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<div class="section">
<h2><a name="SIP-Transport_Integration"></a>SIP-Transport Integration</h2>
<p>You need Apache OpenMeetings <b>version 3.0+</b> to apply this guide!</p>
<p>You need Asterisk <b>version 13+</b> to apply this guide!</p>
<p>Here is instruction how-to set up red5sip transport integration with OpenMeetings on Ubuntu 12.10.</p>
</div>
<div class="section">
<h2><a name="Prerequisites"></a>Prerequisites</h2>
<div>
Run the commands
<div class="source"><pre class="prettyprint">
sudo apt-get update &amp;&amp; sudo apt-get upgrade
</pre></div>
</div>
</div>
<div class="section">
<h2><a name="Building_and_setting_up_Asterisk"></a>Building and setting up Asterisk</h2>
<div>
Run the commands
<div class="source"><pre class="prettyprint">
sudo mkdir /usr/src/asterisk &amp;amp;&amp;amp; cd /usr/src/asterisk
sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-13.9.0.tar.gz
sudo tar -xvzf asterisk-13.9.0.tar.gz
cd ./asterisk-13.9.0
sudo make clean
sudo contrib/scripts/install_prereq install
sudo ./configure
sudo make menuconfig
</pre></div>
Make sure you have selected <tt>Add-ons -&gt; res_config_mysql</tt>, Press F12 to save
<div class="source"><pre class="prettyprint">
sudo make
sudo make install
sudo make samples
sudo make config
sudo service asterisk start
</pre></div>
</div>
</div>
<div class="section">
<h2><a name="Configure_Asterisk"></a>Configure Asterisk</h2>
<div>
Enable asterisk MySQL module:<br /><br />
Modify &quot;[modules]&quot; section of <tt>/etc/asterisk/modules.conf</tt> as follows:<br />
<b>Add/uncomment the following lines</b>
<div class="source"><pre class="prettyprint">
preload =&gt; res_config_mysql.so
</pre></div>
</div><br />
<div>
Configure MySQL module:<br /><br />
Set valid data for MySQL in <tt>/etc/asterisk/res_config_mysql.conf</tt> :<br />
<b>Example</b>
<div class="source"><pre class="prettyprint">
[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 <tt>/etc/asterisk/sip.conf</tt><br />
<b>Add/uncomment the following line</b>:<br />
<div class="source"><pre class="prettyprint">
videosupport=yes
rtcachefriends=yes
</pre></div>
<b>Increase maxexpiry value to 43200</b>:<br />
<div class="source"><pre class="prettyprint">
maxexpiry=43200
</pre></div>
<b>Add user for the &quot;SIP Transport&quot;</b>:<br />
<div class="source"><pre class="prettyprint">
[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 <tt>/etc/asterisk/extconfig.conf</tt>:
<div class="source"><pre class="prettyprint">
[settings]
sippeers =&gt; mysql,general,sipusers
</pre></div>
</div><br />
<div>
Modify <tt>/etc/asterisk/extensions.conf</tt><br />
<b>Add the following section</b>:<br />
<div class="source"><pre class="prettyprint">
; *****************************************************
; 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 &#x2013;rx &#x201c;database show&#x201d;
; If you do not receive an output with that resembles openmeetings/rooms/400## where &#x201c;##&#x201d; will equal the extension assigned when you created your room
; If you do not receive the above output check your parameters in /opt/red5/webapps/openmeetings/WEB-INF/classes/openmeetings-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 <tt>/etc/asterisk/confbridge.conf</tt><br />
<b>Add/Modify the following secions</b>:<br />
<div class="source"><pre class="prettyprint">
[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 <tt>/etc/asterisk/manager.conf</tt><br />
<b>Add/Modify the following sections</b>:<br />
<div class="source"><pre class="prettyprint">
[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 creadentials for Asterisk manager.
Modify <tt>/opt/red5/webapps/openmeetings/WEB-INF/classes/openmeetings-applicationContext.xml</tt><br />
find <b>&lt;bean id=&quot;sipDao&quot; class=&quot;org.apache.openmeetings.db.dao.room.SipDao&quot;&gt;</b>
uncomment its parameters and set it to your custom values.
<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 class="prettyprint">
service asterisk restart
</pre></div>
</div><br />
</div>
<div class="section">
<h2><a name="Setup_red5sip_transport"></a>Setup red5sip transport</h2>
<div>
Download red5sip from <tt>https://github.com/openmeetings/red5sip</tt>
</div>
<div>
<div class="source"><pre class="prettyprint">
git clone https://github.com/openmeetings/red5sip.git
</pre></div>
</div>
<div>
Build with Apache Maven
<div class="source"><pre class="prettyprint">
mvn clean package
</pre></div>
All necessary files will be available in <tt>target</tt> folder, copy/move it to /opt/red5sip/
</div>
<div>
Insert proper values to the <tt>/opt/red5sip/settings.properties</tt>
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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 settings in public/config.xml
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
rooms=1 # TBD (not in use)
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Add red5sip to autostart:
<div class="source"><pre class="prettyprint">
sudo cp /opt/red5sip/red5sip /etc/init.d/
sudo chmod a+x /etc/init.d/red5sip
sudo update-rc.d red5sip defaults
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Start openmeetings
<div class="source"><pre class="prettyprint">
service red5 start
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Start red5sip
<div class="source"><pre class="prettyprint">
service red5sip start
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