Please NOTE: this functionality is not yet implemented in 5.0.x

VoIP and SIP Integration

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.

SIP-Transport Integration

You need Apache OpenMeetings version 4.0+ to apply this guide!

You need Asterisk version 13+ to apply this guide!

Here is instruction how-to set up red5sip transport integration with OpenMeetings on Ubuntu 16.04.


Run the commands
sudo apt update && sudo apt upgrade


Building and setting up Asterisk

Run the commands
sudo mkdir /usr/src/asterisk && cd /usr/src/asterisk
sudo wget
sudo tar -xvzf asterisk-13.17.0.tar.gz
cd ./asterisk-13.17.0
sudo make clean
sudo contrib/scripts/install_prereq install
sudo ./configure
sudo make menuconfig

Make sure you have selected Add-ons -> res_config_mysql, Press F12 to save
sudo make
sudo make install
sudo make samples
sudo make config
sudo service asterisk start


Configure Asterisk

Enable asterisk MySQL module:

Modify "[modules]" section of /etc/asterisk/modules.conf as follows:
Add/uncomment the following lines
preload =>


Configure MySQL module:

Set valid data for MySQL in /etc/asterisk/res_config_mysql.conf :
dbhost =
dbname = openmeetings
dbuser = root
dbpass =
dbport = 3306
dbsock = /var/lib/mysql/mysql.sock
dbcharset = utf8


Modify /etc/asterisk/sip.conf
Add/uncomment the following line:

Increase maxexpiry value to 43200:

Add user for the "SIP Transport":


Add next lines into the /etc/asterisk/extconfig.conf:
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/applicationContext.xml
; 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.
; *****************************************************

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

exten => _400X!,1,Confbridge(${EXTEN},default_bridge,sip_user)
exten => _400X!,n,Hangup

; *****************************************************
; Extensions for outgoing calls from Openmeetings room.
; *****************************************************

exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
exten => _400X!,n(notavail),Hangup


Modify /etc/asterisk/confbridge.conf
Add/Modify the following secions:





To enable Asterisk Manager API modify /etc/asterisk/manager.conf
Add/Modify the following sections:
enabled = yes
webenabled = no
port = 5038
bindaddr =

secret = 12345
read = all
write = all


Update OpenMeetings with credentials for Asterisk manager. Modify /opt/om/webapps/openmeetings/WEB-INF/classes/applicationContext.xml
find <bean id="sipDao" class=""> uncomment its parameters and set it to your custom values.
set value for uid property to unique secret value (can be generated here and sync it with of red5sip (see below)

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

Restart asterisk:
service asterisk restart


Setup red5sip transport

  • Download red5sip from
    git clone
  • Build with Apache Maven
    cd red5sip
    mvn clean package
  • All necessary files will be available in target folder, copy/move it to /opt/red5sip/
  • Insert proper values to the /opt/red5sip/ # red5 server address
    om.context=openmeetings # Openmeetings context
    red5.codec.rate=22 # should correlate with mic setting in Admin->Config `flash.mic.rate`
    sip.obproxy= # asterisk adderss # sip phone number
    sip.authid=red5sip_user # sip auth id
    sip.secret=12345 # sip password
    sip.realm=asterisk # sip realm
    sip.proxy= # address of sip proxy
    rooms.forceStart=no # TBD
    uid=87dddad4-9ca5-475b-860f-2e0825d02b76 #can be generated here:
    rooms=1 # TBD (not in use)
  • Set correct permissions on red5sip files:
    sudo chown -R nobody:nogroup /opt/red5sip
  • Add red5sip to autostart:
    sudo cp /opt/red5sip/red5sip /etc/init.d/
    sudo chmod a+x /etc/init.d/red5sip
    sudo update-rc.d red5sip defaults
  • Start openmeetings
    service red5 start
  • Enable SIP in openmeetings:
    Administration->Configuration->red5sip.enable == yes
  • Enable SIP for particular room(s):
    Administration->Conference rooms->Room->Enable SIP transport in the room == checked
    (SIP number will be assigned to room if everything is OK)
  • Start red5sip
    service red5sip start

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