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| <h2 id="not-implemented">Not implemented</h2> |
| </div> |
| <div class="bd-callout bd-callout-danger"> |
| Please NOTE: this functionality is not yet implemented in 5.0.x |
| </div> |
| </section> |
| <section> |
| <h2 id="voip-and-sip-integration">VoIP and SIP Integration</h2> |
| <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. <br> <br> 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. <br> <br> 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>. <br><br> </p> |
| </section> |
| <section> |
| <h2 id="sip-transport-integration">SIP-Transport Integration</h2> |
| <p>You need Apache OpenMeetings <b>version 4.0+</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 16.04.</p> |
| </section> |
| <section> |
| <h2 id="prerequisites1">Prerequisites</h2> |
| <div> |
| Run the commands |
| <div class="source"> |
| <pre>sudo apt update && sudo apt upgrade |
| |
| </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 && 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 -> 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 => 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 => 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 => _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 |
| |
| </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><bean id="sipDao" class="org.apache.openmeetings.db.dao.room.SipDao"></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->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->Configuration->red5sip.enable == yes</code> </li> |
| <li> Enable SIP for particular room(s): <br> <code>Administration->Conference rooms->Room->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> |
| </section> </main> |
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| <li class="h2"> |
| <a class="nav-link" href="#not-implemented" title="Not implemented">Not implemented</a> |
| </li> |
| <li class="h2"> |
| <a class="nav-link" href="#voip-and-sip-integration" title="VoIP and SIP Integration">VoIP and SIP Integration</a> |
| </li> |
| <li class="h2"> |
| <a class="nav-link" href="#sip-transport-integration" title="SIP-Transport Integration">SIP-Transport Integration</a> |
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| <a class="nav-link" href="#setup-red5sip-transport" title="Setup red5sip transport">Setup red5sip transport</a> |
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