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| <div class="page-header"> |
| <h1 id="voip-and-sip-integration">VoIP and SIP Integration</h1> |
| </div> |
| <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> |
| <h1 id="asterisk-integration">Asterisk Integration</h1> |
| <p>You need Apache OpenMeetings <b>version 6.0+</b> to apply this guide!</p> |
| <p>You need Asterisk <b>version 16+</b> to apply this guide!</p> |
| <p>Here is the instruction how-to set up integration between OpenMeetings and Asterisk on Ubuntu 18.04.</p> |
| <section> |
| <h2 id="prerequisites1">Prerequisites</h2> |
| <div class="source"> |
| <pre><code>sudo apt update && sudo apt upgrade |
| </code></pre> |
| </div> |
| </section> |
| <section> |
| <h2 id="building-and-setting-up-asterisk">Building and setting up Asterisk</h2> |
| <div class="source"> |
| <pre><code>sudo apt install build-essential |
| |
| sudo mkdir /usr/src/asterisk && cd /usr/src/asterisk |
| sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-16.13.0.tar.gz |
| sudo tar -xvzf asterisk-16.13.0.tar.gz |
| cd ./asterisk-16.13.0 |
| sudo make clean |
| sudo contrib/scripts/install_prereq install |
| sudo ./configure |
| sudo make menuconfig |
| </code></pre> |
| </div> |
| <p>Make sure you have selected (<a class="externalLink" href="https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients">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> |
| <div class="source"> |
| <pre><code>sudo make |
| sudo make install |
| sudo make samples |
| sudo make config |
| |
| </code></pre> |
| </div> |
| </section> |
| <section> |
| <h2 id="configure-asterisk">Configure Asterisk</h2> |
| <section> |
| <h3 id="enable-asterisk-mysql-module">Enable asterisk MySQL module:</h3> |
| <p>Modify <code>[modules]</code> section of <code>/etc/asterisk/modules.conf</code> <b>Add/uncomment the following lines</b></p> |
| <div class="source"> |
| <pre><code>preload => res_config_mysql.so |
| </code></pre> |
| </div> |
| </section> |
| <section> |
| <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><b>Example</b></p> |
| <div class="source"> |
| <pre><code>[general] |
| dbhost = 127.0.0.1 |
| dbname = openmeetings |
| dbuser = root |
| dbpass = |
| dbport = 3306 |
| dbsock = /var/lib/mysql/mysql.sock |
| dbcharset = utf8 |
| requirements=warn |
| </code></pre> |
| </div> |
| </section> |
| <section> |
| <h3 id="configure-sip-module">Configure SIP module:</h3> |
| <p>Modify <code>/etc/asterisk/sip.conf</code></p> |
| <p><b>Add/uncomment the following lines</b></p> |
| <div class="source"> |
| <pre><code>videosupport=yes |
| rtcachefriends=yes |
| </code></pre> |
| </div> |
| <p><b>Increase maxexpiry value to 43200</b></p> |
| <div class="source"> |
| <pre><code>maxexpiry=43200 |
| </code></pre> |
| </div> |
| <p><b>Add user for the “SIP Transport”</b></p> |
| <div class="source"> |
| <pre><code>[omsip_user] |
| host=dynamic |
| secret=12345 |
| context=rooms-omsip |
| transport=ws,wss |
| type=friend |
| encryption=no |
| avpf=yes |
| icesupport=yes |
| directmedia=no |
| allow=!all,ulaw,opus,vp8 |
| </code></pre> |
| </div> |
| </section> |
| <section> |
| <h3 id="configure-extensions">Configure extensions:</h3> |
| <p>Add next lines into the <code>/etc/asterisk/extconfig.conf</code>:</p> |
| <div class="source"> |
| <pre><code>[settings] |
| sippeers => mysql,general,sipusers |
| </code></pre> |
| </div> |
| <p>Modify <code>/etc/asterisk/extensions.conf</code> <b>Add the following section:</b></p> |
| <div class="source"> |
| <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 –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/${EXTEN})}]?ok:notavail) |
| exten => _400X!,n(ok),SET(PIN=${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-omsip] |
| exten => _400X!,1,GotoIf($[${DB_EXISTS(openmeetings/rooms/${EXTEN})}]?ok:notavail) |
| exten => _400X!,n(ok),Confbridge(${EXTEN},default_bridge,omsip_user) |
| exten => _400X!,n(notavail),Hangup |
| </code></pre> |
| </div> |
| </section> |
| <section> |
| <h3 id="configure-confbridge">Configure Confbridge</h3> |
| <p>Modify <code>/etc/asterisk/confbridge.conf</code></p> |
| <p><b>Add/Modify the following sections:</b></p> |
| <div class="source"> |
| <pre><code>[general] |
| |
| [omsip_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 |
| </code></pre> |
| </div> |
| </section> |
| <section> |
| <h3 id="configure-asterisk-manager">Configure Asterisk Manager</h3> |
| <p>To enable Asterisk Manager API modify <code>/etc/asterisk/manager.conf</code></p> |
| <p><b>Add/Modify the following sections:</b></p> |
| <div class="source"> |
| <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> |
| </div> |
| <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 <b>BEFORE</b> system install/restore otherwise all SIP related room information will be lost </p> |
| </section> |
| <section> |
| <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> |
| <div class="source"> |
| <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> |
| </div> |
| </section> |
| <section> |
| <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> |
| <div class="source"> |
| <pre><code>[transport-wss] |
| type=transport |
| protocol=wss |
| bind=0.0.0.0 |
| ; All other transport parameters are ignored for wss transports. |
| </code></pre> |
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