asterisk18-script

Script to install Asterisk 18 with Pjsip and WEBRTC on CentOS 7

This script :

installs Asterisk 18 with WEBRTC enabled
installs and configures Bind9
Installs and configures apache web server with ssl
will change your SSH port from 22 to your desired port
will enable and configure firewall
will create 2 sip users 100 and 200
installs webphone from @InnovateAsterisk/Browser-Phone

******** Note1: This Script should be installed on a fresh Installation ******** Note2: in case the script exits, there is something wrong with your configuration. unless you see the message INSTALLATION DONE ******** Note3: Please replace yourdomain.com with your own domain. ******** Note4: Before you run this script your domain registrar account set your DNS to ns1.yourdomain.com and ns2.yourdomain.com and allow up to 24 hours for your domain DNS to propagate. ******** Note5: This script should be run with options.

-d yourdomain.com -i your static IP address -p your desired ssh port -e your email address for lets encyprt -t your server timezone

INSTALLTION:

Install a CentOS 7
confirgure your netwrok with static IP address
run the following commands:

sudo yum update -y sudo yum install net-tools npm svn vim curl git telnet wget nano epel-release -y sudo sed -i s/^SELINUX=.*$/SELINUX=disabled/ /etc/selinux/config sudo hostnamectl set-hostname voip.yourdomain.com --static sudo reboot

once your server is back, please download asterisk-webrtc.sh script and run it as follow:

wget https://github.com/mehdizeus/asterisk18-script/blob/main/asterisk18-webrtc.sh chmod 777 asterisk18-webrtc.sh bash -e asterisk18-webrtc.sh -d example.com -i x.x.x.x -p 2449 -e email@email.com -t America/New_York

during the installtion you will see a prompt like below picture. navigate to "Codec Translators" on the left menu and select coded_opus asterisk18-webrtc.png

you can also navigate to "Extra sound package" on the left side and select your prefered sound packages

one the installtion is finished: vim /etc/asterisk/extensions.conf

and insert these lines at the end of the file [speak-exte-nnum] exten => 888,1,NoOp(FEATURE: SPEAK MY EXTENSION NUMBER) same => n,Answer same => n,Wait(1) same => n,Playback(extension) same => n,Wait(1) same => n,SayDigits(${CALLERID(num)}) same => n,Wait(2) same => n,Hangup() ;END of [speak-exte-nnum]

[dial-extension] exten => s,1,NoOp(Calling: ${ARG1}) exten => s,n,Set(JITTERBUFFER(adaptive)=default) exten => s,n,Dial(PJSIP/${ARG1},30) exten => s,n,Hangup()

exten => e,1,Hangup()

[send-text] exten => s,1,NoOp(Sending Text To: ${ARG1}) exten => s,n,Set(PEER=${CUT(CUT(CUT(MESSAGE(from),@,1),<,2),:,2)}) exten => s,n,Set(FROM=${SHELL(asterisk -rx ‘pjsip show endpoint ${PEER}’ | grep ‘callerid ‘ | cut -d’:’ -f2- | sed ‘s/^ *//’ | tr -d ‘‘)}) exten => s,n,Set(CALLERID_NUM=${CUT(CUT(FROM,>,1),<,2)}) exten => s,n,Set(FROM_SIP=${STRREPLACE(MESSAGE(from), exten => s,n,MessageSend(pjsip:${ARG1},${FROM_SIP}) exten => s,n,Hangup()

then in terminal type asterisk -rx "reload"

******** WARNING: the password in pjsip.conf is 1234 and you should change it to a more secure password in a production server

THATS IT!!

Now visit to yourdomain.com and start configuring webphone navigate to webphone settings > account : Asterisk server Address: yourdomain.com Websocket Port: 8089 Websocket Path: /ws Subscribe Extension: 100 Full Name: 100 SIP username: 100 SIP password: 1234 and hit Save

ENJOY!!!

linuxacademyir/asterisk18-script

asterisk18-script