e3fee2cf46
Prior to this commit, the Odoo documentation was mainly split between
two repositories: odoo/odoo/doc and odoo/documentation-user. Some bits
of documentation were also hosted elsewhere (e.g., wiki, upgrade, ...).
This was causing several problems among which:
- The theme, config, Makefile, and similar technical resources had to
be duplicated. This resulted in inconsistent layout, features, and
build environments from one documentation to another.
- Some pages did not fit either documentation as they were relevant
for both users and developers. Some were relevant to neither of the
two (e.g., DB management).
- Cross-doc references had to be absolute links and they broke often.
- Merging large image files in the developer documentation would bloat
the odoo/odoo repository. Some contributions had to be lightened to
avoid merging too many images (e.g., Odoo development tutorials).
- Long-time contributors to the user documentation were chilly about
going through the merging process of the developer documentation
because of the runbot, mergebot, `odoo-dev` repository, etc.
- Some contributors would look for the developer documentation in the
`odoo/documentation-user` repository.
- Community issues about the user documentation were submitted on the
`odoo/odoo` repository and vice-versa.
Merging all documentations in one repository will allow us to have one
place, one theme, one work process, and one set of tools (build
environment, ...) for all of the Odoo docs.
As this is a good opportunity to revamp the layout of the documentation,
a brand new theme replaces the old one. It features a new way to
navigate the documentation, centered on the idea of always letting the
reader know what is the context (enclosing section, child pages, page
structure ...) of the page they are reading. The previous theme would
quickly confuse readers as they navigated the documentation and followed
cross-application links.
The chance is also taken to get rid of all the technical dangling parts,
performance issues, and left-overs. Except for some page-specific JS
scripts, the Odoo theme Sphinx extension is re-written from scratch
based on the latest Sphinx release to benefit from the improvements and
ease future contributions.
task-2351938
task-2352371
task-2205684
task-2352544
Closes #945
226 lines
7.1 KiB
ReStructuredText
226 lines
7.1 KiB
ReStructuredText
============================================
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Configure your VOIP Asterisk server for Odoo
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============================================
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Installing Asterisk server
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==========================
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Dependencies
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~~~~~~~~~~~~
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Before installing Asterisk you need to install the following dependencies:
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- wget
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- gcc
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- g++
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- ncurses-devel
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- libxml2-devel
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- sqlite-devel
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- libsrtp-devel
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- libuuid-devel
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- openssl-devel
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- pkg-config
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In order to install libsrtp, follow the instructions below:
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.. code-block:: console
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cd /usr/local/src/
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wget http://srtp.sourceforge.net/srtp-1.4.2.tgz
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tar zxvf srtp-1.4.2.tgz
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cd /usr/local/src/srtp
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./configure CFLAGS=-fPIC --prefix=/usr/local/lib
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make && make install
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You also need to install PJSIP, you can download the source `here <http://www.pjsip.org/download.htm>`_. Once the source directory is extracted:
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- **Change to the pjproject source directory:**
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.. code-block:: console
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# cd pjproject
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- **run:**
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.. code-block:: console
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# ./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr CFLAGS='-O2 -DNDEBUG'
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- **Build and install pjproject:**
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.. code-block:: console
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# make dep
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# make
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# make install
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- **Update shared library links:**
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.. code-block:: console
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# ldconfig
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- **Verify that pjproject is installed:**
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.. code-block:: console
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# ldconfig -p | grep pj
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- **The result should be:**
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.. code-block:: console
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libpjsua.so (libc6,x86-64) => /usr/lib/libpjsua.so
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libpjsip.so (libc6,x86-64) => /usr/lib/libpjsip.so
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libpjsip-ua.so (libc6,x86-64) => /usr/lib/libpjsip-ua.so
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libpjsip-simple.so (libc6,x86-64) => /usr/lib/libpjsip-simple.so
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libpjnath.so (libc6,x86-64) => /usr/lib/libpjnath.so
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libpjmedia.so (libc6,x86-64) => /usr/lib/libpjmedia.so
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libpjmedia-videodev.so (libc6,x86-64) => /usr/lib/libpjmedia-videodev.so
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libpjmedia-codec.so (libc6,x86-64) => /usr/lib/libpjmedia-codec.so
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libpjmedia-audiodev.so (libc6,x86-64) => /usr/lib/libpjmedia-audiodev.so
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libpjlib-util.so (libc6,x86-64) => /usr/lib/libpjlib-util.so
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libpj.so (libc6,x86-64) => /usr/lib/libpj.so
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Asterisk
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~~~~~~~~
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- 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>`_.
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- Extract Asterisk:
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.. code-block:: console
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tar zxvf asterisk*
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- Enter the Asterisk directory:
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.. code-block:: console
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cd ./asterisk*
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- Run the Asterisk configure script:
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.. code-block:: console
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./configure --with-pjproject --with-ssl --with-srtp
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- Run the Asterisk menuselect tool:
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.. code-block:: console
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make menuselect
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- 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.
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- Compile and install Asterisk:
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.. code-block:: console
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make && make install
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- 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'.
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DTLS Certificates
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~~~~~~~~~~~~~~~~~
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- After you need to setup the DTLS certificates.
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.. code-block:: console
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mkdir /etc/asterisk/keys
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- Enter the Asterisk scripts directory:
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.. code-block:: console
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cd /asterisk*/contrib/scripts
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- Create the DTLS certificates (replace pbx.mycompany.com with your ip address or dns name, replace My Super Company with your company name):
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.. code-block:: console
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./ast_tls_cert -C pbx.mycompany.com -O "My Super Company" -d /etc/asterisk/keys
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Configure Asterisk server
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=========================
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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:
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.. code-block:: console
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;http.conf
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[general]
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enabled=yes
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bindaddr=127.0.0.1 ; Replace this with your IP address
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bindport=8088 ; Replace this with the port you want to listen on
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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:
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.. code-block:: console
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;sip.conf
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[general]
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realm=127.0.0.1 ; Replace this with your IP address
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udpbindaddr=127.0.0.1 ; Replace this with your IP address
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transport=udp
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[1060] ; This will be WebRTC client
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type=friend
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username=1060 ; The Auth user for SIP.js
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host=dynamic ; Allows any host to register
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secret=password ; The SIP Password for SIP.js
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encryption=yes ; Tell Asterisk to use encryption for this peer
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avpf=yes ; Tell Asterisk to use AVPF for this peer
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icesupport=yes ; Tell Asterisk to use ICE for this peer
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context=default ; Tell Asterisk which context to use when this peer is dialing
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directmedia=no ; Asterisk will relay media for this peer
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transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets
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force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
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dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
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dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
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dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
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dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
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dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS
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In the sip.conf and rtp.conf files you also need to add or uncomment the lines:
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.. code-block:: console
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icesupport = true
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stunaddr = stun.l.google.com:19302
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Lastly, set up extensions.conf:
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.. code-block:: console
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;extensions.conf
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[default]
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exten => 1060,1,Dial(SIP/1060) ; Dialing 1060 will call the SIP client registered to 1060
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Configure Odoo VOIP
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===================
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In Odoo, the configuration should be done in the user's preferences.
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.. image:: media/voip_config01.png
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:align: center
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- The SIP Login/Browser's Extension is the number you configured previously in the sip.conf file (in our example: 1060).
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- The SIP Password is the secret you chose in the sip.conf file.
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- 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.
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The configuration should also be done in the General Settings under the "Integrations" section.
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.. image:: media/onsip02.png
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:align: center
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- The PBX Server IP should be the same as the IP you define in the http.conf file.
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- 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.
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