diff --git a/content/applications/general/voip.rst b/content/applications/general/voip.rst index 1a354d096..c3d0399d7 100644 --- a/content/applications/general/voip.rst +++ b/content/applications/general/voip.rst @@ -7,6 +7,5 @@ VoIP (Voice over Internet Protocol) .. toctree:: :titlesonly: - voip/asterisk voip/onsip voip/axivox diff --git a/content/applications/general/voip/asterisk.rst b/content/applications/general/voip/asterisk.rst deleted file mode 100644 index 7fbe2b77b..000000000 --- a/content/applications/general/voip/asterisk.rst +++ /dev/null @@ -1,238 +0,0 @@ -============================================ -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. - - -.. image:: asterisk/voip_config01.png - :align: center - -- 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 General Settings under the "Integrations" section. - -.. image:: onsip/onsip02.png - :align: center - -- The PBX Server IP should be the same as the IP you define in the http.conf file. - -- The WebSocket should be: ws://localhost:XXXX/ws where "localhost" needs to be the same as the IP - defined previously and "XXXX" needs to be the port defined in the http.conf file. diff --git a/content/applications/general/voip/asterisk/voip_config01.png b/content/applications/general/voip/asterisk/voip_config01.png deleted file mode 100644 index 09cf4b85d..000000000 Binary files a/content/applications/general/voip/asterisk/voip_config01.png and /dev/null differ diff --git a/content/applications/general/voip/axivox.rst b/content/applications/general/voip/axivox.rst index 8ccc52b05..0021d7013 100644 --- a/content/applications/general/voip/axivox.rst +++ b/content/applications/general/voip/axivox.rst @@ -6,8 +6,7 @@ Introduction ============ Odoo VoIP can be set up to work together with `Axivox `_. In that case, an -:doc:`Asterisk server ` is not necessary as the infrastructure is hosted and managed by -Axivox. +Asterisk server is not necessary as the infrastructure, as it is hosted and managed by Axivox. To use this service, `contact Axivox `_ to open an account. Before doing so, verify that Axivox covers your area and the areas you wish to call.