============================================ Configure your VOIP Asterisk server for Odoo ============================================ Installing Asterisk server ========================== Dependencies ~~~~~~~~~~~~ Before installing Asterisk you need to install the following dependencies: - wget - gcc - g++ - ncurses-devel - libxml2-devel - sqlite-devel - libsrtp-devel - libuuid-devel - openssl-devel - pkg-config In order to install libsrtp, follow the instructions below: .. code-block:: console cd /usr/local/src/ wget http://srtp.sourceforge.net/srtp-1.4.2.tgz tar zxvf srtp-1.4.2.tgz cd /usr/local/src/srtp ./configure CFLAGS=-fPIC --prefix=/usr/local/lib make && make install You also need to install PJSIP, you can download the source `here `_. Once the source directory is extracted: - **Change to the pjproject source directory:** .. code-block:: console # cd pjproject - **run:** .. code-block:: console # ./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr CFLAGS='-O2 -DNDEBUG' - **Build and install pjproject:** .. code-block:: console # make dep # make # make install - **Update shared library links:** .. code-block:: console # ldconfig - **Verify that pjproject is installed:** .. code-block:: console # ldconfig -p | grep pj - **The result should be:** .. code-block:: console libpjsua.so (libc6,x86-64) => /usr/lib/libpjsua.so libpjsip.so (libc6,x86-64) => /usr/lib/libpjsip.so libpjsip-ua.so (libc6,x86-64) => /usr/lib/libpjsip-ua.so libpjsip-simple.so (libc6,x86-64) => /usr/lib/libpjsip-simple.so libpjnath.so (libc6,x86-64) => /usr/lib/libpjnath.so libpjmedia.so (libc6,x86-64) => /usr/lib/libpjmedia.so libpjmedia-videodev.so (libc6,x86-64) => /usr/lib/libpjmedia-videodev.so libpjmedia-codec.so (libc6,x86-64) => /usr/lib/libpjmedia-codec.so libpjmedia-audiodev.so (libc6,x86-64) => /usr/lib/libpjmedia-audiodev.so libpjlib-util.so (libc6,x86-64) => /usr/lib/libpjlib-util.so libpj.so (libc6,x86-64) => /usr/lib/libpj.so Asterisk ~~~~~~~~ - In order to install Asterisk 13.7.0, you can download the source directly `there `_. - Extract Asterisk: .. code-block:: console tar zxvf asterisk* - Enter the Asterisk directory: .. code-block:: console cd ./asterisk* - Run the Asterisk configure script: .. code-block:: console ./configure --with-pjproject --with-ssl --with-srtp - Run the Asterisk menuselect tool: .. code-block:: console make menuselect - In the menuselect, go to the resources option and ensure that res_srtp is enabled. If there are 3 x’s next to res_srtp, there is a problem with the srtp library and you must reinstall it. Save the configuration (press x). You should also see stars in front of the res_pjsip lines. - Compile and install Asterisk: .. code-block:: console make && make install - If you need the sample configs you can run 'make samples' to install the sample configs. If you need to install the Asterisk startup script you can run 'make config'. DTLS Certificates ~~~~~~~~~~~~~~~~~ - After you need to setup the DTLS certificates. .. code-block:: console mkdir /etc/asterisk/keys - Enter the Asterisk scripts directory: .. code-block:: console cd /asterisk*/contrib/scripts - Create the DTLS certificates (replace pbx.mycompany.com with your ip address or dns name, replace My Super Company with your company name): .. code-block:: console ./ast_tls_cert -C pbx.mycompany.com -O "My Super Company" -d /etc/asterisk/keys Configure Asterisk server ========================= For WebRTC, a lot of the settings that are needed MUST be in the peer settings. The global settings do not flow down into the peer settings very well. By default, Asterisk config files are located in /etc/asterisk/. Start by editing http.conf and make sure that the following lines are uncommented: .. code-block:: console ;http.conf [general] enabled=yes bindaddr=127.0.0.1 ; Replace this with your IP address bindport=8088 ; Replace this with the port you want to listen on Next, edit sip.conf. The WebRTC peer requires encryption, avpf, and icesupport to be enabled. In most cases, directmedia should be disabled. Also under the WebRTC client, the transport needs to be listed as ‘ws’ to allow websocket connections. All of these config lines should be under the peer itself; setting these config lines globally might not work: .. code-block:: console ;sip.conf [general] realm=127.0.0.1 ; Replace this with your IP address udpbindaddr=127.0.0.1 ; Replace this with your IP address transport=udp [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=password ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF for this peer icesupport=yes ; Tell Asterisk to use ICE for this peer context=default ; Tell Asterisk which context to use when this peer is dialing directmedia=no ; Asterisk will relay media for this peer transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11 dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer dtlsverify=no ; Tell Asterisk to not verify your DTLS certs dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS In the sip.conf and rtp.conf files you also need to add or uncomment the lines: .. code-block:: console icesupport = true stunaddr = stun.l.google.com:19302 Lastly, set up extensions.conf: .. code-block:: console ;extensions.conf [default] exten => 1060,1,Dial(SIP/1060) ; Dialing 1060 will call the SIP client registered to 1060 Configure Odoo VOIP =================== In Odoo, the configuration should be done in the user's preferences. The SIP Login/Browser's Extension is the number you configured previously in the sip.conf file. In our example, 1060. The SIP Password is the secret you chose in the sip.conf file. The extension of your office's phone is not a required field but it is used if you want to transfer your call from Odoo to an external phone also configured in the sip.conf file. The configuration should also be done in the sale settings under the title "PBX Configuration". You need to put the IP you define in the http.conf file and the WebSocket should be: ws://127.0.0.1:8088/ws. The part "127.0.0.1" needs to be the same as the IP defined previously and the "8088" is the port you defined in the http.conf file.