[RMV]VOIP:Scrub Asterisk Docs
closes odoo/documentation#7443
X-original-commit: 055e332ef3
Signed-off-by: Zachary Straub (zst) <zst@odoo.com>
Signed-off-by: Timothy Kukulka (tiku) <tiku@odoo.com>
This commit is contained in:
tiku-odoo
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439c4521ad
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@ -7,6 +7,5 @@ VoIP (Voice over Internet Protocol)
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.. toctree::
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.. toctree::
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:titlesonly:
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:titlesonly:
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voip/asterisk
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voip/onsip
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voip/onsip
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voip/axivox
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voip/axivox
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@ -1,238 +0,0 @@
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============================================
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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
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<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
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<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
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3 x’s next to res_srtp, there is a problem with the srtp library and you must reinstall it. Save
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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
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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
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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
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do not flow down into the peer settings very well. By default, Asterisk config files are located in
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/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
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most cases, directmedia should be disabled. Also under the WebRTC client, the transport needs to be
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listed as ‘ws’ to allow websocket connections. All of these config lines should be under the peer
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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:: asterisk/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
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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
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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:: onsip/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
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defined previously and "XXXX" needs to be the port defined in the http.conf file.
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Before Width: | Height: | Size: 64 KiB |
@ -6,8 +6,7 @@ Introduction
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============
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Odoo VoIP can be set up to work together with `Axivox <https://www.axivox.com/>`_. In that case, an
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Odoo VoIP can be set up to work together with `Axivox <https://www.axivox.com/>`_. In that case, an
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:doc:`Asterisk server <asterisk>` is not necessary as the infrastructure is hosted and managed by
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Asterisk server is not necessary as the infrastructure, as it is hosted and managed by Axivox.
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Axivox.
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To use this service, `contact Axivox <https://www.axivox.com/contact/>`_ to open an account. Before
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To use this service, `contact Axivox <https://www.axivox.com/contact/>`_ to open an account. Before
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doing so, verify that Axivox covers your area and the areas you wish to call.
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doing so, verify that Axivox covers your area and the areas you wish to call.
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