How to use Asterisk Goto and GotoIf application on Debian 12

To Use Asterisk Goto And GotoIf Application On Debian 12

Introduction:

Goto allows jumping to a specific context, extension, or priority, while GotoIf enables conditional branching based on certain conditions, providing flexibility in call routing.

Procedure:

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: Go to Following Location by using the below command.

root@linuxhelp:~# cd /etc/asterisk

Step 3: Open and Check extensions.conf file for dialplan configuration by using the below command.

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

exten => 110,1,NoOp(Print the context)
same => n,Dial(PJSIP/110,10)
same => n,Playback(sorry_didnt_get)
same => n,Hangup()

exten => 111,1,NoOp(Print the context)
same => n,Dial(PJSIP/111, 20)
same => n,Playback(sorry_didnt_get)
same => n,Hangup()

exten => 112,1,NoOp(Print the context)
same => n,Dial(PJSIP/112, 20)
same => n,Playback(sorry_didnt_get)
same => n,Hangup()

Step 4: Open another tab in the terminal as shown in below image.

Step 5: Login to the Asterisk Console by using following command.

root@linuxhelp:~# asterisk -rvvvvvvvvvvvvvvvvvvvv
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 = 1043)
linuxhelp*CLI>

Step 6: Reload the dialplan by using the below command.

linuxhelp*CLI> dialplan reload
Dialplan reloaded.
    -- Time to scan old dialplan and merge leftovers back into the new: 0.000080 sec
    -- Time to restore hints and swap in new dialplan: 0.000085 sec
    -- Time to delete the old dialplan: 0.000032 sec
    -- Total time merge_contexts_delete: 0.000197 sec
    -- pbx_config successfully loaded 1 contexts (enable debug for details).

Step 7: Check the connections by making call by using MicroSIP Software as shown in below image.

Step 8: Open and edit the extensions.conf file for use Goto application by using the below command.

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

exten => 110,1,NoOp(Print the context)
same => n,Goto(111,2)
same => n,Dial(PJSIP/110,10)
same => n,Playback(sorry_didnt_get)
same => n,Hangup()

exten => 111,1,NoOp(Print the context)
same => n,Dial(PJSIP/111, 20)
same => n,Playback(sorry_didnt_get)
same => n,Hangup()

exten => 112,1,NoOp(Print the context)
same => n,Dial(PJSIP/112, 20)
same => n,Playback(sorry_didnt_get)
same => n,Hangup()

Step 9: Goto the asterisk console and reload dialplan by using the below command.

linuxhelp*CLI> dialplan reload
Dialplan reloaded.
    -- Time to scan old dialplan and merge leftovers back into the new: 0.000032 sec
    -- Time to restore hints and swap in new dialplan: 0.000005 sec
    -- Time to delete the old dialplan: 0.000004 sec
    -- Total time merge_contexts_delete: 0.000041 sec
    -- pbx_config successfully loaded 1 contexts (enable debug for details).

Step 10: Again, Check the connections by making call by using MicroSIP Software as shown in below image.

Step 11: Again, Open and Edit the extensions.conf file for use GotoIf application by using the below command.

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

exten => 110,1,NoOp(Print the context)
same => n,SET(TEST=1)
same => n,GotoIf($[ ${TEST} = 1]?111,2:110,4)
same => n,Dial(PJSIP/110,10)
same => n,Playback(sorry_didnt_get)
same => n,Hangup()

exten => 111,1,NoOp(Print the context)
same => n,Dial(PJSIP/111, 20)
same => n,Playback(sorry_didnt_get)
same => n,Hangup()

exten => 112,1,NoOp(Print the context)
same => n,Dial(PJSIP/112, 20)
same => n,Playback(sorry_didnt_get)
same => n,Hangup()

Step 12: Goto the asterisk console and reload dialplan by using the below command.

linuxhelp*CLI> dialplan reload
Dialplan reloaded.
    -- Time to scan old dialplan and merge leftovers back into the new: 0.000005 sec
    -- Time to restore hints and swap in new dialplan: 0.000005 sec
    -- Time to delete the old dialplan: 0.000004 sec
    -- Total time merge_contexts_delete: 0.000014 sec
    -- pbx_config successfully loaded 1 contexts (enable debug for details

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

exten => 110,1,NoOp(Print the context)
same => n,SET(TEST=2)
same => n,GotoIf($[ ${TEST} = 1]?111,2:110,4)
same => n,Dial(PJSIP/110,10)
same => n,Playback(sorry_didnt_get)
same => n,Hangup()

exten => 111,1,NoOp(Print the context)
same => n,Dial(PJSIP/111, 20)
same => n,Playback(sorry_didnt_get)
same => n,Hangup()

exten => 112,1,NoOp(Print the context)
same => n,Dial(PJSIP/112, 20)
same => n,Playback(sorry_didnt_get)
same => n,Hangup()

linuxhelp*CLI> dialplan reload
Dialplan reloaded.
    -- Time to scan old dialplan and merge leftovers back into the new: 0.000005 sec
    -- Time to restore hints and swap in new dialplan: 0.000005 sec
    -- Time to delete the old dialplan: 0.000004 sec
    -- Total time merge_contexts_delete: 0.000014 sec
    -- pbx_config successfully loaded 1 contexts (enable debug for details).