First commit - Hello world 🦾🤖

README.md

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 # asterisk
 
+## Installing Dependencies
+Before installing Asterisk, you need to update your package lists and install the necessary dependencies. Open your terminal and execute the following commands:
+
+`sudo apt update`
+`sudo apt upgrade`
+`sudo apt install -y wget build-essential subversion`
+
+## Downloading and Installing Asterisk
+With the dependencies in place, you can now download and compile Asterisk:
+
+`cd /usr/src`
+`sudo wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz`
+`sudo tar xvf asterisk-18-current.tar.gz`
+
+Change into the extracted directory and install additional required packages:
+
+`cd asterisk-18*`
+`sudo contrib/scripts/install_prereq install`
+
+Now, proceed to compile and install Asterisk:
+
+`sudo ./configure`
+`sudo make menuselect`
+`sudo make`
+`sudo make install`
+`sudo make samples`
+`sudo make config`
+`sudo ldconfig`
+
+With Asterisk installed, you can now set up its user and group permissions:
+
+`sudo adduser --system --group --no-create-home asterisk`
+`sudo chown -R asterisk:asterisk /etc/asterisk`
+`sudo chown -R asterisk:asterisk /var/{lib,log,spool}/asterisk`
+`sudo chown -R asterisk:asterisk /usr/lib/asterisk`
+
+## Configuring Asterisk
+Editing the main configuration file is the next step. Open it using your favorite text editor:
+
+`sudo nano /etc/asterisk/asterisk.conf`
+
+Make the necessary changes according to your setup, then start and enable Asterisk at boot:
+
+`sudo systemctl start asterisk`
+`sudo systemctl enable asterisk`
+
+Ensure that Asterisk is running without issues:
+
+`sudo asterisk -vvvr`
+
+## Setting Up SIP Accounts
+To handle VoIP calls, you need to set up SIP accounts. Edit the sip.conf and extensions.conf files:
+
+`sudo nano /etc/asterisk/sip.conf`
+`sudo nano /etc/asterisk/extensions.conf`
+
+Add your SIP users in sip.conf and dial plan in extensions.conf.

extensions.conf

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+[general]
+static=yes
+writeprotect=no
+
+[from-trunk]
+; (888)88-MOENY
+exten => 18888866369,1,Answer()
+ same => n,Wait(0.5)
+ same => n,agi(googletts.agi,"Thank you for calling Moeny! No AI is available to speak right now. I'm just a Google Goog-let. But call back soon and you'll be chatting with AI so good you think it's a real person! Bye for now!",en)
+ same => n,Hangup()
+
+; (559)TALK-2-AI
+exten => 15598255224,1,Goto(s,1)
+
+exten => s,1,Answer()       
+same => n,Wait(0.5)
+same => n,Playback(hello-world&and&goodbye)
+same => n,Wait(0.5)
+;same => n,Dial(PJSIP/101)
+same => n,Hangup()
+
+[from-internal]
+exten => 1234,1,Goto(from-trunk,s,1)
+exten => 5678,1,Goto(from-trunk,18888866369,1)

pjsip.conf

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+[transport-udp]
+type=transport
+protocol=udp
+bind=0.0.0.0:5062
+
+[acl]
+type=acl
+deny=0.0.0.0/0
+permit=100.40.223.130/32
+permit=52.204.242.197/32
+
+[101]
+type=endpoint
+transport=transport-udp
+context=from-internal
+disallow=all
+allow=ulaw
+auth=101
+aors=101
+direct_media=no
+rtp_symmetric=yes
+force_rport=yes
+rewrite_contact=yes  ; Necessary if endpoint does not know/register public ip:port
+ice_support=yes      ; This is specific to clients that support NAT traversal
+                     ;    for media via ICE,STUN,TURN. See the wiki at:
+                     ;    https://docs.asterisk.org/Configuration/Miscellaneous/Interactive-Connectivity-Establishment-ICE-in-Asterisk/
+                     ;    for a deeper explanation of this topic.
+
+[101]
+type=auth
+auth_type=userpass
+password=x9k2m5v7h3p8q4n1w6y0z9j4l8s3d7
+username=101
+
+[101]
+type=aor
+max_contacts=10
+
+[trunk-mtg01]
+type=endpoint
+transport=transport-udp
+context=from-trunk
+disallow=all
+allow=ulaw
+aors=trunk-mtg01-aor
+direct_media=no
+
+[trunk-mtg01-aor]
+type=aor
+max_contacts=1
+
+[trunk-mtg01-identify]
+type=identify
+endpoint=trunk-mtg01
+match=52.204.242.197