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How to create an Asterisk Dialplan Context on Oracle Linux 9.3

To Create An Asterisk Dialplan Context On Oracle Linux9.3

Introduction:

The Dialplan is truly the heart of any Asterisk system, as it defines how Asterisk handles inbound and outbound calls. In a nutshell, it consists of a list of instructions or steps that Asterisk will follow. Unlike traditional phone systems, Asterisk’s Dialplan is fully customizable. To successfully set up your own Asterisk system, you will need to understand the Dialplan.

Procedure:

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

[root@linuxhelp ~]# cat /etc/os-release
NAME="Oracle Linux Server"
VERSION="9.3"
ID="ol"
ID_LIKE="fedora"
VARIANT="Server"
VARIANT_ID="server"
VERSION_ID="9.3"
PLATFORM_ID="platform:el9"
PRETTY_NAME="Oracle Linux Server 9.3"
ANSI_COLOR="0;31"
CPE_NAME="cpe:/o:oracle:linux:9:3:server"
HOME_URL="https://linux.oracle.com/"
BUG_REPORT_URL="https://github.com/oracle/oracle-linux"
ORACLE_BUGZILLA_PRODUCT="Oracle Linux 9"
ORACLE_BUGZILLA_PRODUCT_VERSION=9.3
ORACLE_SUPPORT_PRODUCT="Oracle Linux"
ORACLE_SUPPORT_PRODUCT_VERSION=9.3

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

[root@linuxhelp ~]# systemctl status asterisk.service
● asterisk.service - LSB: Asterisk PBX
     Loaded: loaded (/etc/rc.d/init.d/asterisk; generated)
     Active: active (running) since Mon 2024-02-19 04:02:05 IST; 3 days ago
       Docs: man:systemd-sysv-generator(8)
    Process: 1001 ExecStart=/etc/rc.d/init.d/asterisk start (code=exited, status=0/SUCCESS)
   Main PID: 1062
      Tasks: 37 (limit: 21739)
     Memory: 64.9M
        CPU: 22.697s
     CGroup: /system.slice/asterisk.service
             ├─1056 /bin/sh /usr/sbin/safe_asterisk
             └─5831 /usr/sbin/asterisk -f -vvvg -c

Feb 19 04:02:05 linuxhelp systemd[1]: Starting LSB: Asterisk PBX...
Feb 19 04:02:05 linuxhelp asterisk[1001]: Starting asterisk:
Feb 19 04:02:05 linuxhelp systemd[1]: asterisk.service: Can't open PID file /run/asterisk/asterisk.pid (yet?) after start: Operation not permitted
Feb 19 04:02:05 linuxhelp systemd[1]: asterisk.service: Supervising process 1062 which is not our child. We'll most likely not notice when it exits.
Feb 19 04:02:05 linuxhelp systemd[1]: Started LSB: Asterisk PBX.

Step 3 : Login to the Asterisk CLI mode by using the below command

[root@linuxhelp ~]# asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvv
Asterisk 20.5.0, 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 20.5.0 currently running on linuxhelp (pid = 5831)
    -- Remote UNIX connection
linuxhelp*CLI>

Step 4 : Show the endpoints by using the below command

linuxhelp*CLI> pjsip show endpoints
 Endpoint:  <Endpoint/CID.....................................>  <State.....>  <Channels.>
    I/OAuth:  <AuthId/UserName...........................................................>
        Aor:  <Aor............................................>  <MaxContact>
      Contact:  <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..>
  Transport:  <TransportId........>  <Type>  <cos>  <tos>  <BindAddress..................>
   Identify:  <Identify/Endpoint.........................................................>
        Match:  <criteria.........................>
    Channel:  <ChannelId......................................>  <State.....>  <Time.....>
        Exten: <DialedExten...........>  CLCID: <ConnectedLineCID.......>
==========================================================================================
 Endpoint:  167/167                                              Unavailable   0 of inf
     InAuth:  167/167
        Aor:  167                                                1
  Transport:  tcp_transport             tcp      3     96  0.0.0.0:5060
 Endpoint:  168/168                                              Unavailable   0 of inf
     InAuth:  168/168
        Aor:  168                                                1
  Transport:  tcp_transport             tcp      3     96  0.0.0.0:5060
Objects found: 2

Step 5 : Exit from the Asterisk CLI mode by using the below command

linuxhelp*CLI> exit
Asterisk cleanly ending (0).
Executing last minute cleanups

Step 6: Edit the extensions.conf file to modify the dialplan by using the below command.

[root@linuxhelp ~]# vim /etc/asterisk/extensions.conf
Add the following Lines
[internal]

exten => _XXX,1,NoOp(Call started for ${EXTEN})
exten => _XXX,n,Dial(PJSIP/${EXTEN},10)
exten => _XXX,n,Playback()
exten => _XXX,n,Hangup()


;exten => 167,1,NoOp(Call started for ${EXTEN})
;exten => 167,2,Dial(PJSIP/167,10)
;exten => 167,3,Hangup()

;exten => 168,1,Dial(PJSIP/168)

Step 7: Again login to the Asterisk CLI mode by using the below command.

[root@linuxhelp ~]# asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvv
Asterisk 20.5.0, 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 20.5.0 currently running on linuxhelp (pid = 7289)
linuxhelp*CLI>

Step 8 : 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: 0.000004 sec
    -- Time to restore hints and swap in new dialplan: 0.000006 sec
    -- Time to delete the old dialplan: 0.000003 sec
    -- Total time merge_contexts_delete: 0.000013 sec
    -- pbx_config successfully loaded 1 contexts (enable debug for details).


Result
Connected to Asterisk 20.5.0 currently running on linuxhelp (pid = 7289)
    -- Executing [168@internal:1] NoOp("PJSIP/167-00000000", "Call started for 168") in new stack
    -- Executing [168@internal:2] Dial("PJSIP/167-00000000", "PJSIP/168,10") in new stack
    -- Called PJSIP/168
    -- PJSIP/168-00000001 is ringing
    -- Nobody picked up in 10000 ms
    -- Executing [168@internal:3] Playback("PJSIP/167-00000000", "") in new stack
[Feb 23 00:34:48] WARNING[7522][C-00000001]: app_playback.c:464 playback_exec: Playback requires an argument (filename)
  == Spawn extension (internal, 168, 3) exited non-zero on 'PJSIP/167-00000000'

Step 9: Exit from the Asterisk CLI mode by using the below command

linuxhelp*CLI> exit
Asterisk cleanly ending (0).
Executing last minute cleanups

Step 10: Move to the directory to find the sounds for playback by using the below command

[root@linuxhelp ~]# cd /var/lib/asterisk/sounds/en
[root@linuxhelp en]# ls
1-for-am-2-for-pm.alaw                   easterly.siren7                              north-dakota.wav                                     step-in-stream.g729
1-for-am-2-for-pm.g722                   easterly.sln16                               northerly.alaw                                       step-in-stream.gsm
1-for-am-2-for-pm.g729                   easterly.ulaw                                northerly.g722                                       step-in-stream.siren14
1-for-am-2-for-pm.gsm                    easterly.wav                                 northerly.g729                                       step-in-stream.siren7
1-for-am-2-for-pm.siren14                eastern.alaw                                 northerly.gsm                                        step-in-stream.sln16
1-for-am-2-for-pm.siren7                 eastern.g722                                 northerly.siren14                                    step-in-stream.ulaw
1-for-am-2-for-pm.sln16                  eastern.g729                                 northerly.siren7                                     step-in-stream.wav
1-for-am-2-for-pm.ulaw                   eastern.gsm                                  northerly.sln16                                      sterling.alaw
1-for-am-2-for-pm.wav                    eastern.siren14                              northerly.ulaw                                       sterling.g722
1-yes-2-no.alaw                          eastern.siren7                               northerly.wav                                        sterling.g729
1-yes-2-no.g722                          eastern.sln16                                north.g722                                           sterling.gsm
1-yes-2-no.g729                          eastern.ulaw                                 north.g729                                           sterling.siren14
1-yes-2-no.gsm                           eastern.wav                                  north.gsm                                            sterling.siren7
1-yes-2-no.siren14                       east.g722                                    north.siren14                                        sterling.sln16
1-yes-2-no.siren7                        east.g729                                    north.siren7                                         sterling.ulaw
1-yes-2-no.sln16                         east.gsm                                     north.sln16                                          sterling.wav
1-yes-2-no.ulaw                          east.siren14                                 north.ulaw                                           still_on_phone.alaw
1-yes-2-no.wav                           east.siren7                                  north.wav                                            still_on_phone.g722
abandon-all-hope.alaw                    east.sln16                                   not-auth-pstn.alaw                                   still_on_phone.g729
abandon-all-hope.g722                    east.ulaw                                    not-auth-pstn.g722                                   still_on_phone.gsm
abandon-all-hope.g729                    east.wav                                     not-auth-pstn.g729                                   still_on_phone.siren14
abandon-all-hope.gsm                     echo-test.alaw                               not-auth-pstn.gsm                                    still_on_phone.siren7
abandon-all-hope.siren14                 echo-test.g722                               not-auth-pstn.siren14                                still_on_phone.sln16
abandon-all-hope.siren7                  echo-test.g729                               not-auth-pstn.siren7                                 still_on_phone.ulaw
abandon-all-hope.sln16                   echo-test.gsm                                not-auth-pstn.sln16                                  still_on_phone.wav
abandon-all-hope.ulaw                    echo-test.siren14                            not-auth-pstn.ulaw                                   store-accounting.alaw
abandon-all-hope.wav                     echo-test.siren7                             not-auth-pstn.wav                                    store-accounting.g722
abandons.alaw                            echo-test.sln16                              not-enough-credit.alaw                               store-accounting.g729
abandons.g722                            echo-test.ulaw                               not-enough-credit.g722                               store-accounting.gsm
abandons.g729                            echo-test.wav                                not-enough-credit.g729                               store-accounting.siren14
abandons.gsm                             ed.alaw                                      not-enough-credit.gsm                                store-accounting.siren7
abandons.siren14                         ed.g722                                      not-enough-credit.siren14                            store-accounting.sln16
abandons.siren7                          ed.g729                                      not-enough-credit.siren7                             store-accounting.ulaw
abandons.sln16                           ed.gsm                                       not-enough-credit.sln16                              store-accounting.wav
abandons.ulaw                            ed.siren14                                   not-enough-credit.ulaw                               storm.alaw
abandons.wav                             ed.siren7                                    not-enough-credit.wav                                storm.g722
academic-support.alaw                    ed.sln16                                     nothing-recorded.alaw                                storm.g729
academic-support.g722                    ed.ulaw                                      nothing-recorded.g722                                storm.gsm
academic-support.g729                    ed.wav                                       nothing-recorded.g729                                storm.siren14
academic-support.gsm                     eighteenth.alaw                              nothing-recorded.gsm                                 storm.siren7
academic-support.siren14                 eighteenth.g722                              nothing-recorded.siren14                             storm.sln16
academic-support.siren7                  eighteenth.g729                              nothing-recorded.siren7                              storm.ulaw
academic-support.sln16                   eighteenth.gsm                               nothing-recorded.sln16                               storm.wav
academic-support.ulaw                    eighteenth.siren14                           nothing-recorded.ulaw                                sun.alaw
academic-support.wav                     eighteenth.siren7                            nothing-recorded.wav                                 sun.g722
access-code.alaw                         eighteenth.sln16                             not_me.alaw                                          sun.g729
access-code.g722                         eighteenth.ulaw                              not_me.g722                                          sun.gsm
access-code.g729                         eighteenth.wav                               not_me.g729                                          sunny.alaw
access-code.gsm                          eighth.alaw                                  not_me.gsm                                           sunny.g722
access-code.siren14                      eighth.g722                                  not_me.siren14                                       sunny.g729

Step 11: Add any of the above sound to the extensions.conf file to add the playback sound to the channel by using the below command

[root@linuxhelp en]# vim /etc/asterisk/extensions.conf
[internal]
exten => _XXX,1,NoOp(Call started for ${EXTEN})
exten => _XXX,n,Dial(PJSIP/${EXTEN},10)
exten => _XXX,n,Playback(your-account)
exten => _XXX,n,Hangup()

;exten => 167,1,NoOp(Call started for ${EXTEN})
;exten => 167,2,Dial(PJSIP/167,10)
;exten => 167,3,Hangup()

;exten => 168,1,Dial(PJSIP/168)

Step 12 : Login to the Asterisk CLI mode and reload the Dialplan by using the below command

[root@linuxhelp en]# asterisk -rvvvvvvvvvvvvvvvv
Asterisk 20.5.0, 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 20.5.0 currently running on linuxhelp (pid = 7289)
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: 0.000006 sec
    -- Time to delete the old dialplan: 0.000003 sec
    -- Total time merge_contexts_delete: 0.000014 sec
    -- pbx_config successfully loaded 1 contexts (enable debug for details).


Result
Executing [167@internal:1] NoOp("PJSIP/168-00000006", "Call started for 167") in new stack
    -- Executing [167@internal:2] Dial("PJSIP/168-00000006", "PJSIP/167,10") in new stack
    -- Called PJSIP/167
    -- PJSIP/167-00000007 is ringing
    -- Nobody picked up in 10000 ms
    -- Executing [167@internal:3] Playback("PJSIP/168-00000006", "your-account") in new stack
       > 0x7ffa800391e0 -- Strict RTP learning after remote address set to: 192.168.6.102:4028
    -- <PJSIP/168-00000006> Playing 'your-account.ulaw' (language 'en')
       > 0x7ffa800391e0 -- Strict RTP switching to RTP target address 192.168.6.102:4028 as source
    -- Executing [167@internal:4] Hangup("PJSIP/168-00000006", "") in new stack
  == Spawn extension (internal, 167, 4) exited non-zero on 'PJSIP/168-00000006'

Conclusion:

We have reached the end of this article. In this guide, we have walked you through the steps required to create Asterisk Dialplan context on Oracle Linux 9.3. Your feedback is much welcome.

Tag : Asterisk Dial Linux
FAQ
Q
What is Auth in PJSIP?
A
Address of Record (AOR): According to the official Asterisk documentation, an AOR tells Asterisk "where an endpoint can be contacted."
Q
What is AOR in PJSIP?
A
Address of Record (AOR): According to the official Asterisk documentation, an AOR tells Asterisk "where an endpoint can be contacted."
Q
How to reload pjsip.conf?
A
By running the following command pjsip reload
Q
What is the default port used by PJSIP?
A
Default UDP port 5160
Q
What is PJSIP in Asterisk?
A
pjsip. conf is a flat text file composed of sections like most configuration files used with Asterisk.