There are multiple ways to integrate with VoIP and or SIP. OpenMeetings does not provide out of the box a ready to run VoIP integration / integration to cell phone or usual land lane. The nature of such integrations is that it depends heavily on the infrastructure that you are using and where you would like to integrate OpenMeetings into.

It also depends on a number of factors of which OpenMeetings is impossible to set up for you, for example setting up your VoIP server or provide you with a range of telephone numbers reserved for conference calls in your national phone network. Such an integration project is likely to become a consulting job for a telecommunications consultant.

To get help on the integration you can contact the mailing lists or for example somebody from the list of commercial support.

Asterisk Integration

You need Apache OpenMeetings version 6.0+ to apply this guide!

You need Asterisk version 16+ to apply this guide!

Here is the instruction how-to set up integration between OpenMeetings and Asterisk on Ubuntu 18.04.

Prerequisites

sudo apt update && sudo apt upgrade

Building and setting up Asterisk

sudo apt install build-essential

sudo mkdir /usr/src/asterisk && cd /usr/src/asterisk
sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-18.12.1.tar.gz
sudo tar -xvzf asterisk-18.12.1.tar.gz
cd ./asterisk-18.12.1
sudo make clean
sudo contrib/scripts/install_prereq install
sudo ./configure
sudo make menuconfig

Make sure you have selected (Asterisk WebRTC Config)

Add-ons -> res_config_mysql
Codec Translators -> codec_opus
Resource Modules -> res_crypto
Resource Modules -> res_http_websocket
Resource Modules -> res_pjsip_transport_websocket

Press F12 to save

sudo make
sudo make install
sudo make samples
sudo make config

Configure Asterisk

Enable asterisk MySQL module:

Modify [modules] section of /etc/asterisk/modules.conf Add/uncomment the following lines

preload = res_config_mysql.so

Remove/Comment following lines

;noload = chan_sip.so

Configure MySQL module:

Set valid data for MySQL in /etc/asterisk/res_config_mysql.conf:

Example

[general]
dbhost = 127.0.0.1
dbname = openmeetings
dbuser = om_db_admin
dbpass = 12345
dbport = 3306
dbsock = /var/lib/mysql/mysql.sock
dbcharset = utf8
requirements=warn

Configure SIP module:

Modify /etc/asterisk/sip.conf

Add/uncomment the following lines

videosupport=yes
rtcachefriends=yes

Increase maxexpiry value to 43200

maxexpiry=43200

Add user for the “SIP Transport”

[omsip_user]
host=dynamic
secret=12345
context=rooms-omsip
transport=ws,wss
type=friend
encryption=no
avpf=yes
icesupport=yes
directmedia=no
allow=!all,opus,h264

Configure extensions:

Add next lines into the /etc/asterisk/extconfig.conf:

[settings]
sippeers => mysql,general,sipusers

Modify /etc/asterisk/extensions.conf Add the following section:

; *****************************************************
; The below dial plan is used to dial into a Openmeetings Conference room
; The first line DB_EXISTS(openmeetings/room/ does not belong to the openmeetings application
; but is the name of astDB containing the astDB family/key pair and values
; To Check if your astDB has been created do the following in a terminal window type the following:
; asterisk –rx “database show”
; If you do not receive an output with that resembles openmeetings/rooms/400## where “##” will equal
; the extension assigned when you created your room
; If you do not receive the above output check your parameters in
; /opt/om/webapps/openmeetings/WEB-INF/classes/openmeetings.properties
; Go back into the Administrator Panel and remove the PIN number in each room save the record with
; no PIN number and then re-enter the pin again resave the record.
; *****************************************************

[rooms]
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
exten => _400X!,n(ok),SET(PIN=${DB(openmeetings/rooms/${EXTEN})})
exten => _400X!,n,Set(CONFBRIDGE(user,template)=sip_user)
exten => _400X!,n,Set(CONFBRIDGE(user,pin)=${PIN})
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,)
exten => _400X!,n,Hangup
exten => _400X!,n(notavail),Answer()
exten => _400X!,n,Playback(invalid)
exten => _400X!,n,Hangup

[rooms-originate]
exten => _400X!,1,Confbridge(${EXTEN},default_bridge,sip_user)
exten => _400X!,n,Hangup

[rooms-out]
; *****************************************************
; Extensions for outgoing calls from Openmeetings room.
; *****************************************************

[rooms-omsip]
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,omsip_user)
exten => _400X!,n(notavail),Hangup

Configure Confbridge

Modify /etc/asterisk/confbridge.conf

Add/Modify the following sections:

[general]

[omsip_user]
type=user
marked=yes
dsp_drop_silence=yes
denoise=true

[sip_user]
type=user
end_marked=yes
wait_marked=yes
music_on_hold_when_empty=yes
dsp_drop_silence=yes
denoise=true

[default_bridge]
type=bridge
video_mode=follow_talker

Configure Asterisk Manager

To enable Asterisk Manager API modify /etc/asterisk/manager.conf

Add/Modify the following sections:

[general]
enabled = yes
webenabled = no
port = 5038
bindaddr = 127.0.0.1

[openmeetings]
secret = 12345
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = all
write = all

Update OpenMeetings with credentials for Asterisk manager. Modify /opt/om/webapps/openmeetings/WEB-INF/classes/openmeetings.properties

find all properties start with sip. and set it to your custom values.

IMPORTANT: this step should be done BEFORE system install/restore otherwise all SIP related room information will be lost

Configure Asterisk's built-in HTTP server

To communicate with WebSocket clients, Asterisk uses its built-in HTTP server. Configure /etc/asterisk/http.conf as follows:

[general]
enabled=yes
bindaddr=127.0.0.1  ; or your Asterisk IP
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/asterisk/keys/asterisk.crt
tlsprivatekey=/etc/asterisk/keys/asterisk.key

Configure PJSIP

If you're not already familiar with configuring Asterisk's chan_pjsip driver, visit the res_pjsip configuration page.

Modify /etc/asterisk/pjsip.conf as follows:

[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0
; All other transport parameters are ignored for wss transports.

Call from room to external number