vaheeD khoshnouD

First, we need to install some dependency package on CentOS7

yum install epel-release
yum insall certbot git -y 

Next Step we need to use valid SSL ( try LetsEncrypt )

certbot certonly --standalone -d pbx.example.com
groupadd sslcerts
chgrp sslcerts /etc/letsencrypt/
usermod -a -G sslcerts asterisk

need to edit some asterisk config and reload

vi /etc/asterisk/rtp_custom.conf

and put this config on file

[general]
rtpstart=10000
rtpend=20000
rtpchecksums=no
dtmftimeout=3000
rtcpinterval=5000
strictrtp=no
icesupport=yes
stunaddr=stun.l.google.com:19302
[ice_host_candidates]
xx.xx.xx.xx (local ip) => xx.xx.xx.xx (public ip)

Save and go next…

vi /etc/asterisk/sip_custom.conf

and put this config on file

websocket_enabled=true
maxcallbitrate=5120
accept_outofcall_message=yes
auth_message_requests=no
outofcall_message_context=messages

vi /etc/asterisk/extention_cusrom.conf

and put this config on the end of line

[messages]
exten => _.,1,NoOp(SMS receiving dialplan invoked)
exten => _.,n,NoOp(To ${MESSAGE(to)})
exten => _.,n,NoOp(From ${MESSAGE(from)})
exten => _.,n,NoOp(Body ${MESSAGE(body)})
exten => _.,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)})
exten => _.,n,MessageSend(${ACTUALTO},${MESSAGE(from)})
exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
exten => _.,n,GotoIf($["${MESSAGE_SEND_STATUS}" != "SUCCESS"]?sendfailedmsg)
exten => _.,n,Hangup()
;exten => h,1,Hangup()
;
; Handle failed messaging
exten => _.,n(sendfailedmsg),Set(MESSAGE(body)="[${STRFTIME(${EPOCH},,
%d%m%Y-%H:%M:%S)}] Your message to ${EXTEN} has failed. Retry later.")
exten => _.,n,Set(ME_1=${CUT(MESSAGE(from),<,2)})
exten => _.,n,Set(ACTUALFROM=${CUT(ME_1,@,1)})
exten => _.,n,MessageSend(${ACTUALFROM},ServiceCenter)
exten => _.,n,Hangup()
exten => h,1,Hangup()

Install BrowserPhone Project ( very likely to me to use this, but you can use any WebRTC Phone you want )

cd /tmp
git clone https://github.com/InnovateAsterisk/Browser-Phone.git
cp -rfv Browser-Phone/Phone /var/www/html/Phone

better to run this command :)

chown -R asterisk:asterisk /etc/asterisk/
chown -R asterisk:asterisk /var/www/html/Phone
amportal restart

Now Open Web Interface IssabelPBX and PBX Configuration and Advanced Settings

Go to Asterisk Builtin mini-HTTP server Setting and change blow information

Certificate file: /etc/letsencrypt/live/pbx.example.com/fullchain.pem
Enable HTTPS support for the mini-HTTP Server: True
Enable Static Content: False
Enable the mini-HTTP Server: False
HTTP Bind Address: 0.0.0.0
HTTP Bind Port: 8088
HTTPS Bind Address/Port: 0.0.0.0:8089
Private key file: /etc/letsencrypt/live/pbx.example.com/privkey.pem

Next, Go to SIP Asterisk Setting and check this configuration

NAT Settings    Yes
External IP	0.0.0.0

Audio Codecs
Codecs: opus,ulaw,alaw,gsm

Video Support   Enabled
Codecs:  vp8
Max Bit Rate 5120

After save and Setting go to Extensions and create a SIP WebRTC Extention

All Config Default and need one change

enable rtcp_mux

Save and Submit change.

Congratulation, you can now use this amazing project :)

https://pbx.example.com/Phone
Asterisk Server: pbx.example.com
WebSocket Port: 8089
WebsocketPath: /ws
and put your SIP Extention and password

Router and firewall configuration as you need for that

Please accept and forward in firewall and router

TCP 8089
TCP 443
UDP 10000-20000

Any Questions or problems?