Part 1 – Asterisk Architecture: A Definitive Guide to Understanding Asterisk

What is Asterisk?

Asterisk is an open-source Telephony Enterprise software that provides a complete platform for building communication/Telephony applications. It is widely used in VoIP (Voice over Internet Protocol) applications and supports various communication protocols such as SIP (Session Initiation Protocol), H.323, and IAX (Inter-Asterisk eXchange).

The architecture of Asterisk is modular and flexible, allowing users to customize and extend the platform to fit their specific communication needs. The architecture can be divided into four main components:

• Asterisk Core
  • Channels drivers
  • Applications
  • Add-ons

Asterisk Core

The Asterisk Core is the heart of the Asterisk architecture. It provides the fundamental functionality for processing all modules that makes asterisk functional like call handling, routing, and media processing. The Core includes the call manager, which manages incoming and outgoing calls, the media manager, which manages audio and video streams, and the channel driver interface, which provides an abstraction layer for communication protocols.

Here are the key components included in the Asterisk Core:

1. Channel Drivers: Asterisk Core includes a set of channel drivers that enable communication with various types of endpoints and networks. These drivers facilitate connectivity with traditional telephony interfaces (Analogue, Digital, and ISDN), Voice over IP (VoIP) protocols (SIP, IAX, H.323), and other communication technologies.
2. Dialplan: The Dialplan is a powerful scripting language in Asterisk that defines call routing and call handling logic. It allows you to specify how calls are processed, including call routing, call manipulation, and interaction with various modules and applications.
3. Call Control: Asterisk Core provides call control functions to manage incoming and outgoing calls, call setup, call termination, call transfer, call hold, call forwarding, call parking, and other call-related operations.
4. Media Handling: The Asterisk Core includes media handling capabilities for audio and video streams. It supports audio Codecs for voice transmission and can handle media processing tasks such as recording, playback, conferencing, mixing, and transcoding.
5. Application Interfaces: Asterisk Core offers APIs (Application Programming Interfaces) that allow developers to interact with the system and build custom applications. The APIs include the Asterisk Gateway Interface (AGI) for external script integration, the Asterisk Manager Interface (AMI) for remote management and control, and the Asterisk RESTful Interface (ARI) for building custom telephony applications.
6. Resource Management: Asterisk Core manages system resources such as channels, timers, and memory to optimize performance and handle concurrent operations effectively.
7. Logging and Debugging: Asterisk Core provides logging and debugging facilities to aid in troubleshooting and monitoring system behaviour. It allows you to configure log levels, log file locations, and gather diagnostic information for analysis.
8. Configuration Framework: Asterisk Core includes a flexible and extensive configuration framework that allows you to customize the behaviour of the system. It provides configuration files where you can define settings for various components and modules in Asterisk.

These components collectively form the Asterisk Core, providing the essential building blocks for creating robust and feature-rich communication applications and services.

Modules

Asterisk is built on modules. A module is a loadable component that interacts with the asterisk core API to provide specific functionality; such as a channel driver (for example, chan_pjsip.so) or a resource that allows connection to an external technology (such as func_odbc.so). Asterisk modules are loaded based on the parameters defined in the /etc/asterisk/modules.conf file.

It is actually possible to start Asterisk without any modules at all, although in this state it will not be capable of doing anything.

The Types of Modules in Asterisk

We will explore the different types of modules available in Asterisk. The types of modules in Asterisk include the following:

• Channel Driver Modules
• Call Detail Recording (CDR) Modules
• Channel Event Logging (CEL) Modules
• Codec modules
• Format Interpreter modules
• Application modules
• Bridging Modules
• DialPlan Function modules
• Resource Modules
• PBX Modules
• Addon Modules

For this article, We have labelled asterisk modules based on status, usage and popularity.

Popularity Legend

• Essential: This module is one you’ll never want to be without.
• Useful: This module is current, maintained, popular, and recommended.
• Usable: The module works but may be incomplete and/or is not recommended
• New: This module is new, it works but its completeness is unknown.
• Insignificant: Very Old modules. May be incompatible & has no community support.
• Unreliable: The module is new and is in testing. Not recommended for Production
• Deprecated: This module has been replaced by something considered superior.
• Limited: This module has limitations that may make it unsuitable to your requirements.

Channel Driver Modules

Without channel drivers, Asterisk would have no way to make or receive calls. Asterisk Core includes a set of channel drivers that enable communication with various types of endpoints and networks. These drivers facilitate connectivity with traditional telephony interfaces (Analogue, Digital, and ISDN), Voice over IP (VoIP) protocols (SIP, IAX, H.323), and other communication technologies. These modules are responsible for managing communication between the Asterisk server and the outside world.

Some examples of channel modules and their magnitude of importance include:

Name	Purpose	Status
chan_sip	The SIP (Session Initiation Protocol) channel driver for VoIP communication using the SIP protocol. (For newer asterisk installations, use chan_pjsip	Essential
chan_pjsip	The PJSIP channel driver, a more modern and flexible replacement for chan_sip, supporting SIP and additional protocols. See also: chan_sip vs chan_pjsip	Essential (for newer Installs)
chan_dahdi	Provides support for DAHDI (Digium Asterisk Hardware Device Interface), which is used to connect Asterisk to analogue and digital telephony networks. It consists of two parts. 1. The DAHDI-Linux project contains the individual board drivers for the supported hardware. 2. The DAHDI-Tools project contains an assortment of user space utilities that are used to set up and test the drivers. Image Source: Asterisk Wiki	Useful
chan_iax2	The IAX2 (Inter-Asterisk eXchange version 2) channel driver for communication between Asterisk servers and IAX-compatible endpoints.	Useful
chan_skinny	Skinny or SCCP (Skinny Client Control Protocol), is a channel driver in Asterisk that provides support for Cisco IP phones. It allows Asterisk to communicate and interact with Cisco Unified Communications Manager (formerly known as Cisco CallManager) using the Skinny protocol.	Useful

Call Detail Recording Modules

SIP (Session Initiation Protocol) is a protocol used to initiate, manage, and terminate real-time communication sessions over the Internet. SIP records provide more detailed information about a call, such as the codecs used, the quality of service, and the network information.

While the CDR is produced after the call ends, the SIP record is created when the call begins. CDRs are transferred to other systems for use in various business cases like billing and reporting.

While CDRs are focused on call management and analysis, SIP records provide a more comprehensive picture of the technical aspects of a call. Both types of records are valuable in their own right and can be used together to gain a complete understanding of your communications system.

The CDR modules, listed in Table 3.2., “Call detail recording modules”, are designed to facilitate as many methods of storing call detail records as possible. You can store CDRs in a file (default), a database, RADIUS, or Syslog.

Call detail records are not intended to be used in complex billing applications. If you require more control over billing and call reporting, you will want to look at channel event logging. The advantage of CDR is that it just works.

Name	Purpose	Popularity/Status
cdr_adaptive_odbc	Allows writing of CDRs through ODBC framework with ability to add custom fields	Useful
cdr_csv	Writes CDRs to disk as a comma-separated values file	Usable
cdr_custom	As above, but allows for the addition of custom fields	Useful
cdr_manager	Outputs CDRs to Asterisk Manager Interface (AMI)	Useful
cdr_odbc	Writes CDRs through ODBC framework	Usable
cdr_pgsql	Writes CDRs to PostgreSQL	Useful
cdr_radius	Writes CDRs to RADIUS	Usable—does not support custom fields
cdr_sqlite	Writes CDRs to SQLite2 database	Deprecated—use sqlite3_custom
cdr_sqlite3_custom	Writes CDRs to SQLite3 with custom fields	Useful
cdr_syslog	Writes CDRs to syslog	Useful
cdr_tds	Writes CDRs to Microsoft SQL or Sybase	Usable—requires an old version of libtds

Channel Event Logging Modules

Channel event logging provides much more powerful control over reporting of call activity. By the same token, it requires more careful planning of your dialplan, and by no means will it work automatically. Asterisk’s CEL modules are listed in Table 3.3, “Channel event logging modules”.

Name	Purpose	Popularity/Status
cel_custom	CEL to disk/file	Useful
cel_manager	CEL to AMI	Useful
cel_odbc	CEL to ODBC	Useful
cel_pgsql	CEL to PostgreSQL	Useful
cel_radius	CEL to RADIUS	Usable—does not support custom fields
cel_sqlite3_custom	CEL to Sqlite3	Useful
cel_tds	CEL to Microsoft SQL or Sybase	Usable—requires an old version of libtds

Codec modules

When two endpoints (such as phones or softphones) communicate over a VoIP system, they need to agree on the same codec to use for the call. A codec is a method of encoding and decoding audio or video data, and different codecs have different levels of compression and quality. Asterisk Codec modules are responsible for encoding and decoding audio and video signals. Asterisk supports a wide range of codecs, including G.711, G.729, and H.264. Codec modules are used to ensure that audio and video signals are transmitted in the most efficient and effective manner possible. Some examples of codec modules include:

Name	Purpose	Popularity
codec_alaw	A-law Pulse Code Modulated (PCM) codec is used all over the world on the PSTN (except Canada/USA). This codec (along with ulaw) should be enabled on all your channels	Essential (Excellent Quality)
codec_ulaw	Mu-law PCM codec used on PSTN in Canada/USA. It’s more formally written as μ-law, but not many people have the Greek letter μ on their keyboard, so it’s popularly written as ulaw. This is often the default codec and should be enabled on all your channels	Essential (Excellent Quality)
codec_dahdi	Utilizes proprietary Digium hardware transcoding card	Essential
codec_a_mu	A-law to mu-law direct converter	Useful
codec_opus	Bitrates from 6 kb/s to 510 kb/s (Narrowband to Wideband). Opus is unmatched for interactive speech and music transmission over the Internet but is also intended for storage and streaming applications.	Useful
codec_g722	Wideband audio codec ITU standard codec that provides 7 kHz wideband audio at data rates from 48, 56 and 64 kb/s.	Useful (High Definition)
codec_gsm	Global System for Mobile Communications (GSM) codec. Compress 3.1 kHz audio into between 6.5 and 13 kb/s	Usable
codec_resample	For SLIN Codec. Resamples between 8-bit and 16-bit signed linear.	Usable
codec_h264	provides support for the H.264 video codec, which is commonly used in video conferencing applications.	Useful

Format Interpreter modules

Format modules are responsible for converting audio and video signals from one format to another. Asterisk supports a wide range of audio and video formats, including WAV, MP3, and AVI. Format modules are used to ensure that audio and video signals are compatible with the receiving device. Some examples of format modules include:

Name	Plays files stored in	Distributed with Asterisks	Popularity
format_g722	G.722 .g722	Yes	Useful
format_wav	.wav	Yes	Useful
format_g729	Support G.729 .g729 which is commonly used for VoIP calls	Codec (NO) / Format (YES)	Useful
format_sln	8-bit signed linear .sln .raw	Yes	Useful
format_sln16	16-bit signed linear .sln16	Yes	Useful
format_pcm	Various Pulse-Coded Modulation formats:  .alaw,  .al,  .alw, .pcm,  .ulaw,  .ul,  .mu, .ulw,  .g722,  .au	Yes	Useful
format_siren14	G.722.1 Annex C (14 kHz) .siren14	Yes	New
format_siren7	G.722.1 (7 kHz) .siren7	Yes	New
format_h264	H.264—video .h264	Yes	Usable
format_h263	H.263—video .h263	Yes	Usable
format_ogg_vorbis	Ogg container .ogg	Codec (NO) / Format (YES)	Usable
format_gsm	RPE-LTP (original GSM codec) .gsm	Yes	Usable
format_wav_gsm	GSM audio in a WAV container .WAV, .wav49	Yes	Usable

Application modules

Application modules provide the core functionality for Asterisk applications. These modules allow developers to create custom applications that can interact with the Asterisk server. Some examples of application modules includes:

Name	Purpose	Popularity/Status
app_adsiprog	Loads Analog Display Services Interface (ADSI) scripts into compatible analog phones	Insignificant
app_alarmreceiver	Supports receipt of reports from alarm equipment	Insignificant
app_amd	Detects answering machines	Unreliable
app_authenticate	Compares dual-tone multi-frequency (DTMF) input against a provided string (password)	Useful
app_cdr	Writes ad hoc record to CDR	Useful
app_celgenuserevent	Generates user-defined events for CEL	New
app_chanisavail	Checks the status of a channel	Unreliable
app_channelredirect	Forces another channel into a different part of the dialplan	Useful
app_chanspy	Loads Analog Display Services Interface (ADSI) scripts into compatible analogue phones	Useful
app_confbridge	Provides conferencing (new version)	New— not fully featured yet
app_controlplayback	Plays back a prompt and offers fast forward and rewind functions	Useful
app_dahdibarge	Allows barging in on a DAHDI channel. Deprecated—see app_chanspy	Deprecated
app_dahdiras	Creates a RAS server over a DAHDI channel (no modem emulation)	Insignificant
app_db	Used to add/change/delete records in Asterisk’s built-in Berkeley database Deprecated—see  func_db	Deprecated
app_dial	Used to connect channels together (i.e., allow users to make phone calls)	Essential
app_dictate	Plays back a recording and offers start/stop functions	Useful
app_directed_pickup	Answers a call for another extension	Useful
app_directory	Presents the list of names from  voicemail.conf	Useful
app_disa	Provides dialtone and accepts DTMF input	Useful
app_dumpchan	Dumps channel variables to Asterisk command-line interface (CLI)	Useful
app_echo	Loops received audio back to source channel	Useful
app_exec	Contains Exec(),  TryExec(), and ExecIf(); executes a dialplan application based on conditions	Useful
app_externalivr	Controls Asterisk as with an AGI, only asynchronously	Useful
app_fax	Provides SendFax()  and ReceiveFax()	Useful
app_festival	Enables basic text to speech using Festival TTS engine	Usable
app_flash	Performs a hook-switch flash on channels (primarily analog)	Useful
app_followme	Performs find me/follow me functionality based on followme.conf	Useful
app_forkcdr	Starts new CDR record on current call	Usable
app_getcpeid	Gets the ADSI CPE ID	Insignificant
app_ices	Sends audio to an Icecast server	Usable
app_image	Transmits an image to supported devices	Limited
app_ivrdemo	Sample application for developers	Insignificant
app_jack	Works with JACK Audio Connection Kit to share audio between compatible applications	Useful
app_macro	Triggers dialplan macros. Useful up to Asterisk 16. Deprecated—see  GoSub() for newer installs	Deprecated
app_meetme	Provides multiparty conferencing	Useful— fully featured
app_milliwatt	Generates 1004-Hz tone for testing loss on analog circuits	Useful
app_minivm	Provides primitive functions to allow you to build your own voicemail application in dialplan	Usable
app_mixmonitor	Records both sides of a call and mixes them together	Useful
app_morsecode	Generates Morse code	Usable
app_mp3	Uses mpg123 to play an MP3	Insignificant
app_nbscat	Streams audio from Network Broadcast Stream (NBS)	Insignificant
app_originate	Allows origination of a call	Useful
app_osplookup	Performs Open Settlement Protocol (OSP) lookup	Usable
app_page	Creates multiple audio connections to specified devices for public address (paging)	Useful
app_parkandannounce	Enables automated announcing of parked calls	Usable
app_playback	Plays a file to the channel (does not accept input). plays back audio files or recorded messages.	Useful
app_playtones	Plays pairs of tones of specified frequencies	Useful
app_privacy	Requests input of caller’s phone number if no CallerID is received	Insignificant
app_queue	Provides Automatic Call Distribution (ACD)	Useful
app_read	Requests input of digits from callers and assigns input to a variable	Useful
app_readexten	Requests input of digits from callers and passes call to a designated extension and context	Usable
app_readfile	Loads contents of a text file into a channel variable Deprecated—see the FILE() function in func_env	Deprecated
app_record	Records received audio to a file, i.e. records audio messages or phone calls.	Useful
app_rpt	Provides a method to interface with an audio board for the app_rpt project	Limited
app_sayunixtime	Plays back time in specified format	Useful
app_senddtmf	Transmits DTMF to calling party	Useful
app_sendtext	Sends a text string to compatible channels	Insignificant
app_setcallerid	Sets CallerID on a channel Deprecated—see  func_callerid	Deprecated
app_skel	Sample application for developers	Useful
app_sms	Sends SMS message in compatible countries	Limited
app_softhangup	Requests hangup of channel	Useful
app_speech_utils	Provides utilities relating to speech recognition	Useful
app_stack	Provides Gosub(),  GoSubIf(),  Return(),  StackPop(),  LOCAL(), and  LOCAL_PEEK()	Essential
app_system	Executes commands in a Linux shell	Useful
app_talkdetect	Similar to app_background, but allows for any received audio to interrupt playback	Useful
app_test	Client/server testing application	Usable
app_transfer	Performs a transfer on the current channel	Useful
app_url	Passes a URI to the called channel	Limited
app_userevent	Generates a custom event in the Asterisk Manager Interface (AMI)	Useful
app_verbose	Generates a custom event in the Asterisk CLI	Useful
app_voicemail	Provides voicemail functionality for users.	Essential
app_waitforring	Waits for a RING signalling event (not to be confused with RINGING); most likely unnecessary, as only chan_dahdi with analogue channels where ringing is received (such as an FXO port) generates the RING signalling event	Insignificant
app_waitforsilence	Includes WaitForSilence()  and  WaitForNoise(); listens to the incoming channel for a specified number of milliseconds of noise/silence	Useful
app_waituntil	Waits until the current Linux epoch matches the specified epoch	Useful
app_while	Includes While(), EndWhile(), ExitWhile(), and  ContinueWhile(); provides while-loop functionality in the dialplan	Useful
app_zapateller	Generates SIT tone to discourage telemarketers	Usable

Bridging Modules

Bridging modules perform the actual bridging of channels in the bridging API. When a bridge is created, it acts as a central hub, handling the media streams between multiple participating channels and allowing for seamless communication between endpoints. The Bridge module supports various bridge technologies, including mixing, forwarding, and handling different media types (e.g., voice, video). It enables the establishment of conference calls, calls transfers, call forwarding, and other advanced call routing functionalities.

Each bridge module provides different features, which get used in different situations depending on what a bridge needs. The modules listed in the Table below, “Bridging modules”, is currently only used for (and are essential to) app_confbridge.

Note: The ConfBridge module is specifically designed to facilitate conference calls within Asterisk. It allows multiple participants to join a conference bridge and engage in real-time audio and optionally video communication. ConfBridge supports features like participant muting, moderation controls, dynamic conference creation, and customizable conference layouts. It provides a robust and scalable solution for hosting audio and video conferences.

Name	Purpose	Popularity/Status
bridge_builtin_features	Performs bridging when utilizing built-in user features (such as those found in features.conf).	New
bridge_multiplexed	Performs complex multiplexing, as would be required in a large conference room (multiple participants). Currently only used by app_confbridge.	New
bridge_simple	Performs simple channel-to-channel bridging.	New
bridge_softmix	Performs simple multiplexing, as would be required in a large conference room (multiple participants). Currently only used by app_confbridge.	New

Some other important Asterisk Bridge modules

Module Name	Filename	Description
Bridge	bridge.so	The core bridging module responsible for creating bridges and managing media streams between channels.
ConfBridge	res_conf_bridge.so	Enables conference call functionality, allowing multiple participants to join a conference bridge.
ChanSpy	chan_spy.so	Provides call monitoring and eavesdropping capabilities, allowing authorized users to listen in on active calls.
Transfer	app_transfer.so	Enables call transfer functionality, allowing users to transfer ongoing calls to different destinations.
Local Bridging	res_local.so	Supports local bridging of channels within the same Asterisk server for internal calls.
Remote Bridging	res_rtp_asterisk.so	Enables bridging of channels across multiple Asterisk servers, facilitating communication between instances.

DialPlan Function modules

Function modules provide additional functionality for Asterisk applications. These modules allow developers to create custom functions that can be used in dialplan scripts. Some examples of function modules include:

Name	Purpose	Popularity/Status
func_aes	Encrypts/decrypts an AES string	Useful
func_audiohookinherit	Converts supplied string to a SHA1 hash	Useful
func_base64	Encodes/decodes a base-64 string	Usable
func_blacklist	Writes/reads blacklist in astdb	Useful
func_callcompletion	Gets/sets call completion configuration parameters for the channel	New
func_callerid	Gets/sets CallerID. allows users to modify or manipulate the caller ID information that is displayed on the receiving device.	Useful
func_cdr	Gets/sets CDR variable	Useful
func_channel	Gets/sets channel information	Useful
func_config	Includes AST_CONFIG(); reads variables from config file	Usable
func_connectedline	Changes connected line information on supported handsets	New
func_curl	Uses cURL to obtain data from a URI	Useful
func_cut	Slices and dices strings	Useful
func_db	Provides astdb functions. allows users to store and retrieve data from	Useful
func_devstate	Includes AST_CONFIG(); reads variables from the config file	Useful
func_dialgroup	Creates a group for simultaneous dialing	Useful
func_dialplan	Validates that designated target exists in dialplan	Useful
func_enum	Performs ENUM lookup	Useful
func_env	Gets the state of device	Useful
func_extstate	Returns status of a hinted extension	Useful
func_global	Gets/sets global variables	Useful
func_groupcount	Gets/sets channel count for members of a group	Useful
func_iconv	Converts between character sets	Usable
func_lock	Includes LOCK(), UNLOCK(), and TRYLOCK(); sets a lock that can be used to avoid race conditions in the dialplan	Useful
func_logic	Includes ISNULL(),  SET(),  EXISTS(),  IF(),  IFTIME(), and IMPORT(); performs various logical functions	Useful
func_math	Includes MATH(), INC(), and DEC(); performs mathematical functions	Useful
func_md5	Converts supplied string to an MD5 hash	Useful
func_module	Checks to see if supplied module is loaded into memory	Usable
func_odbc	Allows dialplan integration with ODBC resources	Useful
func_pitchshift	Shifts the pitch of an audio stream	Useful
func_rand	Returns a random number within a given range	Useful
func_realtime	Performs lookups within the Asterisk Realtime Architecture (ARA)	Useful
func_redirecting	Provides access to information about where this call was redirected from	Useful
func_sha1	Performs string format functions similar to C function of the same name	Useful
func_shell	Performs Linux shell operations and returns results	Useful
func_speex	Reduces noise and performs dB gain/loss on an audio stream	Useful
func_sprintf	Performs string format functions similar to C function of same name	Useful
func_srv	Perform SRV lookups in the dialplan	Useful
func_strings	Includes over a dozen string manipulation functions	Useful
func_sysinfo	Gets system information such as RAM, swap, load average, etc.	Useful
func_timeout	Gets/sets timeouts on channel	Useful
func_uri	Converts strings to URI-safe encoding	Useful
func_version	Returns Asterisk version information	Usable
func_vmcount	Returns count of messages in a voicemail folder for a particular user	Useful
func_volume	Sets volume on a channel	Useful

Resource Modules

Resource modules integrate Asterisk with external resources. For example, res_odbc allows Asterisk to interoperate with ODBC database connections. Below is a table for the currently available resource modules in asterisk

Name	Purpose	Popularity/Status
res_adsi	Provides ADSI	Essential
res_ael_share	Provides shared routines for use with pbx_ael	Essential if you’re using AEL
res_agi	Provides Asterisk Gateway Interface	Useful
res_ais	Provides distributed message waiting for the indication (MWI) and device state notifications via an implementation of the AIS standard, such as OpenAIS	Useful
res_calendar	Enables base integration to calendaring systems	Useful
res_calendar_caldav	Provides CalDAV-specific capabilities	Useful
res_calendar_exchange	Provides MS Exchange capabilities	Useful
res_calendar_icalendar	Provides Apple/Google iCalendar capabilities	Useful
res_clialiases	Creates CLI aliases	Useful
res_clioriginate	Originates a call from the CLI	Usable
res_config_curl	Pulls configuration information using cURL	Useful
res_config_ldap	Pulls configuration information using LDAP	Usable
res_config_odbc	Pulls configuration information using ODBC	Useful
res_config_pgsql	Pulls configuration information using PostgreSQL	Usable
res_config_sqlite	Pulls configuration information using SQLite	Usable
res_convert	Uses the CLI to perform file conversions	Usable
res_crypto	Provides cryptographic capabilities	Useful
res_curl	Provides common subroutines for other cURL modules	Useful
res_fax	Provides common subroutines for other fax modules	Useful
res_fax_spandsp	Plug-in for fax using the spandsp package	Useful
res_http_post	Provides POST upload support for the Asterisk HTTP server	Usable
res_jabber	Provides Jabber/XMPP resources	Useful
res_limit	Enables adjusting of system limits on the Asterisk process	Usable
res_monitor	Provides call recording resources	Useful
res_musiconhold	Provides music on hold (MOH) resources	Essential
res_mutestream	Allows muting/unmuting of audio streams	New
res_odbc	Provides common subroutines for other ODBC modules	Useful
res_phoneprov	Provisions phones from Asterisk HTTP server	New
res_pktccops	Provides PacketCable COPS resources	New
res_realtime	Provides CLI commands for the Asterisk Realtime Architecture (ARA)	Useful
res_rtp_asterisk	Provides RTP	Essential
res_rtp_multicast	Provides multicast-RTP	New
res_security_log	Enables security logging	New
res_smdi	Provides voicemail notification using the SMDI protocol	Limited
res_snmp	Provides system status information to an SNMP-managed network	Usable
res_speech	Generic speech recognition API	Limited
res_timing_dahdi	Provides timing using the DAHDI kernel interface	Useful
res_timing_kqueue	Provides timing using a kernel feature in some BSDs, including Mac OS X	New
res_timing_pthread	Provides timing using only parts of the standard pthread API; is less efficient but more portable than other timing modules.	Useful
res_timing_timerfd	Provides timing using the timerfd API provided by newer versions of the Linux kernel	Useful

PBX Modules

The PBX modules are peripheral modules that provide enhanced control and configuration mechanisms. For example, pbx_config is the module that loads the traditional Asterisk dialplan. The currently available PBX modules are listed below:

Name	Purpose	Popularity/Status
pbx_ael	Asterisk Extension Logic (AEL) offers a dialplan scripting language that looks like a modern programming language.	Usable
pbx_config	This is the traditional, and most popular, dialplan language for Asterisk. Without this module, Asterisk cannot read extensions.conf.	Useful
pbx_dundi	Performs data lookups on remote Asterisk systems.	Useful
pbx_loopback	Performs something similar to a dialplan include, but in a deprecated manner.	Insignificant
pbx_lua	Allows creation of a dialplan using the Lua scripting language.	Useful
pbx_realtime	Provides functionality related to the Asterisk Realtime Architecture.	Useful
pbx_spool	Provides outgoing spool support relating to Asterisk call files.	Useful

Addon Modules

Addon modules are community-developed modules with different usage or distribution rights from those of the main code. They are kept in a separate directory and are not compiled and installed by default. To enable these modules, use the menuselect build configuration utility. Currently available addon modules are listed in the Table Below:

Table 2.11. Addon modules

Name	Purpose	Popularity/Status
app_mysql	Executes MySQL queries with a dialplan application	Deprecated—see func_odbc
app_saycountpl	Says Polish counting words	Deprecated—now integrated in  say.conf
cdr_mysql	Logs call detail records to a MySQL database	Usable—we recommend  cdr_adaptive_odbc instead
chan_mobile	Enables making and receiving phone calls using cell phones over Bluetooth	Limited
chan_ooh323	Enables making and receiving VoIP calls using the H.323 protocol	Usable
format_mp3	Allows Asterisk to play MP3 files	Usable
res_config_mysql	Uses a MySQL database as a real-time configuration backend	Useful

Asterisks File Structure

Asterisk file structure may differ based on the Operating system of choice. After installing Asterisk on Ubuntu, the file structure follows a standard layout. Here is an overview of the important directories and files in the Asterisk installation on Ubuntu:

1. Configuration Files
   • /etc/asterisk/: This directory contains the main configuration files for Asterisk, such as asterisk.conf, extensions.conf, pjsip.conf, and voicemail.conf. It also includes subdirectories for additional configuration files related to specific modules or functionalities.
2. Binaries and Scripts:
   • /usr/sbin/asterisk: The main Asterisk binary, responsible for running the Asterisk service and handling call processing.
   • /usr/sbin/asterisk_start: A script used to start the Asterisk service.
   • /usr/sbin/asterisk_stop: A script used to stop the Asterisk service.
   • /usr/sbin/safe_asterisk: A script that provides a safe wrapper for starting Asterisk and automatically restarts it if it crashes.
3. Libraries
   • /usr/lib/asterisk/: This directory contains shared libraries used by Asterisk.
4. Module Configuration
   • /etc/asterisk/modules.conf: This file specifies which modules should be loaded by Asterisk during startup. It defines the list of modules and their configuration options.
     
     Asterisk modules are usually installed to the /usr/lib/asterisk/modules folder. You will not normally have to interact with this folder; however, it will be occasionally helpful to know where the modules are located. For example, if you upgrade Asterisk and select different modules during the menu select phase of the installation, the old (incompatible) modules from the previous Asterisk version will not be deleted, and you will get a warning from the install script. Those old files will need to be deleted from the modules folder. This can be done manually or with the “uninstall” make (make uninstall) target.
5. Log Files
   • /var/log/asterisk/: This directory contains log files generated by Asterisk. It includes files such as messages, full, debug, and cdr-csv.
6. Temporary and Runtime Files:
   • /var/run/asterisk/: This directory stores temporary and runtime files used by Asterisk, including the PID (Process ID) file.
7. Sounds and Prompts:
   • /var/lib/asterisk/sounds/: This directory stores various sound files used by Asterisk for prompts, IVR menus, and voicemail recordings. It typically contains subdirectories for different languages and categories of sounds.
8. Voicemail and Recording:
   The spool is where files that change frequently are stored in Linux. For Asterisk, the spool stores transient items such as voice messages, call recordings, call files, etcetera.
   • /var/spool/asterisk/voicemail/: This directory is used to store voicemail messages recorded by Asterisk users.
   • /var/spool/asterisk/monitor/: This directory is used to store call recordings made by Asterisk.
9. Sample Configurations:
   • /usr/share/doc/asterisk/: This directory contains documentation and sample configuration files for Asterisk.
10. Web Interfaces
    • /var/www/html/: Web interfaces for Asterisk typically reside in the /var/www/html/ directory on Ubuntu systems. This is the default web root directory where web-based applications, including Asterisk web interfaces, are placed.
      When installing a web interface for Asterisk, it often involves copying the necessary files and directories to the /var/www/html/ directory. This allows the webserver to serve the interface to users accessing the appropriate URL.

Please note that the exact file structure may vary depending on the specific version of Asterisk and the installation options chosen. Always refer to the actual file system on your Ubuntu system to ensure accuracy.