How to create basic IVR on Asterisk

To Create Basic IVR on Asterisk

Introduction

IVR stands for Interactive Voice Response, a technology that automates routine customer service interactions by permitting callers to interact using touch-tone digits or their voice. A basic example of an IVR application is an automated attendant or voice menu: callers are presented with a recorded menu and respond by selecting a digit or, in some cases, by entering an extension number.

Procedure

Step 1: Check the OS version by using the below command

root@linuxhelp1:~# lsb_release -a
No LSB modules are available.
Distributor ID:    Ubuntu
Description:    Ubuntu 22.04.2 LTS
Release:    22.04
Codename:    jammy

Step 2: Check the status of the Asterisk by using the below command

root@linuxhelp1:~# systemctl status asterisk
● asterisk.service - LSB: Asterisk PBX
     Loaded: loaded (/etc/init.d/asterisk; generated)
     Active: active (running) since Thu 2023-07-27 07:25:18 IST; 2 weeks 1 day ago
       Docs: man:systemd-sysv-generator(8)
      Tasks: 71 (limit: 4537)
     Memory: 121.0M
        CPU: 13min 58.637s
     CGroup: /system.slice/asterisk.service
             └─1427 /usr/sbin/asterisk

Jul 27 07:25:18 linuxhelp1 systemd[1]: Starting LSB: Asterisk PBX...
Jul 27 07:25:18 linuxhelp1 asterisk[1404]:  * Starting Asterisk PBX: asterisk
Jul 27 07:25:18 linuxhelp1 asterisk[1404]:    ...done.
Jul 27 07:25:18 linuxhelp1 systemd[1]: Started LSB: Asterisk PBX.
Jul 27 07:25:25 linuxhelp1 asterisk[1427]: radcli: rc_read_config: rc_read_config: can't open /etc/radiusc>
Jul 27 07:25:25 linuxhelp1 asterisk[1427]: radcli: rc_read_config: rc_read_config: can't open /etc/radiusc>

Step 3: Move to the Asterisk directory by using the below command

root@linuxhelp1:~# cd /etc/asterisk/

Step 4: Edit the extensions.conf file to add IVR by using the below command

root@linuxhelp1:/etc/asterisk# vim extensions.conf
Add the following lines
[internal]
exten => 168,1,Background(pls-wait-connect-call)
exten => 168,n,Goto(test,s,1)
[test]

exten => s,1,Answer()
 same => n,Background(press-1&or&press2)
 same => n,WaitExten()

exten => 1,1,Playback(you-entered)
 same => n,SayNumber(${EXTEN})
 same => n,Dial(PJSIP/168,20)
 same => n,Hangup()

exten => 2,1,Playback(you-entered)
 same => n,SayNumber(${EXTEN})
 same => n,Dail(PJSIP/169,20)
 same => n,Hangup()

exten => i,1,Playback(option-is-invalid)

Step 5: Login to the Asterisk CLI mode by using the below command

root@linuxhelp1:/etc/asterisk# asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvv
Asterisk 20.3.1, Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 20.3.1 currently running on linuxhelp1 (pid = 1427)

Step 6: Reload the dialplan by using the below command

linuxhelp1*CLI> dialplan reload
Dialplan reloaded.
    -- Including switch 'DUNDi/e164' in context 'ael-dundi-e164-switch'
    -- Including switch 'Lua/' in context 'public'
    -- Including switch 'Lua/' in context 'local'
    -- Including switch 'Lua/' in context 'demo'
    -- Including switch 'Lua/' in context 'default'
    -- Time to scan old dialplan and merge leftovers back into the new: 0.000382 sec
    -- Time to restore hints and swap in new dialplan: 0.000004 sec
    -- Time to delete the old dialplan: 0.000132 sec
    -- Total time merge_contexts_delete: 0.000518 sec
    -- pbx_config successfully loaded 27 contexts (enable debug for details).

Output of the basic IVR on the Asterisk
    -- Contact 169/sip:169@192.168.6.101:50089;transport=TCP;ob is now Reachable.  RTT: 3.135 msec
    -- Executing [168@internal:1] BackGround("PJSIP/167-00000024", "pls-wait-connect-call") in new stack
       > 0x7f533805a750 -- Strict RTP learning after remote address set to: 192.168.6.101:4000
       > 0x7f533805a750 -- Strict RTP learning after remote address set to: 192.168.6.101:4000
    -- <PJSIP/167-00000024> Playing 'pls-wait-connect-call.ulaw' (language 'en')
       > 0x7f533805a750 -- Strict RTP switching to RTP target address 192.168.6.101:4000 as source
    -- Executing [168@internal:2] Goto("PJSIP/167-00000024", "test,s,1") in new stack
    -- Goto (test,s,1)
    -- Executing [s@test:1] Answer("PJSIP/167-00000024", "") in new stack
    -- Executing [s@test:2] BackGround("PJSIP/167-00000024", "press-1&or&press2") in new stack
    -- <PJSIP/167-00000024> Playing 'press-1.ulaw' (language 'en')
    -- <PJSIP/167-00000024> Playing 'or.ulaw' (language 'en')
[Aug 11 07:48:42] WARNING[13945][C-0000001c]: file.c:824 ast_openstream_full: File press2 does not exist in any format
[Aug 11 07:48:42] WARNING[13945][C-0000001c]: file.c:1303 ast_streamfile: Unable to open press2 (format (ulaw)): No such file or directory
[Aug 11 07:48:42] WARNING[13945][C-0000001c]: pbx_builtins.c:1256 pbx_builtin_background: ast_streamfile failed on PJSIP/167-00000024 for press-1&or&press2
    -- Executing [s@test:3] WaitExten("PJSIP/167-00000024", "") in new stack
       > 0x7f533805a750 -- Strict RTP learning complete - Locking on source address 192.168.6.101:4000
    -- Executing [1@test:1] Playback("PJSIP/167-00000024", "you-entered") in new stack
    -- <PJSIP/167-00000024> Playing 'you-entered.ulaw' (language 'en')
    -- Executing [1@test:2] SayNumber("PJSIP/167-00000024", "1") in new stack
    -- <PJSIP/167-00000024> Playing 'digits/1.ulaw' (language 'en')
    -- Executing [1@test:3] Dial("PJSIP/167-00000024", "PJSIP/168,20") in new stack
    -- Called PJSIP/168
    -- PJSIP/168-00000025 is ringing
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [1@test:4] Hangup("PJSIP/167-00000024", "") in new stack
  == Spawn extension (test, 1, 4) exited non-zero on 'PJSIP/167-00000024'
linuxhelp1*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups

Conclusion:

We have reached the end of this article. In this guide, we have walked you through the steps How to Create a Basic IVR on Asterisk. Your feedback is much welcome.