Modules

Modules
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Asterisk is built on modules. A module is a loadable component that provides a specific functionality, such as a channel driver (for example, chan_sip.so), or a resource that allows connection to an external technology (such as func_odbc.so). Asterisk modules are loaded based on the /etc/asterisk/modules.conf file. We will discuss the use of many modules in this book. At this point we just want to introduce the concept of modules, and give you a feel for the types of modules that are available.

It is actually possible to start Asterisk without any modules at all, although in this state it will not be capable of doing anything. It is useful to understand the modular nature of Asterisk in order to appreciate the architecture.

Note

You can start Asterisk with no modules loaded by default and load each desired module manually from the console, but this is not something that you’d want to put into production; it would only be useful if you were performance-tuning a system where you wanted to eliminate everything not required by your specific application of Asterisk.

The types of modules in Asterisk include the following:

Applications
Bridging modules
Call detail recording (CDR) modules
Channel event logging (CEL) modules
Channel drivers
Codec translators
Format interpreters
Dialplan functions
PBX modules
Resource modules
Addons modules
Test modules

In the following sections we will list each module available within these categories, briefly identify its purpose, and give our opinion on its relative popularity and/or importance (while some modules are proven and deservedly popular, others are quite old, are barely ever used anymore, and are only maintained for the purpose of backward-compatibility). The details of how specific modules work will be covered in various chapters throughout the book, depending on what the module is and what it does. Some modules will be covered thoroughly; others may not be covered at all.

Regarding the Popularity/Status column in the tables that follow, the following list contains our opinions with respect to the meanings we have chosen (your mileage may vary):

Insignificant: This module is ancient history. If you use it, be aware that you are mostly on your own when it comes to any sort of community support.
Unreliable: This module is new or experimental, and is not suitable for production.
Useful: This module is current, maintained, popular, and recommended.
Usable: This module works but may be incomplete or unpopular, and/or is not recommended by the authors.
New: This module is quite new, and its completeness and popularity are difficult to gauge at this time.
Deprecated: This module has been replaced by something that is considered superior.
Limited: This module has limitations that may make it unsuitable to your requirements.
Essential: This module is one you’ll never want to be without.

And now, without further ado, let’s take a look at the modules, grouped by module type.

Applications

Dialplan applications are used in extensions.conf to define the various actions that can be applied to a call. The Dial() application, for example, is responsible for making outgoing connections to external resources and is arguably the most important dialplan application. The available applications are listed in Table 2.1, “Dialplan applications”.

Table 2.1. Dialplan applications

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`	Allows a channel to listen to audio on another channel	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`
`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`
`app_dial`	Used to connect channels together (i.e., 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^[a]
`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^[b]
`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	Deprecated—see `GoSub()`
`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)	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`
`app_record`	Records received audio to a file	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`
`app_skel`	Sample application for developers	Useful^[c]
`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^[d]
`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	Essential
`app_waitforring`	Waits for a RING signaling event (not to be confused with RINGING); most likely unnecessary, as only `chan_dahdi` with analog channels where ringing is received (such as an FXO port) generates the RING signaling 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 current Linux epoch matches 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
^[a]The use of (DISA) is considered to be a security risk. ^[b]Requires a suitable DSP engine to handle encoding/decoding of fax signaling (see Chapter 19, Fax). ^[c]If you are a developer. ^[d]Requires an external speech recognition application.

Bridging Modules

Bridging modules are new in Asterisk 1.8: they perform the actual bridging of channels in the new bridging API. Each provides different features, which get used in different situations depending on what a bridge needs. These modules, listed in Table 2.2, “Bridging modules”, are currently only used for (and are essential to) app_confbridge.

Table 2.2. Bridging modules

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

Call Detail Recording Modules

The CDR modules, listed in Table 2.3, “Call detail recording modules”, are designed to facilitate as many methods of storing call detail records as possible. You can store CDRs to a file (default), a database, RADIUS, or syslog.

Note

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, discussed next. The advantage of CDR is that it just works.

Table 2.3. Call detail recording modules

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`

We will discuss some reporting packages that you may wish to use with CDR in Chapter 25, Web Interfaces.

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 2.4, “Channel event logging modules”.

Table 2.4. 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`

Channel Drivers

Without channel drivers, Asterisk would have no way to make calls. Each channel driver is specific to the protocol or channel type it supports (SIP, ISDN, etc.). The channel module acts as a gateway to the Asterisk core. Asterisk’s channel drivers are listed in Table 2.5, “Channel drivers”.

Table 2.5. Channel drivers

Name	Purpose	Popularity/Status
`chan_agent`	Provides agent channel for `Queue()`	Useful
`\|chan_alsa`	Provides connection to Advanced Linux Sound Architecture	Useful
`chan_bridge`	Used internally by the `ConfBridge()` application; should not be used directly	Essential ^[a]
`chan_console`	Provides connection to portaudio	New
`chan_dahdi`	Provides connection to PSTN cards that use DAHDI channel drivers	Useful
`chan_gtalk`	Provides connection to Google Talk	Usable
`chan_h323`	Provides connection to H.323 endpoints	Deprecated—see `chan_ooh323` in Table 2.11, “Addon modules”
`chan_iax2`	Provides connection to IAX2 endpoints	Useful
`chan_jingle`	Provides connection to Jingle-enabled endpoints	Usable
`chan_local`	Provides a mechanism to treat a portion of the dialplan as a channel	Useful
`chan_mgcp`	Media Gateway Control Protocol channel driver	Usable
`chan_misdn`	Provides connection to mISDN supported ISDN cards	Limited
`chan_multicast_rtp`	Provides connection to multicast RTP streams	Useful
`chan_nbs`	Network Broadcast Sound channel driver	Insignificant
`chan_oss`	Open Sound System driver	Useful
`chan_phone`	Linux telephony interface driver (very old)	Insignificant
`chan_sip`	Session Initiation Protocol channel driver	Essential
`chan_skinny`	Cisco Skinny Client Control Protocol (SCCP) channel driver	Usable
`chan_unistim`	Nortel Unistim protocol channel driver	Usable
`chan_usbradio`	Channel driver for CM108 USB cards with radio interface	Usable
`chan_vpb`	Voicetronix channel driver	Insignificant^[b]
^[a]If you are using the `ConfBridge()` application. ^[b]Some Voicetronix hardware is supported by Zaptel using an addon Zaptel module distributed by Voicetronix. However, Zaptel is no longer supported by Asterisk and this driver has not been ported to DAHDI.

Codec Translators

The codec translators (Table 2.6, “Codec translators”) allow Asterisk to convert audio stream formats between calls. So if a call comes in on a PRI circuit (using G.711) and needs to be passed out a compressed SIP channel (e.g., using G.729, one of many codecs that SIP can handle), the relevant codec translator would perform the conversion.^[13]

Note

If a codec (such as G.729) uses a complex encoding algorithm, heavy use of transcoding can place a massive burden on the CPU. Specialized hardware for the decoding/encoding of G.729 is available from hardware manufacturers such as Sangoma and Digium (and likely others).

Table 2.6. Codec translators

Name	Purpose	Popularity/Status
`codec_adpcm`	Adaptive Differential Pulse Coded Modulation codec	Insignificant
`codec_alaw`	A-law PCM codec used all over the world (except Canada/USA) on the PSTN	Essential
`codec_a_mu`	A-law to mu-law direct converter	Useful
`codec_dahdi`	Utilizes proprietary Digium hardware transcoding card	Essential^[a]
`codec_g722`	Wideband audio codec	Useful
`codec_g726`	Flavor of ADPCM	Insignificant
`codec_gsm`	Global System for Mobile Communications (GSM) codec	Useful
`codec_ilbc`	Internet Low Bitrate Codec	Insignificant
`codec_lpc10`	Linear Predictive Coding vocoder (extremely low bandwidth)	Insignificant
`codec_resample`	Resamples between 8-bit and 16-bit signed linear	Usable
`codec_speex`	Speex codec	Usable
`codec_ulaw`	Mu-law PCM codec used in Canada/USA on PSTN	Essential
^[a]If you are using a Digium codec transcoder card.

Format Interpreters

Format interpreters (Table 2.7, “Format interpreters”) perform the function of codec translators, but they do their work on files rather than channels. If you have a recording on a menu that has been stored as GSM, a format interpreter would need to be used to play that recording to any channels not using the GSM codec.^[14]

If you store a recording in several formats (such as WAV, GSM, etc.), Asterisk will determine the least costly format^[15] to use when a channel requires that recording.

Table 2.7. Format interpreters

Name	Plays files stored in	Popularity/Status
`format_g723`	G.723 `.g723`	Insignificant
`format_g726`	G.726 `.g726`	Insignificant
`format_g729`	G.729 `.g729`	Useful
`format_gsm`	RPE-LTP (original GSM codec) `.gsm`	Usable
`format_h263`	H.263—video `.h263`	Usable
`format_h264`	H.264—video `.h264`	Usable
`format_ilbc`	Internet Low Bitrate Codec `.ilbc`	Insignificant
`format_jpeg`	Graphic file `.jpeg` `.jpg`	Insignificant
`format` `_ogg_` `vorbis`	Ogg container `.ogg`	Usable
`format_pcm`	Various Pulse-Coded Modulation formats: `.alaw`, `.al`, `.alw`, `.pcm`, `.ulaw`, `.ul`, `.mu`, `.ulw`, `.g722`, `.au`	Useful
`format_siren14`	G.722.1 Annex C (14 kHz) `.siren14`	New
`format_siren7`	G.722.1 (7 kHz) `.siren7`	New
`format_sln16`	16-bit signed linear `.sln16`	New
`format_sln`	8-bit signed linear `.sln` `.raw`	Useful
`format_vox`	`.vox`	Insignificant
`format_wav`	`.wav`	Useful
`format_wav_gsm`	GSM audio in a WAV container `.WAV`, `.wav49`	Usable

Dialplan Functions

Dialplan functions, listed in Table 2.8, “Dialplan functions”, complement the dialplan applications (see the section called “Applications”). They provide many useful enhancements to things like string handling, time and date wrangling, and ODBC connectivity.

Table 2.8. Dialplan functions

Name	Purpose	Popularity/Status
`func_aes`	Encrypts/decrypts an AES string	Useful
`func_audiohookinherit`	Allows calls to be recorded after transfer	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	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	Useful
`func_devstate`	Gets state of device	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`	Includes `FILE()`, `STAT()`, and `ENV()`; performs operating system actions	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`	Converts supplied string to an SHA1 hash	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

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 in Table 2.9, “PBX modules”.

Table 2.9. PBX modules

Name	Purpose	Popularity/Status
`pbx_ael`	Asterisk Extension Logic (AEL) offers a dialplan scripting language that looks like a modern programming language.	Usable^[a]
`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^[b]
`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
^[a]We have not found too many people using AEL. We suspect this is because most developers will tend to use AGI/AMI if they do not want to use traditional dialplans. ^[b]We’ve never heard of this being used in production.

Resource Modules

Resource modules integrate Asterisk with external resources. For example, res_odbc allows Asterisk to interoperate with ODBC database connections. The currently available resource modules are listed in Table 2.10, “Resource modules”.

Table 2.10. Resource modules

Name	Purpose	Popularity/Status
`res_adsi`	Provides ADSI	Essential^[a]
`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 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^[b]
`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; 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
^[a]While most of the ADSI functionality in Asterisk is never used, the voicemail application uses this resource. ^[b]Requires a separately licensed product in order to be used.

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 Table 2.11, “Addon modules”.

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^[a]
`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
^[a]While `chan_mobile` works great with many phones, problems have been reported with some models. When a problem does occur, it is very difficult for developers to solve unless they have a phone of the same model to test with.

Test Modules

Test modules are used by the Asterisk development team to validate new code. They are constantly changing and being added to, and are not useful unless you are developing Asterisk software.

If you are an Asterisk developer, however, the Asterisk Test Suite may be of interest to you as you can build automated tests for Asterisk and submit those back to the project, which runs on several different operating systems and types of machines. By expanding the number of tests constantly, the Asterisk project avoids the creation of regressions in code. By submitting your own tests to the project, you can feel more confident in future upgrades.

More information about installing the Asterisk Test Suite is available in this blog post: http://blogs.asterisk.org/2010/04/29/installing-the-asterisk-test-suite/. More information about building tests is available in this document: http://svn.asterisk.org/svn/testsuite/asterisk/trunk/README.txt or you can join the #asterisk-testing channel on the Freenode IRC network.

^[13]More information about what codecs are and how they work is available in the section called “Codecs”.

^[14]It is partly for this reason that we do not recommend the default GSM format for system recordings. WAV recordings will sound better and use less CPU.

^[15]Some codecs can impose a significant load on the CPU, such that a system that could support several hundred channels without transcoding might only be able to handle a few dozen when transcoding is in use.