DEV Community: Ekim

Asterisk AGI short sharing

Ekim — Fri, 02 Jul 2021 02:24:48 +0000

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.

Previously

Sharing of how I made my own command

Introduction

Today, I am going to start integrating my previous call flow with the use of AGI. To do that, we need the help of AGI (Asterisk Gateway Interface). The reason we might need to do that is that sometimes we want to record every call and insert the call logs into the database. So that, we could know what calls have been made or received. Moreover, the use of database could help us create an IVR , for it could contain all the IVR logic we have.

What is AGI ?

AGI provides an interface between the Asterisk dialplan and an external program that wants to manipulate a channel in the dialplan.

PostgreSQL installation & DB creation

# Install PostgreSQL
sudo apt install postgresql postgresql-contrib

# start the postgres service
sudo service postgresql start

# Change to user postgres as it can perform all tasks on the database
sudo su postgres

# login to the DB
psql

# You should see something similar in your terminal
postgres=# 

# Create database
# don't miss the colon
CREATE Database <your-database-name-here>;

# Create an account to manage the database
# don't miss the colon and the quotation marks
CREATE USER <your-username-here> WITH PASSWORD 'your-password-here' SUPERUSER;

# Leaving Database and postgres user
ctrl + d 
ctrl + d

# Try login from the current user
psql -U <user_name> -W -h <hostname> <db_name>

Using AGI in Asterisk

In Asterisk, AGI can be treated as a dialplan application in extensions.conf. In the following, I will show you how you could use it as a way to insert data into the PostgreSQL database.

In modules.conf

# add the agi module
load = res_agi.so

In pjsip.conf

[transport-udp-nat]
type = transport
protocol = udp
bind = 0.0.0.0

[calling](!)                 
type=endpoint                
context=interaction          
allow = !all, ulaw, alaw     
direct_media=no              
trust_id_outbound=yes
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
device_state_busy_at=1
dtmf_mode=rfc4733

[auth-userpass](!)       
type = auth              
auth_type = userpass     
[aor-single-reg](!)     
type = aor
max_contacts = 1         

[7000](calling)          
auth=7000                
aors=7000                
callerid = 7000 <7000>   

[7000](auth-userpass)    
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)

In extensions.conf

[interaction]
exten = _7X00,1,NoOp(${EXTEN})
; setting variables for data insertion
same = n,Set(_callInTime=${STRFTIME(,,%Y%b%d-%H%M%S)})
same = n,Set(_callerId=${CALLERID(num)})
same = n,Set(_extension=${EXTEN})
same = n,Set(_callDuration=${CDR(billsec)})
same = n,Answer
same = n,Dial(PJSIP/${EXTEN},30)
same = n,Goto(update_call_duration,s,1)

[update_call_duration]
; call_log is the agi file, the rest are the arguments
exten = s,1,agi(call_log,${callerId},${callInTime},${extension},${callDuration})
same = n,Hangup

In the extensions.conf, we use the agi dialplan application. Think of it as running a program like node abc.js or python3 cde.py.

So, where is our program ?

Actually, we need to go to the default agi-bin path to create the program.

But that would be too much for me today. So, please bear with me and stay tuned for the next week sharing and I will go deeper into the agi program.

Sharing of how I made my own command

Ekim — Fri, 25 Jun 2021 02:34:54 +0000

Weekly sharing

Introduction

Last week was a hectic journey. I didn't have much time working on blog writing. So this blog wouldn't be as detailed as others I've written. Despite that, I would like to share with you some tips for making your own command on Ubuntu.

Previously

Multi-Master VRRP set-up

How I made my own command ?

Writing in ~./bashrc

After writing a .js file or other kinds of files, you could run it with node or other runtime tools.
In the ~./bashrc, we could make use of alias to help us run those files in a much easier way
For example to execute node abcd.js, we could make it with as simple as only one word abcd
Here is how we could do it

       alias abcd="node /var/local/abcd.js"

The down side of writing alias in ~./bashrc is that the script can only be used by the users whose ~./bashrc has those script.
If there are multiple accounts in the server, you could not make sure that everyone could use that shortcut

Writing an executable at /usr/local/bin

An executable is a file which is set to be executed
In that file, you specify how the script is run
Using the above abcd as an example, what you need is the following

       cd /usr/local/bin
       touch abcd
       chmod a+x ./abcd         # enable execution for all groups

       # edit the executable
       vim abcd

       # ------- vim abcd -------
       #!bin/bash

       node "/var/local/abcd.js"
       # ------- vim abcd -------

By doing so, every user could use the abcd command
If the abcd.js accepts 3 arguments, which you include something like let [ var1, var2, var 3] = process.argv.slice(2) in your .js file to get the arguments , you could further do this

       # ------- vim abcd -------
       #!bin/bash

       node "/var/local/abcd.js" $1 $2 $3
       # ------- vim abcd -------

Conclusion

Now, you have created a command called abcd in your system. Input your newly created command into your console and see if it could run as expected. If I did anything wrong, please let me know as well. That's all for today. I hope you enjoy reading.

Multi-Master VRRP set-up

Ekim — Fri, 18 Jun 2021 04:31:15 +0000

Weekly sharing

Introduction

In the last week sharing, I revealed that I had failed to make my VRRP working. Fortunately, that was fixed and is running successfully now. One thing I noticed was that most resources online only taught people how to make master-backup VRRP instead of master-master. Today, I would like to share my configurations of Keepalived for master-master usage.

Previously

PPTP VPN without GUI

Installation and configurations of keepalived

Machine one

sudo apt-get install keepalived
cd /etc/keepalived
vim keepalived.conf

# ----- vim keepalived.conf -----
vrrp_instance machineOne {
        state MASTER
        nopreempt                   # preempt = get back the master position when recovers, nopreempt = stay in backup position when recovers (dual-master setting)
        interface eno1              # interface
        virtual_router_id 101
        priority 101                # same priority to make sure both machines are master
        advert_int 1                # interval between advertisements
        authentication {
                auth_type PASS
                auth_pass abcdefg
        }
        virtual_ipaddress {         # virtual IP 
                192.168.29.11
        }
}
# ----- vim keepalived.conf -----

Machine two

sudo apt-get install keepalived
cd /etc/keepalived
vim keepalived.conf

# ----- vim keepalived.conf -----
vrrp_instance esl35 {
        state MASTER
        nopreempt
        interface eno1
        virtual_router_id 101
        priority 101
        advert_int 1
        authentication {
                auth_type PASS
                auth_pass abcdefg
        }
        virtual_ipaddress {
                192.168.29.11
        }
}
# ----- vim keepalived.conf -----

SSH key integration

Before starting the keepalived service, we need to make sure that the SSH key files are the two machines are the same if master-master VRRP approach is adopted. This will avoid conflicts between machines.

# In any one of the machine
# e.g. 192.168.29.10
cd /etc/ssh
ls 

# ----- ls -----
#moduli        sshd_config       ssh_host_dsa_key.pub    ssh_host_ed25519_key      ssh_host_rsa_key.pub
#ssh_config    sshd_config.d     ssh_host_ecdsa_key      ssh_host_ed25519_key.pub  ssh_import_id
#ssh_config.d  ssh_host_dsa_key  ssh_host_ecdsa_key.pub  ssh_host_rsa_key
# ----- ls -----

# syncing all those files to another machine
# account@server-ip should be like abcd@192.168.29.10
rsync ./* < account@server-ip >:/etc/ssh

Conclusion

I hope you enjoy my sharing so far. I am not sure if I could keep writing and sharing every week. I aimed to make the VRRP working through rVRRPd, but I could not find sufficient resources that could help me set that up successfully. If you do know how to make it working, please comment below no matter how ancient this sharing has become.

PPTP VPN without GUI

Ekim — Fri, 11 Jun 2021 03:14:07 +0000

Weekly sharing

Previously

A wee Asterisk self-signed certificate sharing

Introduction

Recently, I have been working and experimenting on VRRP stuff. Most resources I found on the Internet are related to CISCO, which is not suitable for me. So, I came up with an idea of connecting my ec2 instances through PPTP VPN to achieve VRRP. Things look logical, but I still failed to make the VRRP running. Nevertheless, allow me to share how you could connect a PPTP VPN on Ubuntu without the help of GUI.

PPTP VPN set-up

Since I am using an ASUS router, it is pretty easy to set up a PPTP VPN server.

https://www.asus.com/en/support/FAQ/114892/

Ubuntu PPTP VPN connection

Connecting VPN with GUI is pretty easy. However, as I am working on a headless Ubuntu, I need to do the configurations myself. Here is what I found online, which is very useful.

https://wiki.archlinux.org/title/PPTP_Client

Installation of PPTP client

# Installation of PPTP client
apt-get update
apt-get upgrade
apt-get autoclean
apt-get install pptp-linux

Options set-up

# options set-up
cd /etc/ppp/
vim options

# --------- options ---------
# Lock the port
lock
# We don't need the tunnel server to authenticate itself
noauth
# Turn off compression protocols we know won't be used
nobsdcomp
nodeflate
# We won't do PAP, EAP, CHAP, or MSCHAP, but we will accept MSCHAP-V2
# (you may need to remove these refusals if the server is not using MPPE)
refuse-pap
refuse-eap
refuse-chap
refuse-mschap
# --------- options ---------

chap-secrets set-up

# chap-secrets set-up
chmod 0600 /etc/ppp/chap-secrets
vim /etc/ppp/chap-secrets

# --------- chap-secrets ---------
# Secrets for authentication using CHAP
# client        server  secret                  IP addresses
<username>      PPTP    <password>     *
# --------- chap-secrets ---------

# username and password are your VPN login name and password set in the PPTP server via your router
# remember to use double quote for both of your username and password

tunnel name

# giving name for your vpn connection
touch /etc/ppp/peers/vpn            # I want my vpn connection called "vpn"
vim /etc/ppp/peers/vpn  

# --------- tunnel ---------
pty "pptp <server> --nolaunchpppd"
name <username>
remotename PPTP
require-mppe-128
file /etc/ppp/options
ipparam <tunnel>
# --------- tunnel ---------

# server means the remote address of the VPN server (I used my own public IP)
# tunnel means the name of the connection, which is "vpn" in this example 

# remember to use double quote for both of your username

Implementation

# Connect
sudo pon vpn                                # start PPTP VPN connection4

# Check
ip addr     
# you should see a ppp0 interface with a subnet address

# ppp0: <POINTOPOINT,MULTICAST,NOARP,UP,LOWER_UP> mtu 1446 qdisc fq_codel state UNKNOWN group default qlen 3
    # link/ppp
    # inet 192.168.10.9 peer 192.168.1.1/32 scope global ppp0
       # valid_lft forever preferred_lft forever

# Disconnect
sudo poff vpn                           # turn off the PPTP VPN connection

Routing of servers

If you have multiple devices using the VPN, it does not mean that they could connect with each other automatically. When you ping from one device to another, you would receive no response as the routing has not yet been set up.

Therefore, you may need to add the following command in your devices

# Route all the traffic with a destination of 192.168.10. through ppp0 interface
ip route add 192.168.10.0/24 dev ppp0

Conclusion

Hardly could I find resources on VPN connection fully with command lines. I hope you would somehow find this sharing useful. That's all for today. See you next time.

A wee Asterisk self-signed certificate sharing

Ekim — Fri, 04 Jun 2021 02:36:20 +0000

Weekly sharing

Previously

Basic IVR with Asterisk

Introduction

Last week, I was doing something related to WebRTC, which was an amazing experience. However, there's one thing I got stuck for a pretty long time related to web socket. It turns out I had to manually permit access for different ports I used even though I had created a self-signed certificate. Today, I would like to share how to create a self-signed certificate on Asterisk and import that into your browser.

Create your self-signed certificate

It is not difficult to create a self-signed certificate.

cd /asterisk-18.3.0/contrib/scripts         # go to /contrib/script

# -C = DNS name or our IP address, -O = organizational name, -d = output directory of the keys
# use the same pass phrase you want for the whole registration
sudo ./ast_tls_cert -C <server-ip> -O "<organizational name>" -d /etc/asterisk/keys

# check if the 'keys' file exists
ls -w 1 /etc/asterisk/keys

sudo service asterisk restart

Import your certificate to let waive manual permission access

Now, we are going to import our ca.crt into our browser. I will use Firefox and Chrome as examples.
You may wonder why you need to import the certificate. The reason is that once you import your certificate, you don't need to permit access for browser to go to your web every time.
More importantly, when you play around with WebRTC, you need to enable the web socket every time if you don't do so, which is absolutely annoying.

Firefox browser set-up

If you are using Firefox browser, please:

Go to "Options"

Select "Privacy & Security"

Scroll to the bottom of the page under "Security" and "Certificates"
Select "View Certificates"

Make sure "Authorities is selected"

Import "ca.crt", the self-signed certificate we have just created, which is located at /etc/asterisk/keys
Check "Trust this CA to identify websites" and "Trust this CA to identify email users"

Then click "ok"

Conclusion

If you also face the web socket granting permission problem like I did, I hope you would find this article helpful to you. Or if you are interested in WebRTC, please check out this guide: https://github.com/paneru-rajan/asterisk-jssip. A big shoutout to paneru-rajan who provides the WebRTC set-up guide. In the meantime, please take care and stay healthy, and I will see you next week.

Basic IVR with Asterisk

Ekim — Fri, 28 May 2021 03:14:19 +0000

Weekly sharing

Previously

Asterisk Manager with JavaScript

Introduction

Today, I would like to share how we could make a simple interactive voice response (IVR). I know you might have had some horrible experiences with that, but it would be a weird but fun experience when you hear your voice on the phone talking to you.

What is Interactive Voice Response (IVR) menu?

An IVR menu is a list of options that are presented to an inbound caller which prompts them to make a selection, thus qualifying the call and directing it to the most appropriate contact.

Recording our own audio track through Asterisk for IVR

If you have read my previous posts, you would know that the codecs that telephony supports are limited.
Even if you have recorded a very good quality audio track, you still need to convert it into the format that our phones support.
To save time, I am not going to go through the ffmpeg part again. Instead, I am going to record all the IVR audio track through the Record dialplan application.
In pjsip.conf

    [transport-udp-nat]
    type = transport
    protocol = udp
    bind = 0.0.0.0

    [calling](!)                 
    type=endpoint                
    context=interaction          
    allow = !all, ulaw, alaw     
    direct_media=no              
    trust_id_outbound=yes
    rtp_symmetric=yes
    force_rport=yes
    rewrite_contact=yes
    device_state_busy_at=1
    dtmf_mode=rfc4733

    [auth-userpass](!)       
    type = auth              
    auth_type = userpass     
    [aor-single-reg](!)     
    type = aor
    max_contacts = 1         

    [7000](calling)          
    auth=7000                
    aors=7000                
    callerid = 7000 <7000>   

    [7000](auth-userpass)    
    password = 7000
    username = 7000

    [7000](aor-single-reg)
    mailboxes = 7000@main

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

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

    [7100](aor-single-reg)
    mailboxes = 7100@main

In extensions.conf

[interaction]
exten = *555,1,NoOp(recording)
same = n,Answer
same = n,Record(record/greeting.wav)
same = n,Playback(vm-received)
same = n,Playback(en/digits/1)
same = n,Record(record/press1.wav)
same = n,Playback(vm-received)
same = n,Playback(en/digits/2)
same = n,Record(record/press2.wav)
same = n,Playback(vm-received)
same = n,Playback(en/digits/3)
same = n,Playback(vm-last)
same = n,Record(record/wrong-key.wav)
same = n,Playback(vm-received)
same = n,Playback(en/digits/4)
same = n,Hangup

;exten = _7X00,1,NoOp(Call for ${EXTEN})    
;same = n,Dial(PJSIP/${EXTEN})      
;same = n,Hangup    

exten = 8888,1,NoOp(Call IVR)
same = n,Goto(ivr,s,1)

[ivr]
exten = s,1,NoOp(You are in IVR now)
same = n,Answer                     ; here is the start of the SIP protocol
same = n(play),Playback(record/greeting)
same = n,WaitExten(10)              ; wait 10 seconds for the caller to press key

exten = 1,1,NoOp(Pressed 1)
same = n,Playback(record/press1)
same = n,Goto(s,play)               ; go back to 'n(play),Playback(record/greeting)'

exten = 2,1,NoOp(Pressed 2)
same = n,Playback(record/press2)
same = n,Goto(s,play)

exten = i,1,NoOp(unknown key)       ; handle invalid keys --> when keys other than 1 and 2 are pressed
same = n,Playback(record/wrong-key)
same = n,Goto(s,play)

exten = t,1,NoOp(Waited too long for exten)     ; handle WaitExten timeout
same = n,Goto(s,play)

Implementation

First of all, let's record all the necessary navigation menu tracks.
I have set in the dialplan that when we call *555 , we will start recording each audio file one by one. - When you hear the first 'beep' sound, it is for the recording of the greeting. - For example, "thanks for calling, English, press 1; Deutsch, press 2". - When you finish each recording, remember to press # to save it.
If successful, you will hear the number of recording received.
- If you finished one recording, you will hear "received 1"
When you hear "last", that means you are going to record the last audio.
Here is the sequence of the recording sections
1. greeting
  1. the navigation after pressing 1
  2. the navigation after pressing 2
  3. the navigation when wrong key is pressed
To go into the IVR, dial 8888 and it will start the IVR menu.

Conclusion

Now, you have just made yourself an IVR menu. If you follow tight, you would realize that the whole IVR is a loop. You could feel free to add the Hangup dialplan application wherever you like. One thing I would like to stress here is that this IVR structure is a very basic one as things are repetitive and could be refactorized in another way. A more advanced one might be introduced some time later (I hope ... ) with the use of database and AGI (Asterisk Gateway Interface). That's all for today. See you next time.

Asterisk Manager with JavaScript

Ekim — Fri, 21 May 2021 03:35:42 +0000

Weekly sharing

Previously

Brief intro of Asterisk Manager

Introduction

This time, we will go through how we could control the asterisk through our codes. Below, I will use JavaScript to illustrate how you could fiddle with asterisk without the CLI (command-line interface). And as the previous sharing, my pjsip.conf and extensions.conf align with my first sharing.

Set up your environment

Install the package we need

npm install asterisk-ami-client

Coding section

Create a main.js file and copy these

    const AmiClient = require("asterisk-ami-client");
    let client = new AmiClient();

    client
      .connect("<YOUR USERNAME>", "<YOUR PASSWORD>", { host: "127.0.0.1", port: 5038 })     // connect to your AMI remotely
      .then(() => {

        client
            .on('connect', () => console.log('connect'))                // show connection logs in terminal
            .on('event', event => console.log(event))                   // show AMI event logs in terminal
            .on('response', response => console.log(response))          // show response logs in terminal
            .on('disconnect', () => console.log('disconnect'))          // show disconnection logs in terminal
            .on('reconnection', () => console.log('reconnection'))      // show reconnection logs in terminal
            .on('internalError', error => console.log(error))           // show AMI error logs in terminal
            .action({                                                   // manager action   
                Action:"Originate",                                     // Originate call
                Channel:"PJSIP/7000",                                   // calling from endpoint 7000
                Exten:"7100",                                           // expected to be received by endpoint 7100
                Context:"interaction",
                Priority:"1",
            })

    })
    .catch(error => console.log(error));

please replace "<YOUR USERNAME>" and "<YOUR PASSWORD>" with your AMI login username and password.
Then, it's time to run the code

    node main.js            # make sure you execute the command in the correct path that has the newly created main.js

You should now be able to originate a call without using the CLI. And in your terminal, your could see all the states and processes of asterisk when you connect to the AMI and originate the call. You could use them to do different kinds of things.
For example, think about converting voice messages in the voicemail into texts, and then send those texts to a telegram group. Wouldn't it be amazing?

Conclusion

It is a short and simple sharing, but you could do various things based on it. I tend to keep things easy and avoid people getting overwhelmed by the asterisk thing. I hope you enjoy my reading so far. In the meantime, stay healthy and stay tuned for more content !!!

Brief intro of Asterisk Manager

Ekim — Fri, 14 May 2021 04:02:57 +0000

Weekly sharing

Previously

Make calls through Asterisk CLI

Introduction

After being able to make calls and play around with playback and record dialplan applications, it's time to prepare ourselves for using those functions through codes. But before that, we need to know what Asterisk Manager Interface is.

**Warning: **Only try the following within your own local network.

What is Asterisk Manager Interface ?

It is like a remote control of the asterisk.
Unlike running asterisk commands on the CLI, the state and process of which cannot be seen and controlled, the asterisk manager interface resolves those issues and helps monitor the operation of the asterisk.
Since AMI allows you to remotely control the asterisk, it is advised that you do not use AMI directly with a public IP address without protection like SSL connection or a VPN tunnel.

Steps using AMI

Check if port 5038 is available as we are going to use it.

    netstat -lpn

Configure the manager.conf

    [general]
    enabled = yes       ;by default, it is "no" for security reason.
    port = 5038
    bindaddr = 0.0.0.0  ;bind the ip address to localhost for lcoal testing, need to change when remote control it
    [ekim]              ;login username, often for CRM, ERP or dialer 
    secret=xxxxxxxx     ;login password, should be complex
    deny=0.0.0.0/0.0.0.0        ;deny everything --> much safer to permit each ip address we trust
    permit=XXX.XXX.XX.XX/255.255.255.0      ;permit local network after an entire denial, this can also be done in acl.conf
    read=all        ;read all logs
    write=all       ;write all logs

Restart asterisk after IP and port changes

    sudo service asterisk restart

Use telnet to connect to AMI
- telnet allows access to remote servers, which is based on the TELNET protocol. Only recommend you use telnet to connect within your own network.

    # telnet <permitted IP address> <port>
    telnet XXX.XXX.XX.XX 5038

    # after that, tab "enter" once as we need to continue to login
    Action:login
    Username:ekim
    Secret:xxxxxxxx
    # when finish registering, tab "enter" twice for signaling you're done

After that, you should see the Response: Success and could see all actions in the asterisk.

    Response: Success
    Message: Authentication accepted

    Event: FullyBooted
    Privilege: system,all
    Uptime: 67
    LastReload: 67
    Status: Fully Booted

Now, you have accessed to your own asterisk server and entered the AMI. That means you are no longer playing on the CLI. You are 'remotely controlling' your asterisk pbx.
So, why not we try to originate a call (make a call) and see what would happen ?
The below is based on our first tutorial pjsip and dialplan settings

      Action:Originate
      Channel:PJSIP/7000
      Exten:7100
      Context:interaction
      Priority:1
# remember to tab 'enter' twice to signal that you are finished

Mine is working now, how about yours ? Btw, if you want to leave the AMI, you could do this

      Action:logoff
      # remember to tab 'enter' twice to signal that you are finished

Conclusion

You might want to keep your eyes close to the log data shown in AMI since those are the things we might need if we touch on the coding part. That's all for today. In the meantime, stay healthy and stay tuned for more content !!!

Make calls through Asterisk CLI

Ekim — Fri, 07 May 2021 03:18:29 +0000

Weekly sharing

Previously

Record and Hear your voice through Asterisk

Introduction

As promised, I will go through a more user-friendly process of recording a voice. Also, I would like to show you how to originate a call (make a call) from Asterisk.

Voice Navigation

If you are careful enough, you would have noticed that the Record dialplan application includes a "beep" sound when you dial 41XXXX.
So now, let's make our voice navigation based on that.
Record a track that signals people to start recording their voices after hearing a "beep" sound and press # when they finish recording.

Phone restriction

If you dial 41XXXX to record your navigation audio, just like the previous sharing, you wouldn't need to worry about the audio restrictions. However, if you record your navigation audio through your smartphone application or other devices, the recording you got is highly likely incompatible with the telephony codecs.
Often, for hardphones, there are restrictions for the audio file.
- Codec → .gsm (convention and saves space) | .wav (better quality but occupying more space)
- Channels → mono
- Sample rate → 8000hz (at max)
- Bits per sample → 16bits
So, we need to convert the file into what we need using ffmpeg.

ffmpeg -i <file> -ac 1 -ar 8000 -c:a pcm_s16le "<new file>.wav"

Audio track location

After that, let's move the converted file to the location where Asterisk stores sounds.

cd /var/lib/asterisk/sounds/

I would like to create a folder for better organization, so I create a folder called "playback"

mkdir playback

Dialplan modifications

Going back to our extensions.conf,

[recordAndPlayback]
exten = _41XXXX,1,Answer
; Playback is the application; playback is the folder in /var/lib/asterisk/sounds/ ; xxxxx is the converted audio name where .wav is excluded.
same = n,Playback(playback/xxxxx)   
; record your voice and save it in the name of XXXX, the any 4 digits you dialed and in .gsm format
same = n,Record(record/${EXTEN:2}.gsm)      
same = n,Wait(1)
same = n,Hangup()

exten = _42XXXX,1,Answer
same = n,Wait(1)
; play the audio file you recorded from /var/lib/asterisk/sounds/record
same = n,Playback(record/${EXTEN:2})        
same = n,Wait(1)
same = n,Hangup()

As always, after doing changes in the dialplan, reload it in the CLI.

sudo asterisk -rvvvvv
dialplan reload

Now, it is more user-friendly.

Calling through CLI

Before we originate a call through CLI, we need to do some settings in pjsip.conf, extensions.conf and modules.conf.
In pjsip.conf,

[transport-udp-nat]
type = transport
protocol = udp
bind = 0.0.0.0


[endpoint-useragent](!)
type=endpoint
context=calling         ;Context is now "calling"
allow = !all,ulaw,alaw
direct_media=no
trust_id_outbound=yes
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
device_state_busy_at=1
dtmf_mode=rfc4733

[auth-userpass](!)
type = auth
auth_type = userpass

[aor-single-reg](!)
type = aor
max_contacts = 1

[1112](endpoint-useragent)      ;created 1112 endpoint
auth=1112
aors=1112
callerid = 1112 <1112>

[1112](auth-userpass)
password = 1112
username = 1112

[1113](endpoint-useragent)      ;created 1113 endpoint
auth=1113
aors=1113
callerid = 1113 <1113>

[1113](auth-userpass)
password = 1113
username = 1113

In extensions.conf

[calling]
exten = _111X,1,Dial(PJSIP/${EXTEN})
 same = n,Hangup()

In modules.conf

;Application
load = app_bridgewait.so
load = app_dial.so
load = app_playback.so
load = app_stack.so
load = app_verbose.so
load = app_voicemail.so
load = app_directory.so
load = app_confbridge.so
load = app_queue.so
load = app_record.so
load = app_originate.so     ; added 
load = res_clioriginate.so  ; added

Also, don't forget to set up those endpoints on your two different softphones. For me, I use MicroSIP and Linphone.
And remember, restart after adding new modules in modules.conf, and reload the dialplan and pjsip.

sudo service asterisk restart
sudo asterisk -rvvvvv
dialplan reload     ; not necessary after restart
core reload     ; not necessary after restart
channel originate LOCAL/1112@calling application Dial(PJSIP/1113)       # this is how you originate a call via CLI

TLDR: This is how you originate a call from CLIchannel originate LOCAL/1112@calling application Dial(PJSIP/1113).

Conclusion

I hope you know more about the codec for telephony system and know how to make calls from CLI. That's it for today. If you have problems regarding ffmpeg installation, which I omitted in this sharing, please comment below and see if I could help. In the meantime, stay healthy and stay tuned for more content !!!

Record and Hear your voice through Asterisk

Ekim — Fri, 30 Apr 2021 02:30:44 +0000

Weekly sharing

Previously

Asterisk basic set up

Introduction

Last week was quite a fuzzy journey. Not much concrete output has been done. Although the progress is slow, I still would like to share how to record and play audio through Asterisk.

Recording through dialing 41XXXX

Let's create an endpoint of 1114 for our softphone.

In pjsip.conf,

;================================ TRANSPORTS ==
; Our primary transport definition for UDP communication behind NAT.
[transport-udp-nat]
type = transport
protocol = udp
bind = 0.0.0.0
; NAT settings
;local_net = 10.0.0.0/8
;external_media_address = 203.0.113.1
;external_signaling_address = 203.0.113.1

;================================ CONFIG FOR SIP ITSP ==

[endpoint-useragent](!)
type=endpoint
context=recordAndPlayback
allow = !all,ulaw,alaw
direct_media=no
trust_id_outbound=yes
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
device_state_busy_at=1
dtmf_mode=rfc4733

[auth-userpass](!)
type = auth
auth_type = userpass

[aor-single-reg](!)
type = aor
max_contacts = 1

[1114](endpoint-useragent)
auth=1114
aors=1114
callerid = 1114 <1114>

[1114](auth-userpass)
password = 1114
username = 1114

What I want to do here is to dial a number, which starts with 41 and is followed by any 4 digits, to record my voice on the softphone. And then, I would like hear my voice again when I dial 42 + any 4 digits I've just dialed.

So, in the extensions.conf

[recordAndPlayback]
exten = _41XXXX,1,Answer
same = n,Record(record/${EXTEN:2}.gsm)      ; record your voice and save it in the name of XXXX, the any 4 digits you dialed and in .gsm format
same = n,Wait(1)
same = n,Hangup()

exten = _42XXXX,1,Answer
same = n,Wait(1)
same = n,Playback(record/${EXTEN:2})        ; play the audio file you recorded from /var/lib/asterisk/sounds/record
same = n,Wait(1)
same = n,Hangup()

After that, add the load = app_record.so in your modules.conf.

;Application
load = app_bridgewait.so
load = app_dial.so
load = app_playback.so
load = app_stack.so
load = app_verbose.so
load = app_voicemail.so
load = app_directory.so
load = app_confbridge.so
load = app_queue.so
load = app_record.so        ; add this into your modules.conf, so that you do not need to load the module every time.

It's time to restart the asterisk to make things ready.

sudo service asterisk restart

Then remember to go to CLI to reload the dialplan and pjsip.

sudo asterisk -rvvvvv

dialplan reload
core reload

Open you softphone and do the following settings

The SIP server and Domain should be your Ubuntu's IP. You can check it by this command:

ip addr
If you have problem connecting your softphone to your Ubuntu, please check that you've done these firewall settings: https://www.voip-info.org/asterisk-firewall-rules/ .

Hearing your voice

So, if you followed the aforementioned steps, you should be able to record your voice by dialing, for example, 411234. Noted that when you finish your recording, press # to finish the call. Do Not End The Call Right Away, or it won't save the recording.

After recording your voice, it's time to appreciate the cuss words you've just sworn yourself. (Yeah, I know it !!!! We all do it to ourselves.)

When you dial 421234, you should hear your recording straight away.

Conclusion

Usually, when we want someone to record an audio track, we won't just use the Record dialplan application alone because they won't have an idea that they need to press # to save the recording. Hence, it is better to record a track that guides them what to do after they record their voices. That's what we'll do next time. For the time being, stay healthy and stay tuned !!!

Asterisk basic set up

Ekim — Fri, 23 Apr 2021 04:16:17 +0000

Weekly sharing

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 !!!