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How to create a basic Asterisk IVR on Debian 12

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To Create A Basic Asterisk IVR On Debian 12

Introduction

Asterisk includes a wealth of functions that make it a powerful IVR platform: audio playback and recording, digit collection, database and web service access, calendar integration, and optional speech recognition and synthesis.

Procedure Steps

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

root@linuxhelp:~# cat /etc/os-release
PRETTY_NAME="Debian GNU/Linux 12 (bookworm)"
NAME="Debian GNU/Linux"
VERSION_ID="12"
VERSION="12 (bookworm)"
VERSION_CODENAME=bookworm
ID=debian
HOME_URL="https://www.debian.org/"
SUPPORT_URL="https://www.debian.org/support"
BUG_REPORT_URL=https://bugs.debian.org/

Step 2: Goto the following location by using the below command.

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

Step 3: Open and Edit the extensions.conf file by using the below command.

root@linuxhelp:/etc/asterisk# vim extensions.conf
[internal]

exten => 110,1,NoOp(Calling ${EXTEN} From Caller ${CALLERID})
same=> n,Answer()
same=> n,Playback(for-service&press-1&for-sales&press-2)
same=> n,Read(VAR,,,,,4)
same=> n,GotoIf($[ "${VAR}" = "1" ]?111,1)
same=> n,GotoIf($[ "${VAR}" = "2" ]?112,1:110,1)

exten => 111,1,NoOp(For Service Calling From ${CALLERID})
same=> n,Answer()
same=> n,SayDigits(1)
same=> n,Playback(for-service)
same=> n,Dial(PJSIP/111,10)
same=> n,Hangup()

exten => 112,1,NoOp(For Sales Calling From ${CALLERID})
same=> n,Answer()
same=> n,SayDigits(2)
same=> n,Playback(for-sales)
same=> n,Dial(PJSIP/112,10)
same=> n,Hangup()

Step 4: Login to the asterisk console by using the below command.

root@linuxhelp:/etc/asterisk# asterisk -rvvvvvvvvvvvvvvvvvvv
Asterisk 20.4.0, 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 20.4.0 currently running on linuxhelp (pid = 1294)
linuxhelp*CLI>

Step 5: Reload the dialplan by using following command.

linuxhelp*CLI> dialplan reload
Dialplan reloaded.
    -- Time to scan old dialplan and merge leftovers back into the new: 0.000004 sec
    -- Time to restore hints and swap in new dialplan: 0.000003 sec
    -- Time to delete the old dialplan: 0.000004 sec
    -- Total time merge_contexts_delete: 0.000011 sec
    -- pbx_config successfully loaded 1 contexts (enable debug for details).

Step 6: Make the call to check service call from IVR menu by using the below command. snap 1

linuxhelp*CLI> 
    -- Executing [110@internal:1] NoOp("PJSIP/112-00000006", "Calling 110 From Caller ") in new stack
    -- Executing [110@internal:2] Answer("PJSIP/112-00000006", "") in new stack
       > 0x7f5654010aa0 -- Strict RTP learning after remote address set to: 192.168.6.107:4010
    -- Executing [110@internal:3] Playback("PJSIP/112-00000006", "for-service&press-1&for-sales&press-2") in new stack
    -- <PJSIP/112-00000006> Playing 'for-service.ulaw' (language 'en')
       > 0x7f5654010aa0 -- Strict RTP switching to RTP target address 192.168.6.107:4010 as source
    -- <PJSIP/112-00000006> Playing 'press-1.ulaw' (language 'en')
    -- <PJSIP/112-00000006> Playing 'for-sales.ulaw' (language 'en')
    -- <PJSIP/112-00000006> Playing 'press-2.ulaw' (language 'en')
    -- Executing [110@internal:4] Read("PJSIP/112-00000006", "VAR,,,,,4") in new stack
       > 0x7f5654010aa0 -- Strict RTP learning complete - Locking on source address 192.168.6.107:4010
    -- User entered '1'
    -- Executing [110@internal:5] GotoIf("PJSIP/112-00000006", "1?111,1") in new stack
    -- Goto (internal,111,1)
    -- Executing [111@internal:1] NoOp("PJSIP/112-00000006", "For Service Calling From ") in new stack
    -- Executing [111@internal:2] Answer("PJSIP/112-00000006", "") in new stack
    -- Executing [111@internal:3] SayDigits("PJSIP/112-00000006", "1") in new stack
    -- <PJSIP/112-00000006> Playing 'digits/1.ulaw' (language 'en')
    -- Executing [111@internal:4] Playback("PJSIP/112-00000006", "for-service") in new stack
    -- <PJSIP/112-00000006> Playing 'for-service.ulaw' (language 'en')
    -- Executing [111@internal:5] Dial("PJSIP/112-00000006", "PJSIP/111,10") in new stack
    -- Called PJSIP/111
    -- PJSIP/111-00000007 is ringing
  == Everyone is busy/congested at this time (1:1/0/0)
    -- Executing [111@internal:6] Hangup("PJSIP/112-00000006", "") in new stack
  == Spawn extension (internal, 111, 6) exited non-zero on 'PJSIP/112-00000006'

Step 7: Make the call to check sales call from ivr menu by using the below command. snap 2

-- Executing [110@internal:1] NoOp("PJSIP/112-00000008", "Calling 110 From Caller ") in new stack
  -- Executing [110@internal:2] Answer("PJSIP/112-00000008", "") in new stack
     > 0x7f5654010aa0 -- Strict RTP learning after remote address set to: 192.168.6.107:4012
  -- Executing [110@internal:3] Playback("PJSIP/112-00000008", "for-service&press-1&for-sales&press-2") in new stack
  -- <PJSIP/112-00000008> Playing 'for-service.ulaw' (language 'en')
     > 0x7f5654010aa0 -- Strict RTP switching to RTP target address 192.168.6.107:4012 as source
  -- <PJSIP/112-00000008> Playing 'press-1.ulaw' (language 'en')
  -- <PJSIP/112-00000008> Playing 'for-sales.ulaw' (language 'en')
  -- <PJSIP/112-00000008> Playing 'press-2.ulaw' (language 'en')
  -- Executing [110@internal:4] Read("PJSIP/112-00000008", "VAR,,,,,4") in new stack
     > 0x7f5654010aa0 -- Strict RTP learning complete - Locking on source address 192.168.6.107:4012
  -- User entered '2'
  -- Executing [110@internal:5] GotoIf("PJSIP/112-00000008", "0?111,1") in new stack
  -- Executing [110@internal:6] GotoIf("PJSIP/112-00000008", "1?112,1:110,1") in new stack
  -- Goto (internal,112,1)
  -- Executing [112@internal:1] NoOp("PJSIP/112-00000008", "For Sales Calling From ") in new stack
  -- Executing [112@internal:2] Answer("PJSIP/112-00000008", "") in new stack
  -- Executing [112@internal:3] SayDigits("PJSIP/112-00000008", "2") in new stack
  -- <PJSIP/112-00000008> Playing 'digits/2.ulaw' (language 'en')
  -- Executing [112@internal:4] Playback("PJSIP/112-00000008", "for-sales") in new stack
  -- <PJSIP/112-00000008> Playing 'for-sales.ulaw' (language 'en')
  -- Executing [112@internal:5] Dial("PJSIP/112-00000008", "PJSIP/112,10") in new stack
  -- Called PJSIP/112
  -- PJSIP/112-00000009 is ringing
== Everyone is busy/congested at this time (1:1/0/0)
  -- Executing [112@internal:6] Hangup("PJSIP/112-00000008", "") in new stack
== Spawn extension (internal, 112, 6) exited non-zero on 'PJSIP/112-00000008'

Step 8: Make the call to check ivr menu by using the below command. snap 3

-- Executing [110@internal:1] NoOp("PJSIP/112-0000000a", "Calling 110 From Caller ") in new stack
 -- Executing [110@internal:2] Answer("PJSIP/112-0000000a", "") in new stack
    > 0x7f5654010aa0 -- Strict RTP learning after remote address set to: 192.168.6.107:4016
 -- Executing [110@internal:3] Playback("PJSIP/112-0000000a", "for-service&press-1&for-sales&press-2") in new stack
 -- <PJSIP/112-0000000a> Playing 'for-service.ulaw' (language 'en')
    > 0x7f5654010aa0 -- Strict RTP switching to RTP target address 192.168.6.107:4016 as source
 -- <PJSIP/112-0000000a> Playing 'press-1.ulaw' (language 'en')
 -- <PJSIP/112-0000000a> Playing 'for-sales.ulaw' (language 'en')
 -- <PJSIP/112-0000000a> Playing 'press-2.ulaw' (language 'en')
 -- Executing [110@internal:4] Read("PJSIP/112-0000000a", "VAR,,,,,4") in new stack
    > 0x7f5654010aa0 -- Strict RTP learning complete - Locking on source address 192.168.6.107:4016
 -- User entered '3'
 -- Executing [110@internal:5] GotoIf("PJSIP/112-0000000a", "0?111,1") in new stack
 -- Executing [110@internal:6] GotoIf("PJSIP/112-0000000a", "0?112,1:110,1") in new stack
 -- Goto (internal,110,1)
 -- Executing [110@internal:1] NoOp("PJSIP/112-0000000a", "Calling 110 From Caller ") in new stack
 -- Executing [110@internal:2] Answer("PJSIP/112-0000000a", "") in new stack
 -- Executing [110@internal:3] Playback("PJSIP/112-0000000a", "for-service&press-1&for-sales&press-2") in new stack
 -- <PJSIP/112-0000000a> Playing 'for-service.ulaw' (language 'en')
 -- <PJSIP/112-0000000a> Playing 'press-1.ulaw' (language 'en')
 -- <PJSIP/112-0000000a> Playing 'for-sales.ulaw' (language 'en')
 -- <PJSIP/112-0000000a> Playing 'press-2.ulaw' (language 'en')
 -- Executing [110@internal:4] Read("PJSIP/112-0000000a", "VAR,,,,,4") in new stack
 -- User entered nothing.
 -- Executing [110@internal:5] GotoIf("PJSIP/112-0000000a", "0?111,1") in new stack
 -- Executing [110@internal:6] GotoIf("PJSIP/112-0000000a", "0?112,1:110,1") in new stack
 -- Goto (internal,110,1)
 -- Executing [110@internal:1] NoOp("PJSIP/112-0000000a", "Calling 110 From Caller ") in new stack
 -- Executing [110@internal:2] Answer("PJSIP/112-0000000a", "") in new stack
 -- Executing [110@internal:3] Playback("PJSIP/112-0000000a", "for-service&press-1&for-sales&press-2") in new stack
 -- <PJSIP/112-0000000a> Playing 'for-service.ulaw' (language 'en')
 -- <PJSIP/112-0000000a> Playing 'press-1.ulaw' (language 'en')
 -- <PJSIP/112-0000000a> Playing 'for-sales.ulaw' (language 'en')
 -- <PJSIP/112-0000000a> Playing 'press-2.ulaw' (language 'en')
 -- Executing [110@internal:4] Read("PJSIP/112-0000000a", "VAR,,,,,4") in new stack
 -- User disconnected

Conclusion:

We have reached the end of this article. In this guide, we have walked you through the steps required to create a basic Asterisk IVR on Debian 12. Your feedback is much welcome.

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Frequently asked questions ( 5 )

How do I create a basic IVR menu with options for callers to choose from?

This is a common starting point for many Asterisk IVR setups. It involves using the Read application to capture user input and Goto or GotoIf commands to direct calls based on selected options.

Can I integrate my IVR with a database to access or update information?

Yes, Asterisk can connect to databases through ODBC or DBI modules. This allows storing and retrieving information like customer details or dynamic menu options during the IVR interaction.

How do I handle situations where callers enter invalid input or don't respond?

Error handling and retries are crucial for a smooth IVR experience. Asterisk offers tools like GotoIf, Wait, and Playback to handle invalid inputs, provide error messages, and offer repeat options.

Can I use speech recognition with my IVR for hands-free interaction?

Yes, Asterisk can integrate with speech recognition engines like Google Speech-to-Text or Sphinx. This allows callers to navigate options or provide information using their voice.

How do I secure my IVR system and prevent unauthorized access?

Security is vital for any phone system, especially when dealing with sensitive information. Asterisk offers password protection, access control lists, and encryption options to secure your IVR environment.