
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*/
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int conf_set_menu_to_user(const char *menu_name, struct conference_bridge_user *conference_bridge_user);
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/*!
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* \brief Finds a menu_entry in a menu structure matched by DTMF sequence.
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*
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* \note the menu entry found must be destroyed using conf_menu_entry_destroy()
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*
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* \retval 1 success, entry is found and stored in result
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* \retval 0 failure, no entry found for given DTMF sequence
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*/
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int conf_find_menu_entry_by_sequence(const char *dtmf_sequence, struct conf_menu *menu, struct conf_menu_entry *result);
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/*!
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* \brief Destroys and frees all the actions stored in a menu_entry structure
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*/
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void conf_menu_entry_destroy(struct conf_menu_entry *menu_entry);
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/*!
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* \brief Once a DTMF sequence matches a sequence in the user's DTMF menu, this function will get
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* called to perform the menu action.
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*
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* \param bridge_channel, Bridged channel this is involving
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* \param conference_bridge_user, the conference user to perform the action on.
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* \param menu_entry, the menu entry that invoked this callback to occur.
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* \param menu, an AO2 referenced pointer to the entire menu structure the menu_entry
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* derived from.
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*
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* \note The menu_entry is a deep copy of the entry found in the menu structure. This allows
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* for the menu_entry to be accessed without requiring the menu lock. If the menu must
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* be accessed, the menu lock must be held. Reference counting of the menu structure is
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* handled outside of the scope of this function.
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*
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* \retval 0 success
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* \retval -1 failure
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*/
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int conf_handle_dtmf(
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struct ast_bridge_channel *bridge_channel,
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struct conference_bridge_user *conference_bridge_user,
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struct conf_menu_entry *menu_entry,
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struct conf_menu *menu);
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/*! \brief Looks to see if sound file is stored in bridge profile sounds, if not
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* default sound is provided.*/
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const char *conf_get_sound(enum conf_sounds sound, struct bridge_profile_sounds *custom_sounds);
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int func_confbridge_helper(struct ast_channel *chan, const char *cmd, char *data, const char *value);
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#endif
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@ -0,0 +1,302 @@
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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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; --- 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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;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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;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 user with this option set in their
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; user profile after the last Marked user exists the conference.
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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 ; The time in milliseconds of sound above what the dsp has
|
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; established as base line silence for a user before a user
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; is considered to be talking. This value affects several
|
||||
; operations and should not be changed unless the impact on
|
||||
; 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 stream
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; if talking is detected. If this value is set too
|
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; loose the user will hear themselves briefly each
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; time they begin talking until the dsp has time to
|
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; establish that they are in fact talking.
|
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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 AMI event to fire. If this value is set too tight
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; AMI events may be falsely triggered by variants in
|
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; room noise.
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||||
; 3. The drop_silence option depends on this value to determine
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||||
; when the user's audio should be mixed into the bridge
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; after periods of silence. If this value is too loose
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||||
; the beginning of a user's speech will get cut off as they
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; transition from silence to talking.
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;
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||||
; By default this value is 160 ms. Valid values are 1 through 2^31
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;dsp_silence_threshold=2000 ; The time in milliseconds of sound falling within the what
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||||
; the dsp has established as baseline silence before a user
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||||
; 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
|
||||
Loading…
Reference in new issue