Asterisk basic set up

Weekly sharing

Hi everyone, I am Ekim, a fresh Bootcamp graduate and an IT helper (I don't dare to call myself a programmer yet). Every Friday, I will share some of the work that I've done over the last week in a bid to get feedbacks from you guys and record my journey to become a programmer.

Introduction

Over the last week, I've been familiarizing myself with Asterisk, the open-source communications toolkit, which powers IP PBX systems, VoIP gateways, and conference servers. It is such a bitter start for a coding newbie like me, in which I struggled a lot in the installation process and making a simple phone call under the same network. This article aims at sharing a laconic work flow of the asterisk set-up and how a video call is made.

Before getting started ...

I'm using WSL Ubuntu to do the Asterisk set-up.

And before getting straight into the code, I would like to share several links that helped me a lot when I started from zero.

A simplified Chinese installation guide of Asterisk 17 (Thanks for Google Translate)
- https://zh.codepre.com/how-to-12071.html
Call making via SIP Softphone
- https://www.youtube.com/watch?v=rtHFdhCm434&t=432s&ab_channel=RocketSystems
Basic PBX set-up
- https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality
Asterisk firewall rules
- https://www.voip-info.org/asterisk-firewall-rules/

Get started!

To start with, as always,


 bash
sudo su     # enter the superuser mode 
apt update
apt upgrade
apt autoclean

How to install Asterisk ?

Installing Asterisk 18



cd /var/local/  # Go to /var/local as the download location
wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz    # Download Asterisk 18
tar zxf asterisk-18-current.tar.gz      # Unzip it
apt install flex bison subversion       # Install flex, bison, and subversion
contrib/scripts/get_mp3_source.sh       # Download the mp3 decoder save it in ./asterisk 18.xxx/contrib/scripts
contrib/scripts/install_prereq install  # Ensure all dependencies are set up
./configure     # Set up the asterisk based on the above
make menuconfig # Choose the modules you need
make -j 3       # Allocate cpu cores to run the installation
install     # Install all the things we've just set up

Softphone recommendation

I use MicroSIP and Linphone to make calls on Windows.

Connecting softphone to WSL Ubuntu

Asterisk Firewall Rules

Config of Linux IPTables firewall (https://www.voip-info.org/asterisk-firewall-rules/)



  # SIP on UDP port 5060. Other SIP servers may need TCP port 5060 as well
  iptables -A INPUT -p udp -m udp --dport 5060 -j ACCEPT

   # IAX2- the IAX protocol
  iptables -A INPUT -p udp -m udp --dport 4569 -j ACCEPT

   # IAX - most have switched to IAX v2, or ought to
  iptables -A INPUT -p udp -m udp --dport 5036 -j ACCEPT

   # RTP - the media stream
   # (related to the port range in /etc/asterisk/rtp.conf) 
  iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT


   # MGCP - if you use media gateway control protocol in your configuration
  iptables -A INPUT -p udp -m udp --dport 2727 -j ACCEPT

Basic calling set-up

Going to set our configs.


 bash
  cd /etc/asterisk
  nano pjsip.conf

The below is the pjsip.conf for basic calling (Account creation).


 conf
  ;================================ TRANSPORTS ==
  ; Our primary transport definition for UDP communication behind NAT.
  [transport-udp-nat]
  type = transport
  protocol = udp
  bind = 0.0.0.0
  ;================================ CONFIG FOR SIP ITSP ==

  [calling](!)                    ; template
  type=endpoint                   ; specify the below are the configurations related to an endpoint (a device)
  context=interaction             ; *** it refers to the context set in the dialplan (extensions.conf) ***
  allow = !all, ulaw, alaw  ; do not allow all codecs and only allow audio codecs - ulaw and alaw
  direct_media=no                   ; do not allow two devices directly talking to each other
  trust_id_outbound=yes
  rtp_symmetric=yes
  force_rport=yes
  rewrite_contact=yes
  device_state_busy_at=1
  dtmf_mode=rfc4733

  [auth-userpass](!)            ; template 
  type = auth                       ; specify the below are the configurations related to authentication
  auth_type = userpass      ; specify the type of the authentication is based on username and password

  [aor-single-reg](!)           ; template
  type = aor
  max_contacts = 1            ; specify the maximum sip addresses allocated. Here, that means each sip address could only connect to one device.

  [7000](calling)                 ; endpoint 7000 inherits the settings in calling template
  auth=7000                       ; authentication = 7000
  aors=7000                       ; address of record = 7000
  callerid = 7000 <7000>      ; caller's id = 7000

  [7000](auth-userpass)     ; account creation for endpoint 7000
  password = 7000
  username = 7000

  [7000](aor-single-reg)

  [7100](calling)
  auth=7100
  aors=7100
  callerid = 7100 <7100>

  [7100](auth-userpass)
  password = 7100
  username = 7100

  [7100](aor-single-reg)

The below are the configurations of the extensions.conf (Dialplan)



  [globals]
  INTERNAL_DIAL_OPT=,30

  [interaction]                       ; refers to the `context=interaction` in calling template in pjsip.conf
  exten = 7000,1,Answer()               ; the first argument 7000 refers to the number you dial on your softphone. 
  same = n,Dial(PJSIP/7000,60)      ; PJSIP/7000 refers to the endpoint 7000 , 60 means waiting for 60 seconds
  same = n,Playback(vm-nobodyavail)
  same = n,Voicemail(7000@main)
  same = n,Hangup()

  exten = 7100,1,Answer()
  same = n,Dial(PJSIP/7100,60)
  same = n,Playback(vm-nobodyavail)
  same = n,Voicemail(7000@main)
  same = n,Hangup()

  ; same means following the same extension
  ; n means the next action.

After setting the pjsip.conf and extensions.conf, you need to reload the pjsip and dialplan.



  asterisk -r   # go to the CLI of Asterisk (command line interface)



  core reload       # reload pjsip and modules

  dialplan reload       # reload dialplan

  pjsip show endpoints # showing the endpoints you have

The below is how you set up on the MicroSIP softphone

Do the same on another softphone for the 7100 endpoint.
Then call 7100 on 7000 endpoint / 7000 on 7100 endpoint to each other.
- Why I can call certain numbers on my device and can ring another device ?
https://wiki.asterisk.org/wiki/display/AST/Creating+Dialplan+Extensions
This is because we've done the set-up in the extensions.conf, in which we've set a number, for example, 6001, in the illustration below, for the dial action.



                  [from-internal]
                  exten=>6001,1,Dial(SIP/demo-alice,20)
                  exten=>6002,1,Dial(SIP/demo-bob,20)

Video calling

To achieve video calling, you only need to add the video codec in the pjsip.conf to make it work. And there's no other differences from the basic calling settings.


 conf
  [calling](!)
              type=endpoint
              context=interaction
              allow = !all, ulaw, alaw, vp8, h263, h263p        ; video codec vp8, h263 and h263p are added 
              direct_media=no
              trust_id_outbound=yes
              rtp_symmetric=yes
              force_rport=yes
              rewrite_contact=yes
              device_state_busy_at=1
              dtmf_mode=rfc4733

After changing the pjsip.conf, remember to reload the pjsip like the above again.

Conclusion

By now, I hope you could achieve video calling in your own network. In the meantime, stay healthy and stay tuned for more content !!!

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