- 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>`_.
- 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.
- 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'.
- 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.