I recently had a play with conference calling in Asterisk. It’s pretty easy to use the built in applications in combination with the
asterisk-extra-sounds sound libraries to produce quite a professional feeling conferencing system.
One of the things people most often ask for in a new Asterisk installation is the ability to host teleconferences (essentially multi-way phonecalls). Asterisk offers a flexible way to achieve this.
I’ll be using Asterisk 13 (the latest and greatest at the time of writing), but there shouldn’t be too much difference for earlier Asterisk versions.
ConfBridge or MeetMe?
Asterisk ships with two conferencing applications - MeetMe (
app_meetme) and ConfBridge (
app_confbridge). ConfBridge has now pretty much replaced MeetMe in terms of functionality, but there might still be a few cases for using MeetMe. This post will cover ConfBridge, since that seems to have become the most accepted tool in newer versions of Asterisk (11+).
It should also be said that MeetMe requires DAHDI for a number of internal tasks such as timing and channel mixing, so that may restrict you from using it in some situations where you don’t want to run your Asterisk installation with DAHDI.
I’ll assume we’ve already got Asterisk installed and operational.
Now, we’re British, so Allison is much too American for our liking - as lovely as her voice is. Joking aside, the
en_GB prompts are more suitable for UK companies. You can download a wide variety of sounds from the Asterisk Downloads site for free. For this example, we’ll need the
asterisk-extra-sounds-en_GB sound archive to be extracted to a
sounds/en_GB/ directory in the
astdatadir directory (typically
You should choose the codec that fits the one used in the environment for which you’re setting up Asterisk. If you’re not sure,
alaw is a safe enough bet.
ConfBridge is configured in
confbridge.conf in your Asterisk configuration directory (typically
/etc/asterisk). The configuration file can define three types of entity: Conference Bridge Profiles (
type=bridge), User Profiles (
type=user) and Menus (
From this point on, all configuration is being added to
confbridge.conf, since we’re setting up ConfBridge itself.
Conference Bridge Profiles
A Conference Bridge Profile defines a template for conferences created with that profile. Here is the simplest possible Conference Bridge Profile:
This just defines a Conference Bridge Profile with the language set to
en_GB, meaning the IVR (Interactive Voice Response) prompts used for conferences using this Conference Bridge Profile will use sounds from the
sounds/en_GB/ subdirectory under the
astdatadir directory configured in
The Asterisk Wiki has information about the possible options for Conference Bridge Profiles.
A User Profile defines a template for users joining a conference. It includes configuration for how the user should be treated by Asterisk at various parts of the conference lifecycle. For instance, the
music_on_hold_when_empty option controls whether or not users with this profile should hear music-on-hold if they’re in an empty conference.
Here’s a simple User Profile:
Users with this profile will also be asked to say their name when they join a conference, and their name will be announced by Asterisk to the rest of the conference right before they’re added.
As before, the Asterisk Wiki has information about the possible options for User Profiles.
A Menu defines the buttons that a user can press while they’re in a conference call, and what those buttons do. You do as much or as little as you want here. I like to allow the user to adjust their listening volume and to mute and unmute themselves, so here’s a menu to do that:
This might seem a little complex, but it’s actually straightforward.
When a user presses
*, Asterisk will execute the
* menu item -
playback_and_continue(...). The string between the parenthesis is a concatenation of the sounds that should be played to the user. The sounds will read: “Press 1 to decrease the listening volume. Press 2 to increase the listening volume. Press 3 to mute or unmute yourself. To exit the menu, press 0.”
If the user presses an option after pressing
*, the relevant option (
*0) will be executed. If the user presses an option without first pressing
*, the relevant option will be executed (
The Asterisk Wiki has information about the other options you can present to users in conference menus. There are even special admin options that you can offer to users with the
is_admin flag set to 1 in their User Profile.
Creating a Conference
Once you’ve configured a Conference Bridge Profile, a User Profile and a Menu, you can start creating conferences based on them.
From this point on, all the configuration should go in
extensions.conf, since we’re setting up the dialplan.
Initiating a Conference
First, we’ll create an macro that will initiate new conferences:
Then we’ll create an extension that users can call to initiate a new conference:
That’s it - users can create conferences by calling the
9000 extension. Here’s an example of what that sounds like for a user:
Now we need a way for users to enter existing conferences.
First, we’ll write a macro that requests a conference number from the user and then dumps them into that conference if it exists:
With the macro in place, we can call it from anywhere we need to. For instance, straight in an inbound context for users dialling in to a dedicated conference call number:
This way, users dialling in on the
5554202 extension will be given the option to join a conference by keying in their conference number. Here’s an example of how joining a conference sounds to a user:
Note that you can’t hear the DTMF tones of the keys I’m pressing when I enter the Conference Number here because they’re being sent as NTEs (Named Telephone Events) in the RTP data stream (see RFC 4733), not as in-band audio tones as you might expect with a traditional telephone. Asterisk will handle tones arriving in-band just fine too (for instance from a POTS line connected via an FXO card), however.
Let me know if this helps you out with setting up conference calling in Asterisk.