EVOLUTION-MANAGER

Edit File: voip-sip-integration.html

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</pre> 
  </div> 
 </div> 
</section> 
<section> 
 <h2 id="building-and-setting-up-asterisk">Building and setting up Asterisk</h2> 
 <div>
   Run the commands 
  <div class="source"> 
   <pre>sudo mkdir /usr/src/asterisk &amp;&amp; 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

</pre> 
  </div> Make sure you have selected 
  <code>Add-ons -&gt; res_config_mysql</code>, Press F12 to save 
  <div class="source"> 
   <pre>sudo make
sudo make install
sudo make samples
sudo make config
sudo service asterisk start

</pre> 
  </div> 
 </div> 
</section> 
<section> 
 <h2 id="configure-asterisk">Configure Asterisk</h2> 
 <div>
   Enable asterisk MySQL module: 
  <br> 
  <br> Modify "[modules]" section of 
  <code>/etc/asterisk/modules.conf</code> as follows: 
  <br> 
  <b>Add/uncomment the following lines</b> 
  <div class="source"> 
   <pre>preload =&gt; res_config_mysql.so

</pre> 
  </div> 
 </div> 
 <br> 
 <div>
   Configure MySQL module: 
  <br> 
  <br> Set valid data for MySQL in 
  <code>/etc/asterisk/res_config_mysql.conf</code> : 
  <br> 
  <b>Example</b> 
  <div class="source"> 
   <pre>[general]
dbhost = 127.0.0.1
dbname = openmeetings
dbuser = root
dbpass =
dbport = 3306
dbsock = /var/lib/mysql/mysql.sock
dbcharset = utf8
requirements=warn

</pre> 
  </div> 
 </div> 
 <br> 
 <div>
   Modify 
  <code>/etc/asterisk/sip.conf</code> 
  <br> 
  <b>Add/uncomment the following line</b>: 
  <br> 
  <div class="source"> 
   <pre>videosupport=yes
rtcachefriends=yes

</pre> 
  </div> 
  <b>Increase maxexpiry value to 43200</b>: 
  <br> 
  <div class="source"> 
   <pre>maxexpiry=43200

</pre> 
  </div> 
  <b>Add user for the "SIP Transport"</b>: 
  <br> 
  <div class="source"> 
   <pre>[red5sip_user]
type=friend
secret=12345
disallow=all
allow=ulaw
allow=h263
host=dynamic
nat=force_rport,comedia
context=rooms-red5sip

</pre> 
  </div> 
 </div> 
 <br> 
 <div>
   Add next lines into the 
  <code>/etc/asterisk/extconfig.conf</code>: 
  <div class="source"> 
   <pre>[settings]
sippeers =&gt; mysql,general,sipusers

</pre> 
  </div> 
 </div> 
 <br> 
 <div>
   Modify 
  <code>/etc/asterisk/extensions.conf</code> 
  <br> 
  <b>Add the following section</b>: 
  <br> 
  <div class="source"> 
   <pre>; *****************************************************
; 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 =&gt; _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
exten =&gt; _400X!,n(ok),SET(PIN=${DB(openmeetings/rooms/${EXTEN})})
exten =&gt; _400X!,n,Set(CONFBRIDGE(user,template)=sip_user)
exten =&gt; _400X!,n,Set(CONFBRIDGE(user,pin)=${PIN})
exten =&gt; _400X!,n(ok),Confbridge(${EXTEN},default_bridge,)
exten =&gt; _400X!,n,Hangup
exten =&gt; _400X!,n(notavail),Answer()
exten =&gt; _400X!,n,Playback(invalid)
exten =&gt; _400X!,n,Hangup

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

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

[rooms-red5sip]
exten =&gt; _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail)
exten =&gt; _400X!,n(ok),Confbridge(${EXTEN},default_bridge,red5sip_user)
exten =&gt; _400X!,n(notavail),Hangup

</pre> 
  </div> 
 </div> 
 <br> 
 <div>
   Modify 
  <code>/etc/asterisk/confbridge.conf</code> 
  <br> 
  <b>Add/Modify the following secions</b>: 
  <br> 
  <div class="source"> 
   <pre>[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

</pre> 
  </div> 
 </div> 
 <br> 
 <div>
   To enable Asterisk Manager API modify 
  <code>/etc/asterisk/manager.conf</code> 
  <br> 
  <b>Add/Modify the following sections</b>: 
  <br> 
  <div class="source"> 
   <pre>[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

</pre> 
  </div> 
 </div> 
 <br> 
 <div>
   Update OpenMeetings with credentials for Asterisk manager. Modify 
  <code>/opt/om/webapps/openmeetings/WEB-INF/classes/applicationContext.xml</code> 
  <br> find 
  <b>&lt;bean id="sipDao" class="org.apache.openmeetings.db.dao.room.SipDao"&gt;</b> uncomment its parameters and set it to your custom values. 
  <br> set value for 
  <code>uid</code> property to unique secret value (can be generated here 
  <a class="externalLink" href="https://www.uuidtools.com">https://www.uuidtools.com</a>) and sync it with 
  <code>settings.properties</code> of red5sip (see below) 
  <p style="font-size: larger; color: blue;"> IMPORTANT: this step should be done <b>BEFORE</b> system install/restore otherwise all SIP related room information will be lost </p> 
 </div> 
 <br> 
 <div>
   Restart asterisk: 
  <div class="source"> 
   <pre>service asterisk restart

</pre> 
  </div> 
 </div> 
 <br> 
</section> 
<section> 
 <h2 id="setup-red5sip-transport">Setup red5sip transport</h2> 
 <ul> 
  <li>Download red5sip from <code>https://github.com/openmeetings/red5sip</code> 
   <div class="source"> 
    <pre>git clone https://github.com/openmeetings/red5sip.git

</pre> 
   </div> </li> 
  <li>Build with Apache Maven 
   <div class="source"> 
    <pre>cd red5sip
mvn clean package

</pre> 
   </div> </li> 
  <li>All necessary files will be available in <code>target</code> folder, copy/move it to /opt/red5sip/</li> 
  <li>Insert proper values to the <code>/opt/red5sip/settings.properties</code> 
   <div class="source"> 
    <pre>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-&gt;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)

</pre> 
   </div> </li> 
  <li>Set correct permissions on red5sip files: 
   <div class="source"> 
    <pre>sudo chown -R nobody:nogroup /opt/red5sip

</pre> 
   </div> </li> 
  <li>Add red5sip to autostart: 
   <div class="source"> 
    <pre>sudo cp /opt/red5sip/red5sip /etc/init.d/
sudo chmod a+x /etc/init.d/red5sip
sudo update-rc.d red5sip defaults

</pre> 
   </div> </li> 
  <li>Start openmeetings 
   <div class="source"> 
    <pre>service red5 start

</pre> 
   </div> </li> 
  <li> Enable <code>SIP</code> in openmeetings: <br> <code>Administration-&gt;Configuration-&gt;red5sip.enable == yes</code> </li> 
  <li> Enable SIP for particular room(s): <br> <code>Administration-&gt;Conference rooms-&gt;Room-&gt;Enable SIP transport in the room == checked</code><br> (SIP number will be assigned to room if everything is OK) </li> 
  <li>Start red5sip 
   <div class="source"> 
    <pre>service red5sip start

</pre> 
   </div> </li> 
 </ul> 
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        <a class="nav-link" href="#voip-and-sip-integration" title="VoIP and SIP Integration">VoIP and SIP Integration</a>
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        <a class="nav-link" href="#sip-transport-integration" title="SIP-Transport Integration">SIP-Transport Integration</a>
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