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Organize the pipeline https://www.odoo.com/documentation/user/11.0/discuss/plan_activities.html (clone it here) Manage lost opportunities -> https://docs.google.com/document/d/1Uxr5L0tmNTv4cUiq948_8KgHMAgtB--Ib-h9K_XjYQM/edit Manage multiple sales teams -> https://docs.google.com/document/d/1xbQzJozfPyGJxtu1dlYaBvmtU4aZkkGnRT3jmdzp84s/edit Acquire leads Convert leads into opportunities -> https://docs.google.com/document/d/15huZHD7NLqUC-XyogT1cR_oBof-DlC6U-olfxJLuGfY/edit Generate leads from emails -> https://docs.google.com/document/d/1qt5LAyb7jjArZqFQbwHANKhkE1uifVuc0I2pnf9eF7k/edit Generate leads from your website contact page -> https://docs.google.com/document/d/1shcyTp4gMuekRMIbJqbHfFIIk10N0GHtHIQlPXuax-Q/edit Send quotations -> https://docs.google.com/document/d/1wYfImnN1XxxaPQMjTvvIDgMkhXE5ud1D-9pbkr0khsA/edit Assign and track leads Track your prospect visits -> https://docs.google.com/document/d/1wSoV9wgI6HwD7pogTbPuapIfxCgQ5fUWhZ3frU_iOTk/edit Assign leads automatically based on scoring -> https://docs.google.com/document/d/1cBc4XOTAnSjjfg3Q-BS14dB2lvv09NaUCEpIlMKyp6U/edit Analyze performance Check your Win/Loss ratio -> https://docs.google.com/document/d/1_-I50VbyWwSrp63pRYq7_Lo7QSr0b-0ru4FLQZv3C2k/edit Get an accurate probable turnover -> https://docs.google.com/document/d/1tfCntvywMehg9xVF5NsjA6-6m7QQ-FryfwGkPAOJPNE/edit Optimize your Day-to-Day work Synchronize Google Calendar with Odoo -> https://www.odoo.com/documentation/user/11.0/crm/calendar/google_calendar_credentials.html (Change name) Use VOIP services in Odoo with OnSIP -> https://www.odoo.com/documentation/user/11.0/crm/leads/voip/onsip.html (Change name) Configure your VOIP Asterisk server for Odoo -> https://www.odoo.com/documentation/user/11.0/crm/leads/voip/setup.html (Change name)
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217 lines
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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 <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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The SIP Login/Browser's Extension is the number you configured previously in the
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sip.conf file. In our example, 1060. The SIP Password is the secret you chose in
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the sip.conf file. The extension of your office's phone is not a required field
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but it is used if you want to transfer your call from Odoo to an external phone
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also configured in the sip.conf file.
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The configuration should also be done in the sale settings under the title "PBX
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Configuration". You need to put the IP you define in the http.conf file and the
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WebSocket should be: ws://127.0.0.1:8088/ws. The part "127.0.0.1" needs to be
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the same as the IP defined previously and the "8088" is the port you defined in
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the http.conf file.
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