How to configure ConfBridge conference in Asterisk 11.20.0

ConfBridge Conference

ConfBridge provides four internal concepts:

  1. Conference Number
  2. Bridge Profile
  3. User Profile
  4. Conference Menu

A Conference Number is a numerical representation for an instance of the bridge. Callers joined to the same conference number will be in the same conference bridge; they’re connected. Callers joined to different conference numbers are not in the same conference bridge; they’re separated. Conference Numbers are assigned in the dialplan. Unlike MeetMe, they’re not pre-reserved.

A Bridge Profile is a named set of options that control the behavior of a particular conference bridge. Each bridge must have its own profile. A single bridge cannot have more than one Bridge Profile.

A User Profile is a named set of options that control the user’s experience as a member of a particular bridge. Each user participating in a bridge can have their own individual User Profile.

A Conference Menu is a named set of options that are provided to a user when they present DTMF keys while connected to the bridge. Each user participating in a bridge can have their own individual Conference Menu.

Let’s Start configuring:

Step 1: Open your confbridge.conf file and make changes into it.

[general]
 ;This is where you enter your general configuration
[default_user]
type=user
marked=yes ; Sets if this is a marked user or not. 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.
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.
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.
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.
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.
[default_bridge]
type=bridge
mixing_interval=10 ; 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.
sound_join=yes ; The sound played to everyone when someone enters the conference.
sound_leave=yes ; The sound played to everyone when someone leaves the conference.
sound_has_joined=yes ; 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=yes ; The sound played when announcing someone's name has
                ; left the conference. This is used for user intros.
                ; Example "_____ has left the 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
 ;2=leave_conference
 ;3=dialplan_exec(addcallers,conference_joiner,1,1)
 ;3=dialplan_exec(conferencerooms,conference_joiner2,1)
0=dialplan_exec(conferencerooms,conference_joiner,1)

When Dialing out or inviting other extension in the conference room pressing “0” will take you out in the room for a moment and ask to key-in the extension you want to call

 

Step 2: Now we can proceed to edit the dialplan extensions.conf

[conferencerooms]
 ; standard participant
exten => 1030,1,NoOp()
 same => n,Goto(conference,1)

exten => conference,1,NoOp()
 same => n,Set(thisBridge=primary)
 same => n,ConfBridge(${thisBridge},,,sample_user_menu)

exten => conference_joiner,1,NoOp()
 same => n,Read(numberToDial,vm-enter-num-to-call)
 same => n,Playback(vm-dialout)
 same => n,Originate(SIP/${numberToDial},exten,conferencerooms,1030,1)
 same => n,Playback(conf-hasjoin)
 ; DO NOT ADD Hangup() in here

Now that we have created the conference room called “1030” you can try now to dial in your softphone/hardphone to test but before that reload first the confbridge module and dialplan

Step 3: Reload the ConfBridge module and DialPlan open your terminal and type

[[email protected] ~]# asterisk -rvvv
OR
[[email protected] ~]# rasterisk

pbx*CLI> module reload app_confbridge.so
pbx*CLI> dialplan reload

if you want to see some action in your terminal console enable the “debug and verbose” by typing below in your asterisk console

pbx*CLI> core set verbose 4
pbx*CLI> core set debug 4

 

 

4 Comments on "How to configure ConfBridge conference in Asterisk 11.20.0"

  1. Hi,

    I went through to this article and i followed all steps which are mentioned above. When i dialed 1030, my soft phone is saying that “Call can not be completed as dialed. Please check the number and dial again.”. I am new in conf bridge. So please suggest me that which step is causes this issue.

  2. asteriskcoder | March 20, 2018 at 20:54 | Reply

    Hi, I tried following this tutorial but I can’t seem to get the add caller to work. When I press 0 nothing happens. Could it be because I am running Asterisk 13.4.0?

    Any help you could provide I greatly appreciate it!

    Thanks!

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