Ubuntu 14.04 Asterisk server installation

gistfile1.txt

https://www.odoo.com/apps/modules/8.0/crm_voip/

# Install asterisk server on Ubuntu 14.04

# 1. Install dependencies 
sudo apt-get update
sudo apt-get install wget
sudo apt-get install gcc
sudo apt-get install g++
sudo apt-get install ncurses-dev
sudo apt-get install libxml2-dev
sudo apt-get install libsqlite3-dev
sudo apt-get install libsrtp-dev
sudo apt-get install uuid-dev
sudo apt-get install libssl-dev
sudo apt-get install libjansson-dev

# 2. Install PJSIP
sudo wget http://www.pjsip.org/release/2.4.5/pjproject-2.4.5.tar.bz2
sudo tar -xjvf pjproject-2.4.5.tar.bz2
cd pjproject-2.4.5
sudo ./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr CFLAGS='-O2 -DNDEBUG'
sudo make dep
sudo make
sudo make install
sudo ldconfig

# 3. Install Asterisk
cd ..
sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-13.7.0.tar.gz
sudo tar -zxvf asterisk-13.7.0.tar.gz
cd asterisk-13.7.0
sudo ./configure --libdir=/usr/lib64
# sudo make menuselect
# Go to 'Resource Modules'
# 'res_srtp' has to be enabled (*)
sudo make && make install
# Install sample configs with 'sudo make samples'
# Install startup script with 'sudo make config'

# 4. Setup DTLS certificates
sudo mkdir /etc/asterisk/keys
cd contrib/scripts
sudo ./ast_tls_cert -C 192.168.61.130 -O "Van Roey" -d /etc/asterisk/keys/

# 5. Configure Asterisk server
sudo nano /etc/asterisk/http.conf

# Uncomment these
# ;http.conf
# [general]
# enabled=yes
# bindaddr=127.0.0.1                  Replace by ip adres used in step 4
# bindport=8088                       Port to listen to

sudo nano /etc/asterisk/sip.conf

# Uncomment these
# ;sip.conf
# [general]
# realm=127.0.0.1                     Replace by ip adres used in step 4
# udpbindaddr=0.0.0.0                 Replace by ip adres used in step 4
# transport=udp

# Add these
# [1060]                      ; This will be WebRTC client
# type=friend
# username=1060               ; The Auth user for SIP.js
# host=dynamic                ; Allows any host to register
# secret=vanroey              ; 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

# Uncomment 
# icesupport=true

# Add
# stunaddr=stun.l.google.com:19302

sudo nano /etc/asterisk/extensions.conf

# Add under [default]
# exten => 1060,1,Dial(SIP/1060) ; Dialing 1060 will call the SIP client registered to 1060