Not implemented

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.

Prerequisites

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 http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-13.17.0.tar.gz
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 => res_config_mysql.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 = root
dbpass =
dbport = 3306
dbsock = /var/lib/mysql/mysql.sock
dbcharset = utf8
requirements=warn

Modify /etc/asterisk/sip.conf
Add/uncomment the following line: 
videosupport=yes
rtcachefriends=yes
Increase maxexpiry value to 43200: 
maxexpiry=43200
Add user for the "SIP Transport": 
[red5sip_user]
type=friend
secret=12345
disallow=all
allow=ulaw
allow=h263
host=dynamic
nat=force_rport,comedia
context=rooms-red5sip

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/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.
; *****************************************************

[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-red5sip]
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: 
[general]

[red5sip_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

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/applicationContext.xml
find <bean id="sipDao" class="org.apache.openmeetings.db.dao.room.SipDao"> uncomment its parameters and set it to your custom values.
set value for uid property to unique secret value (can be generated here https://www.uuidtools.com) and sync it with settings.properties 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 https://github.com/openmeetings/red5sip 
    git clone https://github.com/openmeetings/red5sip.git
  • 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/settings.properties 
    red5.host=127.0.0.1 # red5 server address
om.context=openmeetings # Openmeetings context
red5.codec=asao
red5.codec.rate=22 # should correlate with mic setting in Admin->Config `flash.mic.rate`
sip.obproxy=127.0.0.1 # asterisk adderss
sip.phone=red5sip_user # sip phone number
sip.authid=red5sip_user # sip auth id
sip.secret=12345 # sip password
sip.realm=asterisk # sip realm
sip.proxy=127.0.0.1 # address of sip proxy
rooms.forceStart=no # TBD
uid=87dddad4-9ca5-475b-860f-2e0825d02b76 #can be generated here: https://www.uuidtools.com/
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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