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<section>
<a id="VoIP_and_SIP_Integration"></a>
<h1 id="voip-and-sip-integration">VoIP and SIP Integration</h1>
<p>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.</p>
<p>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.</p>
<p>To get help on the integration you can contact the <a href="mailing-lists.html">mailing lists</a> or for example somebody from the list of <a href="commercial-support.html">commercial support</a>.</p>
</section>
<section>
<a id="Asterisk_Integration"></a>
<h1 id="asterisk-integration">Asterisk Integration</h1>
<p>You need Apache OpenMeetings <strong>version 9.0+</strong> to apply this guide!</p>
<p>You need Asterisk <strong>version 22+</strong> to apply this guide!</p>
<p>Here is the instruction how-to set up integration between OpenMeetings and Asterisk on Ubuntu 24.04.</p>
<section>
<a id="Prerequisites"></a>
<h2 id="prerequisites">Prerequisites</h2>
<pre><code>sudo apt update && sudo apt upgrade
</code></pre>
</section>
<section>
<a id="Building_and_setting_up_Asterisk"></a>
<h2 id="building-and-setting-up-asterisk">Building and setting up Asterisk</h2>
<pre><code>sudo apt install build-essential
cd /usr/src
sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-22.8.2.tar.gz
sudo tar -xzf asterisk-22.8.2.tar.gz
cd ./asterisk-22.8.2
sudo make clean
sudo contrib/scripts/install_prereq install
sudo ./configure
sudo make menuconfig
</code></pre>
<p>Make sure you have selected (<a href="https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients" class="externalLink">Asterisk WebRTC Config</a>)</p>
<ul>
<li>Add-ons -> res_config_mysql</li>
<li>Codec Translators -> codec_opus</li>
<li>Resource Modules -> res_crypto</li>
<li>Resource Modules -> res_http_websocket</li>
<li>Resource Modules -> res_pjsip_transport_websocket</li>
</ul>
<p>Press F12 to save</p>
<pre><code>sudo make
sudo make install
sudo make samples
sudo make config
</code></pre>
</section>
<section>
<a id="Configure_Asterisk"></a>
<h2 id="configure-asterisk">Configure Asterisk</h2>
<section>
<a id="Enable_asterisk_MySQL_module3A"></a>
<h3 id="enable-asterisk-mysql-module">Enable asterisk MySQL module:</h3>
<p>Modify <code>[modules]</code> section of <code>/etc/asterisk/modules.conf</code> <strong>Add/uncomment the following lines</strong></p>
<pre><code>preload = res_config_mysql.so
noload = chan_sip.so
</code></pre>
</section>
<section>
<a id="Configure_MySQL_module3A"></a>
<h3 id="configure-mysql-module">Configure MySQL module:</h3>
<p>Set valid data for MySQL in <code>/etc/asterisk/res_config_mysql.conf</code>:</p>
<p><strong>Example</strong></p>
<pre><code>[openmeetings]
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
</code></pre>
</section>
<section>
<a id="Configure_sorcery3A"></a>
<h3 id="configure-sorcery">Configure sorcery:</h3>
<p>Add next lines into the /etc/asterisk/sorcery.conf</p>
<pre><code>[res_pjsip] ; Realtime PJSIP configuration wizard
endpoint=realtime,ps_endpoints
endpoint=config,pjsip.conf,criteria=type=endpoint
auth=realtime,ps_auths
auth=config,pjsip.conf,criteria=type=auth
aor=realtime,ps_aors
aor=config,pjsip.conf,criteria=type=aor
</code></pre>
</section>
<section>
<a id="Configure_extensions3A"></a>
<h3 id="configure-extensions">Configure extensions:</h3>
<p>Add next lines into the <code>/etc/asterisk/extconfig.conf</code>:</p>
<pre><code>[settings]
ps_endpoints => mysql,openmeetings,om_sipuser_endpoint
ps_aors => mysql,openmeetings,om_sipuser_aor
ps_auths => mysql,openmeetings,om_sipuser_auth
</code></pre>
<p>Modify <code>/etc/asterisk/extensions.conf</code> <strong>Add the following section:</strong></p>
<pre><code>; *****************************************************
; 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 -f -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/${EXTEN})}]?ok:notavail)
same => n(ok),SET(PIN=${DB(openmeetings/rooms/${EXTEN})})
same => n,Set(CONFBRIDGE(user,template)=ombridge_inbound_user)
same => n,Set(CONFBRIDGE(user,pin)=${PIN})
same => n(ok),Confbridge(${EXTEN},default_bridge,)
same => n,Hangup
same => n(notavail),Answer()
same => n,Playback(invalid)
same => n,Hangup
[rooms-originate]
exten => _400X!,1,Confbridge(${EXTEN},default_bridge,ombridge_inbound_user)
same => n,Hangup
[rooms-out]
; *****************************************************
; Extensions for outgoing calls from Openmeetings room.
; *****************************************************
[rooms-omsip]
exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
same => n(ok),Confbridge(${EXTEN},default_bridge,ombridge_outbound_user)
same => n(notavail),Hangup
</code></pre>
</section>
<section>
<a id="Configure_Confbridge"></a>
<h3 id="configure-confbridge">Configure Confbridge</h3>
<p>Modify <code>/etc/asterisk/confbridge.conf</code></p>
<p><strong>Add/Modify the following sections:</strong></p>
<pre><code>[general]
[ombridge_outbound_user]
type=user
marked=yes
dsp_drop_silence=yes
denoise=true
[ombridge_inbound_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
</code></pre>
</section>
<section>
<a id="Configure_Asterisk_Manager"></a>
<h3 id="configure-asterisk-manager">Configure Asterisk Manager</h3>
<p>To enable Asterisk Manager API modify <code>/etc/asterisk/manager.conf</code></p>
<p><strong>Add/Modify the following sections:</strong></p>
<pre><code>[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
</code></pre>
<p>Update OpenMeetings with credentials for Asterisk manager. Modify <code>/opt/om/webapps/openmeetings/WEB-INF/classes/openmeetings.properties</code></p>
<p>find all properties start with <code>sip.</code> and set it to your custom values.</p>
<p style="font-size: larger; color: blue;">IMPORTANT: this step should be done <strong>BEFORE</strong> system install/restore otherwise all SIP related room information will be lost</p>
</section>
<section>
<a id="Configure_Asterisk27s_built-in_HTTP_server"></a>
<h3 id="configure-asterisks-built-in-http-server">Configure Asterisk's built-in HTTP server</h3>
<p>To communicate with WebSocket clients, Asterisk uses its built-in HTTP server. Configure <code>/etc/asterisk/http.conf</code> as follows:</p>
<pre><code>[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
</code></pre>
</section>
<section>
<a id="Configure_PJSIP"></a>
<h3 id="configure-pjsip">Configure PJSIP</h3>
<p>If you're not already familiar with configuring Asterisk's chan_pjsip driver, visit the res_pjsip configuration page.</p>
<p>Modify <code>/etc/asterisk/pjsip.conf</code> as follows:</p>
<pre><code>; So users can login via for ex. soft-phone
;
[transport-udp]
type=transport
protocol=udp ;udp,tcp,tls,ws,wss,flow
bind=0.0.0.0
; This one for internal OM-to-SIP connections
;
[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0
; All other transport parameters are ignored for wss transports.
</code></pre>
<p><strong>Add user for the “SIP Transport”</strong></p>
<pre><code>[omsip_user]
type=aor
max_contacts=1
remove_existing=yes
[omsip_user]
type=endpoint
context=rooms-omsip
disallow=all
allow=ulaw,opus,h264
transport=transport-wss
auth=omsip_user
aors=omsip_user
[omsip_user]
type=auth
auth_type=userpass
password=12345
username=omsip_user
</code></pre>
</section>
</section>
<section>
<a id="Call_from_room_to_external_number"></a>
<h2 id="call-from-room-to-external-number">Call from room to external number</h2>
<p>Modify <code>/etc/asterisk/pjsip.conf</code></p>
<p><strong>Add your external provider</strong></p>
<pre><code>register => tls://<name>:<password>@sipnet.ru
[SIPNET]
type=friend
username=<name>
secret=<password>
callerid=<caller id> ; can be commented out
host=sipnet.ru
nat=route
fromuser=<fromuser> ; can be commented out
fromdomain=sipnet.ru
dtmfmode=rfc2833
insecure=very
context=rooms
disallow=all
allow=alaw
canreinvite=no
callbackextension=<extension for incoming calls> ; can be commented out
</code></pre>
<p>Modify <code>/etc/asterisk/extensions.conf</code></p>
<p><strong>Add external numbers you are going to call to [rooms-out] section</strong></p>
<pre><code>[rooms-out]
; *****************************************************
; Extensions for outgoing calls from Openmeetings room.
; *****************************************************
exten => _00000,1,Answer
same => n,Dial(PJSIP/00000@SIPNET,30)
same => n,HangUp
</code></pre>
</section>
</section> </main>
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<ul class="nav--sidebar nav flex-column flex-nowrap">
<li class="h1"><a class="nav-link" href="#voip-and-sip-integration" title="VoIP and SIP Integration">VoIP and SIP Integration</a></li>
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