[MOV] general: moving VoIP docs from "CRM" to "general"

- moving VoIP docs from "CRM" to "general"
- renaming "setup.rst" as "asterisk.rst" which is more accurate
- redirections:

  * crm/optimize/onsip --> general/voip/onsip
  * crm/optimize/setup --> general/voip/asterisk

general/voip.rst

@@ -0,0 +1,10 @@
+===================================
+VoIP (Voice over Internet Protocol)
+===================================
+
+.. toctree::
+   :titlesonly:
+
+   voip/asterisk
+   voip/onsip
+   voip/asterisk

general/voip/asterisk.rst

@@ -0,0 +1,225 @@
+============================================
+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 <http://www.pjsip.org/download.htm>`_. 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 <http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-13.7.0.tar.gz>`_.
+
+- 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:: media/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:: media/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.
+

general/voip/onsip.rst

@@ -0,0 +1,79 @@
+====================================
+Use VOIP services in Odoo with OnSIP
+====================================
+
+Introduction
+============
+
+Odoo VoIP can be set up to work together with OnSIP (www.onsip.com). In that case, the installation and setup of an Asterisk server is not necessary as the whole infrastructure is hosted and managed by OnSIP.
+
+You will need to open an account with OnSIP to use this service. Before doing so, make sure that your area and the areas you wish to call are covered by the service. After opening an OnSIP account, follow the configuration procedure below.
+
+Configuration
+=============
+
+Go to Apps and install the module **VoIP OnSIP**.
+
+.. image:: media/onsip01.png
+  :align: center
+
+Go to Settings/General Settings. In the section Integrations/Asterisk (VoIP), fill in the 3 fields:
+
+- **OnSIP Domain** is the domain you chose when creating an account on www.onsip.com. If you don't know it, log in to https://admin.onsip.com/ and you will see it in the top right corner of the screen.
+- **WebSocket** should contain wss://edge.sip.onsip.com
+- **Mode** should be Production
+
+.. image:: media/onsip02.png
+  :align: center
+
+Go to **Settings/Users**. In the form view of each VoIP user, in the Preferences tab, fill in the section **PBX Configuration**:
+
+- **SIP Login / Browser's Extension**: the OnSIP 'Username'
+- **OnSIP authorization User**: the OnSIP 'Auth Username'
+- **SIP Password**: the OnSIP 'SIP Password'
+- **Handset Extension**: the OnSIP 'Extension'
+
+You can find all this information by logging in at https://admin.onsip.com/users, then select the user you want to configure and refer to the fields as pictured below.
+
+.. image:: media/onsip03.png
+  :align: center
+
+You can now make phone calls by clicking the phone icon in the top right corner of Odoo (make sure you are logged in as a user properly configured in Odoo and in OnSIP).
+
+.. note::
+
+        If you see a *Missing Parameters* message in the Odoo softphone, make sure to refresh your Odoo window and try again.
+
+        .. image:: media/onsip04.png
+          :align: center
+
+.. note::
+
+        If you see an *Incorrect Number* message in the Odoo softphone, make sure to use the international format, leading with the plus (+) sign followed by the international country code. E.g.: +16506913277 (where +1 is the international prefix for the United States).
+
+        .. image:: media/onsip05.png
+          :align: center
+
+You can now also receive phone calls. Your number is the one provided by OnSIP. Odoo will ring and display a notification.
+
+.. image:: media/onsip06.png
+  :align: center
+
+OnSIP on Your Cell Phone
+========================
+
+In order to make and receive phone calls when you are not in front of your computer, you can use a softphone app on your cell phone in parallel of Odoo VoIP. This is useful for on-the-go calls, but also to make sure to hear incoming calls, or simply for convenience. Any SIP softphone will work.
+
+On Android and iOS, OnSIP has been successfully tested with `Grandstream Wave <https://play.google.com/store/apps/details?id=com.grandstream.wave>`_. When creating an account, select OnSIP in the list of carriers. You will then have to configure it as follows:
+
+- **Account name**: OnSIP
+- **SIP Server**: the OnSIP 'Domain'
+- **SIP User ID**: the OnSIP 'Username'
+- **SIP Authentication ID**: the OnSIP 'Auth Username'
+- **Password**: the OnSIP 'SIP Password'
+
+Aside from initiating calls from Grandstream Wave on your phone, you can also initiate calls by clicking phone numbers in your browser on your PC. This will make Grandstream Wave ring and route the call via your phone to the other party. This approach is useful to avoid wasting time dialing phone numbers. In order to do so, you will need the Chrome extension `OnSIP Call Assistant <https://chrome.google.com/webstore/detail/onsip-call-assistant/pceelmncccldedfkcgjkpemakjbapnpg?hl=en>`_.
+
+.. warning::
+
+        The downside of using a softphone on your cell phone is that your calls will not be logged in Odoo as the softphone acts as an independent separate app.