Asterix open-source telephony Server can be used to build multitude of applications .
check the asterisk available version from https://downloads.asterisk.org/pub/telephony/asterisk/ use wget to download
wget https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-17.6.0.tar.gz
tar -xvf asterisk-17.6.0.tar.gz
libpri library allows Asterisk to communicate with ISDN connections.
DAHDI ( Digium Asterisk Hardware Device Interface) library allows Asterisk to communicate with analog and digital telephones and telephone lines / PSTN
dahdi-linux
wget https://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-current.tar.gz
tar -xvf dahdi-linux-current.tar.gz
dadhi-tools
wget https://downloads.asterisk.org/pub/telephony/dahdi-tools/dahdi-tools-current.tar.gz
tar -xvf dahdi-tools-current.tar.gz
Remember you make to make and install DAHDI and libpri before building asterisk
make
sudo make install
make config
SIP stack ina Asterisk
- chan_sip SIP channel driver ( old)
- chan_pjsip
Build with later using
cd asterisk-17.6.0
./configure --with-pjproject-bundled --with-jansson-bundled
output
config.status: creating makeopts
config.status: creating autoconfig.h
configure: Menuselect build configuration successfully completed
.$$$$$$$$$$$$$$$=..
.$7$7.. .7$$7:.
.$$:. ,$7.7
.$7. 7$$$$ .$$77
..$$. $$$$$ .$$$7
..7$ .?. $$$$$ .?. 7$$$.
$.$. .$$$7. $$$$7 .7$$$. .$$$.
.777. .$$$$$$77$$$77$$$$$7. $$$,
$$$~ .7$$$$$$$$$$$$$7. .$$$.
.$$7 .7$$$$$$$7: ?$$$.
$$$ ?7$$$$$$$$$$I .$$$7
$$$ .7$$$$$$$$$$$$$$$$ :$$$.
$$$ $$$$$$7$$$$$$$$$$$$ .$$$.
$$$ $$$ 7$$$7 .$$$ .$$$.
$$$$ $$$$7 .$$$.
7$$$7 7$$$$ 7$$$
$$$$$ $$$
$$$$7. $$ (TM)
$$$$$$$. .7$$$$$$ $$
$$$$$$$$$$$$7$$$$$$$$$.$$$$$$
$$$$$$$$$$$$$$$$.
configure: Package configured for:
configure: OS type : linux-gnu
configure: Host CPU : x86_64
configure: build-cpu:vendor:os: x86_64 : pc : linux-gnu :
configure: host-cpu:vendor:os: x86_64 : pc : linux-gnu :
USe menu select option on Asterisk to install packages pick and choose application and other options such as music on hold , extra sound packages , PBX modules etc
make menuselect
output
**************************************************
Asterisk Module and Build Option Selection
**************************************************
Press 'h' for help.
Add-ons (See README-addons.txt)
Applications
Bridging Modules
Call Detail Recording
Channel Event Logging
Channel Drivers
Codec Translators
Format Interpreters
Dialplan Functions
PBX Modules
Resource Modules
Test Modules
Compiler Flags
Utilities
AGI Samples
Core Sound Packages
Music On Hold File Packages
---> Extras Sound Packages
Compile asterisk on the system
make
output
Building Documentation For: third-party channels pbx apps codecs formats cdr cel bridges funcs tests main res addons
+--------- Asterisk Build Complete ---------+
+ Asterisk has successfully been built, and +
+ can be installed by running: +
+ +
+ make install +
+-------------------------------------------+
Install
sudo make install
output
done
+---- Asterisk Installation Complete -------+
+ +
+ YOU MUST READ THE SECURITY DOCUMENT +
+ +
+ Asterisk has successfully been installed. +
+ If you would like to install the sample +
+ configuration files (overwriting any +
+ existing config files), run: +
+ +
+ For generic reference documentation: +
+ make samples +
+ +
+ For a sample basic PBX: +
+ make basic-pbx +
+ +
+ +
+----------------- or ---------------------+
Install samples
sudo make samples
Run Asterisk
➜ asterisk-17.6.0 sudo asterisk -c
Asterisk 17.6.0, Copyright (C) 1999 - 2018, Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
...
Asterisk Ready.
*CLI>
ports uses
> ps -ef | grep asterisk
asterisk 28922 altanai 5u IPv4 3449570 0t0 TCP localhost:38108->localhost:postgresql (CLOSE_WAIT)
asterisk 28922 altanai 9u IPv4 3451466 0t0 UDP *:40388
asterisk 28922 altanai 10u IPv6 3451467 0t0 UDP *:52092
asterisk 28922 altanai 12u IPv4 3449574 0t0 UDP *:2727
asterisk 28922 altanai 14u IPv4 3449575 0t0 UDP *:iax
asterisk 28922 altanai 15u IPv4 3451995 0t0 TCP *:cisco-sccp (LISTEN)
asterisk 28922 altanai 18u IPv4 3449577 0t0 UDP *:5000
asterisk 28922 altanai 19u IPv4 3449578 0t0 UDP *:4520
You can use safe_asterisk to enable auto restart after crash thus minimizing downtime. It also creates core dump file .
One can continue on the sandbox environment to run the modules I describe in this repo.
show version
*CLI> core show version
Asterisk 17.6.0 built by root @ altanai-Inspiron-15-5578 on a x86_64 running Linux on 2020-07-30 15:33:43 UTC
Asterisk is an open source framework for building communications applications.
Default installation paths Path Description /etc/asterisk Configuration files /usr/sbin Location of binary executable /var/log/asterisk message(error) logs and CDR /usr/lib/asterisk/modules Component module libraries
Default ports
Protocol Port number Transport SIP 5060/5061 TCP/UDP IAX2 4569 UDP MGCP 2727 UDP SCCP 2000 TCP RTP 10,00 – 20,000 UDP Manager 5038 TCP H323 1720 TCP Dundi 4520 UDP Unistim 5000 UDP
Listen address for SIP in sip.conf , Listen on
specific IPv4 address. Example: bindaddr=192.0.2.1 specific IPv6 address. Example: bindaddr=2001:db8::1 IPv4 wildcard. Example: bindaddr=0.0.0.0 IPv4 and IPv6 wildcards. Example: bindaddr=::
asterisk/modules support teh modular structure of Asterisk
Custom applications such as res_musiconhold.so
convert media on disk and channel
Interface with asterisk and issue commands
Other interface for asterisk control are AMI ( Asterisk AManger Interface) and AGI ( Asterisk Gateway Interface ) which operate on APIs like PHP , C++ , Java , Perl
Reload SIP
*CLI> pjsip reload
[Jul 31 09:54:58] NOTICE[22399]: sorcery.c:1345 sorcery_object_load: Type 'system' is not reloadable, maintaining previous values
Module 'res_pjsip.so' reloaded successfully.
Module 'res_pjsip_authenticator_digest.so' reloaded successfully.
Module 'res_pjsip_endpoint_identifier_ip.so' reloaded successfully.
Module 'res_pjsip_mwi.so' reloaded successfully.
Module 'res_pjsip_notify.so' reloaded successfully.
Module 'res_pjsip_outbound_publish.so' reloaded successfully.
Module 'res_pjsip_publish_asterisk.so' reloaded successfully.
Module 'res_pjsip_outbound_registration.so' reloaded successfully.
Settings show
*CLI> core show settings
PBX Core settings
-----------------
Version: 17.6.0
Build Options: BUILD_NATIVE, OPTIONAL_API
Maximum calls: Not set
Maximum open file handles: 1024
Root console verbosity: 0
Current console verbosity: 0
Debug level: 0
Trace level: 0
Maximum load average: 0.000000
Minimum free memory: 0 MB
Startup time: 10:33:28
Last reload time: 10:33:28
System: Linux/4.15.0-62-generic built by root on x86_64 2020-07-30 15:33:43 UTC
System name:
Entity ID: 7c:67:a2:eb:ff:a5
PBX UUID: fc3a2e05-e800-47e7-aa27-36ad924f85e0
Default language: en
Language prefix: Enabled
User name and group: /
Executable includes: Disabled
Transcode via SLIN: Enabled
Transmit silence during rec: Disabled
Generic PLC: Enabled
Generic PLC on equal codecs: Disabled
Hide Msg Chan AMI events: Disabled
Min DTMF duration:: 80
Cache media frames: Enabled
RTP use dynamic payloads: 1
RTP dynamic payload types: 35-63,96-127
* Subsystems
-------------
Manager (AMI): Disabled
Web Manager (AMI/HTTP): Disabled
Call data records: Enabled
Realtime Architecture (ARA): Disabled
* Directories
-------------
Configuration file: /etc/asterisk/asterisk.conf
Configuration directory: /etc/asterisk
Module directory: /usr/lib/asterisk/modules
Spool directory: /var/spool/asterisk
Log directory: /var/log/asterisk
Run/Sockets directory: /var/run/asterisk
PID file: /var/run/asterisk/asterisk.pid
VarLib directory: /var/lib/asterisk
Data directory: /var/lib/asterisk
ASTDB: /var/lib/asterisk/astdb
IAX2 Keys directory: /var/lib/asterisk/keys
AGI Scripts directory: /var/lib/asterisk/agi-bin
channel types
*CLI> core show channeltypes
Type Description Devicestate Presencestate Indications Transfer
------------- ------------- ------------- ------------- ------------- -------------
Recorder Bridge Media Recording Channel Driver no no yes no
Announcer Bridge Media Announcing Channel Driver no no yes no
Phone Standard Linux Telephony API Driver no no yes no
Console OSS Console Channel Driver no no yes no
USTM UNISTIM Channel Driver no no yes no
CBAnn Conference Bridge Announcing Channel no no yes no
CBRec Conference Bridge Recording Channel no no no no
PJSIP PJSIP Channel Driver yes no yes yes
UnicastRTP Unicast RTP Media Channel Driver no no no no
MulticastRTP Multicast RTP Paging Channel Driver no no no no
Skinny Skinny Client Control Protocol (Skinny) yes no yes no
IAX2 Inter Asterisk eXchange Driver (Ver 2) yes no yes yes
MGCP Media Gateway Control Protocol (MGCP) yes no yes no
Local Local Proxy Channel Driver yes no yes no
Surrogate Surrogate channel used to pull channel f no no no no
----------
15 channel drivers registered.
Set verbosity level ( 0-4)
*CLI> core set verbose 1
Console verbose was OFF and is now 1.
Custom application that can be built - Call Queue monitoring
Add a dialplan in extensions
[internal_users]
exten => 6000,1,Answer()
exten => 6000,2,Wait(1)
exten => 6000,2,Wait(1)
exten => 6000,3,Playback(hello-world)
exten => 6000,4,Hangup()
Reload and see the dialplan
altanai-Inspiron-15-5578*CLI> dialplan reload
Dialplan reloaded.
altanai-Inspiron-15-5578*CLI> dialplan show
...
[ Context 'internal_users' created by 'pbx_config' ]
'6000' => 1. Answer() [extensions.conf:866]
2. Wait(1) [extensions.conf:867]
3. Playback(hello-world) [extensions.conf:868]
4. Hangup() [extensions.conf:869]
During Call
*CLI> == Setting global variable 'SIPDOMAIN' to '192.168.1.114'
== Spawn extension (internal_users, 6000, 5) exited non-zero on 'PJSIP/7000-00000000'
Dial two Calls
extensions.conf
exten => 6001,1,Dial(SIP/phone-2,20)
exten => 6001,2,Verbose("---------------- Forward to phone2 -------")
reload dialplan and validate that its taken effect
*CLI> dialplan show 6001@internal_users
[ Context 'internal_users' created by 'pbx_config' ]
'6001' => 1. Dial(SIP/phone-2,20) [extensions.conf:872]
2. Verbose("---------------- Forward to phone2 -------") [extensions.conf:873]
-= 1 extension (2 priorities) in 1 context. =-
Dial 6001 and see channels while call is active
core show channels
Channel Location State Application(Data)
0 active channels
0 active calls
1 call processed
Note that the sounds for asterisk playback , background and various other applications are loaded in /var/lib/asterisk/sounds/en/
More core and extra sounds can be downloaded from https://www.asterisksounds.org/en/download in various other languages like german apnish italian etc such as
wget https://www.asterisksounds.org/sites/asterisksounds.org/files/sounds/en/download/asterisk-sounds-extra-en-2.9.15.zip
Issue1 Cannot find sip commands
altanai-Inspiron-15-5578*CLI> sip show peers
No such command 'sip show peers' (type 'core show help sip show' for other possible commands)
solution First check is module is installed
module show like chan_sip.so
When module is installed
altanai-Inspiron-15-5578*CLI> module show like chan_sip.so
Module Description Use Count Status Support Level
chan_sip.so Session Initiation Protocol (SIP) 0
1 modules loadedwhen modules is not installed
altanai-Inspiron-15-5578*CLI> module show like chan_sip.so
Module Description Use Count Status Support Level
0 modules loadedIf module is not loaded Make sure Asterisk is configured to load the module via modules.conf There may be a case that res_pjsip is loaded inplace of chan_sip
; Do not load chan_sip by default, it may conflict with res_pjsip.
noload => chan_sip.so
In that case chan_pjsip would be loaded
altanai-Inspiron-15-5578*CLI> module show like chan_pjsip
Module Description Use Count Status Support Level
chan_pjsip.so PJSIP Channel Driver 0 Running core
1 modules loaded
On forcing loading of chan_sip module
altanai-Inspiron-15-5578*CLI> module load chan_sip
Loaded chan_sip
SIP channel loading...
[Jul 31 13:25:30] WARNING[815]: chan_sip.c:35477 deprecation_notice: chan_sip has no official maintainer and is deprecated. Migration to
[Jul 31 13:25:30] WARNING[815]: chan_sip.c:35478 deprecation_notice: chan_pjsip is recommended. See guides at the Asterisk Wiki:
[Jul 31 13:25:30] WARNING[815]: chan_sip.c:35479 deprecation_notice: https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
[Jul 31 13:25:30] WARNING[815]: chan_sip.c:35480 deprecation_notice: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip
Issue3 Unable to authenticate
*CLI> m[Aug 1 10:07:20] NOTICE[11076]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'REGISTER' from '<sip:[email protected]>' failed for '127.0.0.1:56039' (callid: OgfL6JO~OO) - No matching endpoint found
[Aug 1 10:07:25] NOTICE[11076]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'INVITE' from '<sip:[email protected]>' failed for '127.0.0.1:44672' (callid: c62tK1JzlDnDS9VFDGMQfg..) - No matching endpoint found
[Aug 1 10:07:25] NOTICE[11076]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'INVITE' from '<sip:[email protected]>' failed for '127.0.0.1:44672' (callid: c62tK1JzlDnDS9VFDGMQfg..) - No matching endpoint found
[Aug 1 10:07:25] NOTICE[11076]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'INVITE' from '<sip:[email protected]>' failed for '127.0.0.1:44672' (callid: c62tK1JzlDnDS9VFDGMQfg..) - Failed to authenticate
solution Define the extension context internal_users
extensions.conf
[internal_users]
exten => 6000,1,Answer()
exten => 6000,2,Wait(1)
exten => 6000,3,Playback(hello-world)
exten => 6000,4,Hangup()
and edit sip.conf or pjsip.conf and reload
sip.conf
[7000]
type=friend
host=dynamic
context=internal_users
secret=1234
pjsip.conf
[7000]
type=endpoint
context=internal_users
disallow=all
allow=ulaw
transport=transport-udp
auth=7000
aors=7000
[7000]
type=auth
auth_type=userpass
password=1234
username=7000
[7000]
type=aor
max_contacts=1
reload
sip reload
pjsip reload
Issue3 Unable to play hello world via playback
solution check if application is registered
core show application playback
If registered
-= Info about application 'Playback' =-
[Synopsis]
Play a file.
[Description]
Plays back given filenames (do not put extension of wav/alaw etc). The playback
command answer the channel if no options are specified. If the file is
non-existant it will fail
This application sets the following channel variable upon completion:
${PLAYBACKSTATUS}: The status of the playback attempt as a text string.
SUCCESS
FAILED
See Also: Background (application) -- for playing sound files that are
interruptible
WaitExten (application) -- wait for digits from caller, optionally play music
on hold
[Syntax]
Playback(filename[&filename2[&...]][,options])
[Arguments]
options
Comma separated list of options
skip: Do not play if not answered
noanswer: Playback without answering, otherwise the channel will be
answered before the sound is played.
NOTE: Not all channel types support playing messages while still on hook.
[See Also]
Background(), WaitExten(), ControlPlayback(), stream file, control stream file,
If not registered
altanai-Inspiron-15-5578*CLI> core show application playback
Your application(s) is (are) not registered
Issue 4 Public / private address conflicts
solution Change bidn addr from 0.0.0.0 to private IP and reload
Issue5 While working on IVR , WaitExtern , Background and DTMF applications you may find the prompt and press sounds missing
[Aug 2 08:55:44] WARNING[25971][C-0000002d]: file.c:1262 ast_streamfile: Unable to open are-you-still-there (format (ulaw)): No such file or directory
[Aug 2 08:55:44] WARNING[25971][C-0000002d]: app_playback.c:497 playback_exec: Playback failed on PJSIP/7000-0000002b for are-you-still-there
or
[Aug 2 08:54:52] WARNING[25971][C-0000002d]: file.c:789 ast_openstream_full: File press-1 does not exist in any format
[Aug 2 08:54:52] WARNING[25971][C-0000002d]: file.c:1262 ast_streamfile: Unable to open press-1 (format (ulaw)): No such file or directory
[Aug 2 08:54:52] WARNING[25971][C-0000002d]: pbx_builtins.c:1175 pbx_builtin_background: ast_streamfile failed on PJSIP/7000-0000002b for press-1&or&press-2
solution Download sound files or make your own prompts
React
Typescript
Django
Laravel
Facebook
Microsoft
Google
Alibaba
D3
Tencent