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Asterisk: The Future of Telephony, 2nd Edition by Leif Madsen, Jared Smith, Jim Van Meggelen

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Appendix D. Configuration Files


This appendix contains a reference to the configuration files not covered in the previous appendices. If you are looking for VoIP channel configurations, refer to Appendix A. For a dialplan reference, you’ll want to use Appendix B.

A configuration file is required for each Asterisk module you wish to use. These .conf files contain channel definitions, describe internal services, define the locations of other modules, or relate to the dialplan. You do not need to configure all of them to have a functioning system, only the ones required for your configuration. Although Asterisk ships with samples of all of the configuration files, it is possible to start Asterisk without any of them. This will not provide you with a working system, but it clearly demonstrates the modularity of the platform.

If no .conf files are found, Asterisk will make some decisions with respect to modules. For example, the following steps are always taken:

  • The Asterisk Event Logger is loaded, and events are logged to /var/log/asterisk/event_log.

  • Manager actions are registered.

  • The PBX core is initialized.

  • The RTP port range is allocated from 5,000 through 31,000.

  • Several built-in applications are loaded, such as Answer(), Background(), GotoIf(), NoOp(), and Set().

  • The dynamic loader is started; this is the engine responsible for loading modules defined in modules.conf.

This appendix starts with an in-depth look at the modules.conf configuration file. We’ll then briefly examine all the other files that you may need to configure for your Asterisk system.


The modules.conf file controls which modules are loaded or not loaded at Asterisk startup. This is done through the use of the load => or noload => constructs.


This file is a key component to building a secure Asterisk installation: best practice suggests that only required modules be loaded.

The modules.conf file always starts with the [modules] header. The autoload statement tells Asterisk whether to automatically load all modules contained within the modules directory or to load only those modules specifically defined by load => statements. We recommend you manually load only those modules you need, but many people find it easier to let Asterisk attempt to autoload whatever it finds in /usr/lib/asterisk/modules. You can then exclude[164] certain modules with noload => statements.

Here’s a sample modules.conf file:

    autoload=no                ; set this to yes and Asterisk will load any
                               ; modules it finds in /usr/lib/asterisk/modules

    load => res_adsi.so
    load => pbx_config.so      ; Requires: N/A
    load => chan_iax2.so       ; Requires: res_crypto.so, res_features.so
    load => chan_sip.so        ; Requires: res_features.so
    load => codec_alaw.so      ; Requires: N/A
    load => codec_gsm.so       ; Requires: N/A
    load => codec_ulaw.so      ; Requires: N/A
    load => format_gsm.so      ; Requires: N/A
    load => app_dial.so        ; Requires: res_features.so, res_musiconhold.so

Since we assume Asterisk is built on Linux, all the module names we use end in a .so extension. However, this may not be the case if you have built Asterisk on a different operating system.

As of this writing, there are eight module types: resources, applications, Call Detail Record database connectors, channels, codecs, formats, PBX modules, and standalone functions.


The Analog Display Services Interface (ADSI) was designed to allow telephone companies to deliver enhanced services across analog telephone circuits. In Asterisk, you can use this file to send ADSI commands to compatible telephones. Please note that the phone must be directly connected to a Zapata channel. ADSI messages cannot be sent across a VoIP connection to a remote analog phone.

The res_adsi.so module is required for the Voicemail() application; however, the adsi.conf file is not necessarily used. Detailed information about ADSI is not publicly available, and documentation needs to be purchased from Telcordia.


Prior to Voice over IP, Voice over Frame Relay (VoFR) enjoyed brief fame as a means of carrying packetized voice. Supporting VoFR through Adtran equipment is part of the history of Asterisk.

This feature is no longer popular in the community, though, so it may be difficult to find support for it.


This file allows you to create and manage agents for your call center. If you are using the Queue() application, you may want to configure agents for the queue. The agents.conf file is used to configure the AGENT channel driver.

The [general] section in agents.conf currently contains only two parameters. The persistentagents parameter tells Asterisk whether or not to save the status of agents who use the callback feature of queues in the local Asterisk database. If set to yes, a logged-in remote agent will then remain logged in across a reboot (unless removed from the database through some other means). The multiplelogin parameter tells Asterisk whether or not multiple agents can log in from the same extension.

The following parameters, which are specified in the [agents] section, are used to define agents and the way the system interacts with them. The settings apply to all agents, unless otherwise specified in the individual agent definitions:


The maximum number of times an agent may attempt to log in. Defaults to 3.


Accepts an argument (in seconds) defining how long an agent channel should ring for before the agent is deemed unavailable and logged off.


Define autologoffunavail to have agents automatically logged out when the Dial() application returns a CHANUNAVAIL status while trying to dial that agent. Default is “no”.


Accepts the arguments yes and no. If set to yes, requires a callback agent to acknowledge log in by pressing the # key after logging in. This works in conjunction with the AgentCallbackLogin() application.


If set to yes, allows an agent to hang up a call by pressing the * key. Defaults to yes. Set this to no to have Asterisk do nothing when the agent presses the * key.


You can configure this parameter to allow agents a few seconds of downtime after completing a call before the queue presents them with another call. This setting is measured in milliseconds.

musiconhold => class

Accepts a music-on-hold class as its argument. This setting applies to all agents.


Defines the default goodbye sound for agents.


Accepts the arguments yes and no. Used to define whether the source channel in the CDRs should be set to agent/agent_id to determine which agent generated the calls.


Defines the groups to which an agent belongs, specified with integers. Specify that an agent belongs to multiple groups by separating the integers with commas.


Accepts the arguments yes and no. Defines whether or not agent calls should be recorded.


Defines the format to record files in. The argument specified should be wav, gsm, or wav49. The default recording format is wav.


Accepts a string as its argument. The string can be formed as a URL and is appended to the start of the text to be added to the name of the recording.


Accepts a filesystem path as its argument. Allows you to override the default path of /var/spool/asterisk/monitor/ with one of your choosing.


Since the storage of calls will require a large amount of hard drive space, you will want to define a strategy to handle storing and managing these recordings.

This location should probably reside on a separate volume, one with very high performance characteristics.


Accepts a filename as its argument. Can be used to define a custom notification tone to signal to an always-connected agent that there is an incoming call.

The final parameter is used to define agents. Just like in the zapata.conf file, configuration parameters are inherited from above the agent => definition. Agents are defined with the following format:

agent => agent_id,agent_password,name

For example, we can define agent Happy Tempura with the agent ID 1000 and password 1234, as follows.

agent => 1000,1234,Happy Tempura

Be aware that an agents.conf file is a complement to the queue configuration process. The most critical configuration file for your queues is queues.conf. You can configure a very basic queue without agents.conf.



The AlarmReceiver() application is not approved by Underwriter’s Laboratory (UL) and should not be used as the primary or sole means of receiving alarm messages or events. This application is not guaranteed to be reliable, so don’t depend on it unless you have extensively tested it. Use of this application without extensive testing may place your life and/or property at risk.

The alarmreceiver.conf file is used by the AlarmReceiver() application, which allows Asterisk to accept alarms using the SIA (Ademco) Contact ID protocol. When a call is received from an alarm panel, it should be directed to a context that calls the AlarmReceiver() application. In turn, AlarmReceiver() will read the alarmreceiver.conf configuration file and perform the configured actions as required. All parameters are specified under the [general] heading.

The sample configuration file will contain the current settings for this application and is very well documented.


The alsa.conf file is used to configure Asterisk to use the Advanced Linux Sound Architecture (ALSA) to provide access to a sound card, if desired. You can use this file to configure the CONSOLE channel, which is most commonly used to create an overhead paging system (although, as with any other channel, there are all kinds of creative ways this can be used). Keep in mind that the usefulness of the ALSA channel by itself is limited due to its lack of a user interface.[165]


This is the configuration for the answering matching detection application in Asterisk, called AMD(). It is used to adjust the different parameters for detecting an answering machine based on items such as initial silence, greeting length, silence after the greeting, and so forth.


The asterisk.conf file defines the locations for the configuration files, the spool directory, and the modules, as well as a location to write logfiles to. The default settings are recommended unless you understand the implications of changing them. The asterisk.conf file is generated automatically when you run the make samples command, based on information it collects about your system. It will contain a [directories] section such as the following:

astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdatadir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run
astlogdir => /var/log/asterisk

Additionally, you can specify an [options] section, which will allow you to define startup options (command-line switches) in the configuration file. The following example shows the available options and the command-line switches that they effectively enforce:

;Under "options" you can enter configuration options
;that you also can set with command line options

verbose = 0             ; Verbosity level for logging (-v)
debug = 3               ; Debug: "No" or value (1-4)
nofork=yes | no         ; Background execution disabled (-f)
alwaysfork=yes | no     ; Always background, even with -v or -d (-F)
console= yes | no       ; Console mode (-c)
highpriority = yes | no ; Execute with high priority (-p)
initcrypto = yes | no   ; Initialize crypto at startup (-i)
nocolor = yes | no      ; Disable ANSI colors (-n)
dumpcore = yes | no     ; Dump core on failure (-g)
quiet = yes | no        ; Run quietly (-q)
timestamp = yes | no    ; Force timestamping in CLI verbose output (-T)
runuser = asterisk     ; User to run asterisk as (-U) NOTE: will require changes to
                       ; directory and device permissions
rungroup = asterisk    ; Group to run asterisk as (-G)
internal_timing = yes | no   ; Enable internal timing support (-I)
                             ; These options have no command line equivalent
cache_record_files = yes | no    ; Cache record() files in another directory until 
                                 ; completion
record_cache_dir = <dir>
transcode_via_sln = yes | no       ; Build transcode paths via SLINEAR
transmit_silence_during_record = yes | no ; send SLINEAR silence while channel is 
                                          ; being recorded
maxload = 1.0             ; The maximum load average we accept calls for
maxcalls = 255            ; The maximum number of concurrent calls you want 
                          ; to allow
execincludes = yes | no   ; Allow #exec entries in configuration files
dontwarn = yes | no       ; Don't over-inform the Asterisk sysadm, he's a guru
systemname = <a_string>   ; System name. Used to prefix CDR uniqueid and to 
                          ; fill ${SYSTEMNAME}
languageprefix = yes | no ; Should language code be last component of sound file 
                          ; name or first?
                          ; When off, sound files are searched as 
                          ; <path>/<lang>/<file>
                          ; When on, sound files are search as <lang>/<path>/<file>
                          ; (only affects relative paths for sound files)


The cdr.conf file is used to enable call detail record logging to a flat file or a database. Storing call records is useful for all sorts of purposes including billing, fraud prevention, QoS evaluations, and more. The cdr.conf file contains some general parameters that are not specific to any particular database, but rather indicate how Asterisk should handle the passing of information to the database. All options are under the [general] heading of the cdr.conf file:


Accepts the arguments yes and no. Specifies whether or not to use CDR logging. If set to no, this will override any CDR module explicitly loaded. The default is yes.


Accepts the arguments yes and no. Allows Asterisk to write data to a buffer instead of writing to the database at the end of every call, to reduce load on the system.


Note that if the system dies unexpectedly when this option is set to yes, data loss may occur.


Sets the maximum number of CDRs to accumulate in the buffer before posting to the backend CDR storage systems. This setting only takes effect if the batch setting is set to yes. This setting defaults to 100 records.


Accepts an integer (in seconds) as its argument. Sets the number of seconds before Asterisk flushes the buffer and writes the CDRs to the database, regardless of the number of records in the buffer (as defined by size). The default is 300 seconds (5 minutes).


Accepts the arguments yes and no. If you are generating a massive volume of CDRs on a system that is pushing them to a remote database, setting scheduleronly to yes may be of benefit. Since the scheduler cannot start a new task until the current one is finished, slow CDR writes may adversely affect other processes needing the scheduler. This setting will instruct Asterisk to handle CDR writes in a new thread, essentially assigning a dedicated scheduler to this function. In normal operation, this would yield very little benefit.


Accepts the arguments yes and no. Setting safeshutdown to yes will prevent Asterisk from shutting down completely until the buffer is flushed and all information is written to the database. If this parameter is set to no and you shut down Asterisk with information still residing in the buffers, that information will likely be lost.


Normally, CDR records are not closed out until after all extensions are finished executing. By enabling this option, the CDR will be ended before executing the h extension so that CDR values such as end and billsec may be retrieved inside of this extension. Defaults to no.

The rest of cdr.conf contains setup for several of the backend CDR engines. See the sample cdr.conf for more information.


The cdr_manager.conf file simply contains a [general] heading and a single option, enabled, which you can use to specify whether or not the Asterisk Manager API generates CDR events. If you want CDR events to be generated, you will need the following lines in your cdr_manager.conf file:


The Manager API will then output CDR events containing the following fields:

Event: Cdr


Asterisk can store CDR data in a local or remote database via the ODBC interface. The cdr_odbc.conf file contains the information Asterisk needs to connect to the database. The cdr_odbc.so module will attempt to load the cdr_odbc.conf file, and if information is found for connecting to a database, the CDR data will be recorded there.


If you are going to use a database for storing CDR data, you will have to select one of the many that are available. Asterisk does not like having multiple CDR databases to connect to, so do not have extra cdr_engine.conf files hanging about your Asterisk configuration directory.


Asterisk can store CDR data in a PostgreSQL database via the cdr_pgsql.so module. When the module is loaded the necessary information will be read from the cdr_pgsql.conf file, and Asterisk will connect to the PostgreSQL database to write and store CDR data.


Asterisk can also store CDR data to a FreeTDS database (including MS SQL) with the use of the cdr_tds.so module. The configuration file cdr_tds.conf is read once the module is loaded. Upon a successful connection, CDR data will be written to the database.


Most codecs do not have any configurable parameters; they are what they are, and that’s all they are.

Some codecs, however, are capable of behaving in different ways. This primarily means that they can be optimized for a particular goal, such as cutting down on latency, making best use of a network, or perhaps delivering high quality audio.

The codecs.conf file is fairly new in Asterisk, and as of this writing it allows configuration of Speex parameters only. The settings are self-explanatory, as long as you are familiar with the Speex protocol (see http://www.speex.org).

codecs.conf also allows you to configure Packet Loss Concealment (PLC). You need to define a [plc] section and indicate genericplc => true. This will cause Asterisk to attempt to interpolate any packets that are missed. (Enabling this functionality will incur a small performance penalty.)


This file is used to configure whether Asterisk should perform DNS lookups on a regular basis, and how often those lookups should be performed.


The DUNDi protocol is used to dynamically look up the VoIP address of a phone number on a network, and to connect to that address. Unlike the ENUM standard, DUNDi has no central authority. The dundi.conf file contains DUNDi extensions used to control what is advertised; it also contains the peers to whom you will submit lookup requests and from whom you will accept lookup requests. The DUNDi protocol was explored in Chapter 14.


The Electronic Numbering (ENUM) system is used in conjunction with the Internet’s DNS system to map E.164 ITU standard (ordinary telephone) numbers to email addresses, web sites, VoIP addresses, and the like. An ENUM number is created in DNS by reversing the phone number, separating each digit with a period, and appending e164.arpa (the primary DNS zone). If you want Asterisk to perform ENUM lookups, configure the domain(s) in which to perform the lookups within the enum.conf file. In addition to the official e164.arpa domain, you can have Asterisk perform lookups in the publicly accessible e164.org domain.


Asterisk can write configuration data to and load configuration data from a database using the external configuration engine (also known as realtime). This enables you to map external configuration files (static mappings) to a database, allowing the information to be retrieved from the database. It also allows you to map special runtime entries that permit the dynamic creation and loading of objects, entities, peers, and so on without a reload. These mappings are assigned and configured in the extconfig.conf file, which is used by both res_odbc and realtime.


At the center of every good universe is a dialplan. The extensions.conf file is the means by which you tell Asterisk how you want calls to be handled. The dialplan contains a list of instructions that, unlike traditional telephony systems, is entirely customizable. The dialplan is so important that rather than defining it in this appendix, we have dedicated all of Chapters 5 and 6, as well as Appendix B, to this topic. Go forth, read, and enjoy!


This file is the equivalent of extensions.conf, only it’s for dialplans written in the AEL language. When Asterisk loads the dialplan, it reads the AEL dialplan from extensions.ael and merges it with the dialplan from extensions.conf.


features.conf, the file formally known as parking.conf, contains configuration information related to call parking and call transfers. Call parking configuration options include:

  • The extension to dial to park calls (parkext =>)

  • The extension range to park calls in (parkpos =>)

  • Which context to park calls in (context =>)

  • How long a call can remain parked for before ringing the extension that parked it (parkingtime =>)

  • The sound file played to the parked caller when the call is removed from parking (courtesytone =>)

  • ADSI parking announcements (asdipark=yes|no)

In addition to the call parking options, in this file you can configure the button mappings for blind transfers, attended transfers, one-touch recording, disconnections, and the pickup extension (which allows you to answer a remotely ringing extension).


The Festival text-to-speech engine allows Asterisk to read text files to the end user with a computer-generated voice. Festival is covered in Chapter 14.


The term findme/follow me is intended to give the impression that a PBX system has the intelligence to locate a user wherever they may be, such that their calls will find them and follow them. The followme.conf file is used to configure the FollowMe() dialplan application.


The func_odbc dialplan function was one of the most anticipated capabilities to have been added to Asterisk 1.4. This function provides a simple mechanism with which to connect to ODBC databases through the dialplan. The SQL queries are defined in this configuration file, and a dialplan function is automatically created.


This configuration file is used to specify the parameters for connectivity with Google Talk.


Asterisk has a very simple HTTP daemon built into it, which is used by the Asterisk GUI and AJAM. This functionality is discussed in Chapter 11.


Similar to sip.conf, the iax.conf file is where you configure options related to the IAX protocol. Your end devices and service providers are also configured here. The iax.conf file is covered in detail in Appendix A.


This file is used by Asterisk to allow the system to provision and upgrade the firmware on an IAXy device.


The indications.conf file is used to tell Asterisk how to generate the various telephone sounds common in different parts of the world; a dial tone in England sounds very different from a dial tone in Canada, but your Asterisk system will be pleased to make the sounds you want to hear. This file consists of a list of sounds a telephone system might need to produce (dial tone, busy signals, and so forth), followed by the frequencies used to generate those sounds.

By default (and without an indications.conf file), Asterisk will use the tones common in North America. You can change the default country for your system by specifying the two-letter country code in the [general] section. Supported country codes are listed in the indications.conf.sample file located in /usr/src/asterisk/configs. If you have the required information, your country can easily be added. Here’s what the configuration for North America looks like:

description = United States / North America
ringcadance = 2000,4000
dial = 350+440
busy = 480+620/500,0/500
ring = 440+480/2000,0/4000
congestion = 480+620/250,0/250
callwaiting = 440/300,0/10000
dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,0


The jabber.conf file specifies the information needed to allow Asterisk to interact with an XMPP (Jabber) server.


The logger.conf file specifies the type and verbosity of messages logged to the various logfiles in the /var/log/asterisk/ directory. It has two sections, [general] and [logfile].


Settings under the [general] section are used to customize the output of the logs (and can safely be left blank, as the defaults serve most people very well). However, if you love to customize such things, read on.

You can define exactly how you want your timestamps to look through the use of the dateformat parameter:

dateformat=%F %T

The Linux manpage for strftime(3) lists all of the ways you can do this.

If you want to append your system’s hostname to the names of the logfiles, set appendhostname=yes. This can be useful if you have a lot of systems delivering logfiles to you.

If for some reason you do not want to log events from your queues, you can set queue_log=no.

If generic events do not interest you, instruct Asterisk to omit them from the logfiles by setting event_log=no.


The [logfiles] section defines the types of information you wish to log. There are multiple ranks for the various bits of information that will be logged, and it can be desirable to separate log entries into different files. The general format for lines in the [logfiles] section is filename => levels, where filename is the name of the file to save the logged information to and levels are the types of information you wish to save.


Using console for the filename is a special exception that allows you to control the type of information sent to the Asterisk console.

A sample [logfiles] section might look like this:

console => notice,warning,error
messages => notice,warning,error

You can specify logging of the following types of information:


Enabling debugging gives far more detailed output about what is happening in the system. For example, with debugging enabled, you can see what DTMF tones the users entered while accessing their voicemail boxes. Debugging information should be logged only when you are actually debugging something, as it will create massive logfiles very rapidly.


When you connect to the Asterisk console and set a verbosity of 3 or higher, you’ll see output on the console showing what Asterisk is doing. You can save this output to a logfile by adding a line such as verbose_log => verbose to your logger.conf file. Note that a high amount of verbosity can quickly eat up hard drive space.


A notice is used to inform you of minor changes to the system, such as when a peer changes state. It is normal to see these types of messages, and the events they indicate generally have no adverse effects on the server.


A warning happens when Asterisk attempts to do something and is unsuccessful. These types of errors are usually not fatal, but they should be investigated, especially if a lot of them are seen.


Errors are often related to Out of Memory errors. They generally indicate serious problems that may lead to Asterisk to crash or freeze.


The Asterisk Manager Interface is an API that external programs can use to communicate with and control Asterisk, much as you would do from the Asterisk console.


The Manager gives programs the ability to run commands and request information from the Asterisk server. However, it is not very secure; its authentication mechanism defaults to using plain-text passwords, and all connected terminals default to receiving all events. The Asterisk Manager should be used only on a trusted local area network, or locally on the box. The permit and deny constructs allow you to restrict access to certain extensions or subnets.

Many of the available graphical interfaces to Asterisk—such as the Flash Operator Panel—use the Manager to pull data and determine the status of applications. The manager.conf file defines the way programs authenticate with the Manager.

The Manager commands (which you can list by typing show manager commands at the Asterisk console) have varying degrees of privilege. You can control the read and write permissions for these commands with the use of the read and write options in the manager.conf file.

Here’s a sample manager.conf file:

enabled = no
port = 5038
bindaddr =

secret = notvery
read = system,call,log,verbose,command,agent,user,config
write = system,call,log,verbose,command,agent,user,config

For more information on the Asterisk Manager Interface, see Chapter 10.


MeetMe is one of the more remarkable applications in Asterisk. It allows you to set up predefined audio conference rooms. This rather simple concept has proven to be extremely expensive to implement in every other PBX, but what seems like a big deal to them is simple to Asterisk. Whether by using a dedicated server, or through the use of a service, Asterisk now delivers this functionality as a standard application.

MeetMe conferences can be created either dynamically, with the d flag in the Dial() application, or statically in the meetme.conf file. The format for creating conference rooms is as follows:

conf => conference_number[,pin][,administrator_pin]

All conferences must be defined under the [rooms] section header.

conf => 4569
conf => 5060,54377017
conf => 3389,4242,1337
conf => 333,,2424


The Media Gateway Control Protocol (MGCP) has only primitive support in Asterisk. This is likely due to the fact that SIP has stolen the limelight from every other VoIP protocol (except IAX, of course). Because of this, you should attempt to use Asterisk’s MCGP channel in a production environment only if you are prepared to perform extensive testing, are willing to pay to have features and patches implemented within your time frames, and have in-house expertise with the protocol.

Having said that, we are not prepared to pronounce MGCP dead. SIP is not yet the panacea it has been touted as, and MGCP has proven itself to be very useful in carrier backbone environments. Many believe MGCP will fill a niche or void that has not yet been discovered, and we remain interested in it.


The modem.conf file is used by Asterisk to communicate with ISDN-BRI interfaces through the ISDN4Linux driver. Since ISDN4Linux lacks many core ISDN features, it is not generally used. For BRI, the most popular add-on seems to be chan_capi, available from http://www.junghanns.net.


The musiconhold.conf file is used to configure different classes of music and their locations for use in music-on-hold applications. Asterisk can play hold music in any native file formats. Asterisk can also makes use of a certain version of mpg123 to play MP3 files, but this is discouraged. You can specify arguments for a class, allowing you to use an external application to stream music either locally or over a network.


The Open Settlement Protocol (OSP) is officially documented in ETSI TS 101 321, a European Telecommunication Standards Institute (ETSI) document that came out of the work of the TIPHON working group. As far as we can tell, OSP is another attempt to apply old-style telecom thinking to disruptive technologies.


The oss.conf file is used to configure Asterisk to use the Open Sound System (OSS) driver to allow communications with the sound card via the CONSOLE channel. Note that ALSA is now the preferred interface for the CONSOLE channel.


The phone.conf file is used to configure a Quicknet PhoneJACK card. The PhoneJACK card seems to provide something like an FXS interface, in that you can plug an analog telephone into it and pass calls through Asterisk.


The privacy.conf file is used to control the maximum number of tries a user has to enter his 10-digit telephone number in the PrivacyManager() application. The PrivacyManager() application determines if a Caller ID is set for the incoming call. The user enters his 10-digit number within the number of tries configured in privacy.conf, the application sets the PRIVACYMGRSTATUS channel variable to either SUCCESS or FAILED. If the Caller ID is set, the application does nothing.


The PrivacyManager() application can also accept the arguments in the dialplan. This allows the value to remain in memory, instead of an I/O operation to the disk to read the configuration file, which you would only worry about with high usage of this application (many calls per second).


Asterisk provides basic call center functionality via its queueing system, but those who are using it in more mission-critical environments often report that their solutions required customization. You can do this customization in the queues.conf file.

The [general] section of queues.conf contains settings that will apply to all queues. If the persistentmembers parameter is set to yes, a member that is added to the system via the AddQueueMember() application or through the Asterisk Manager Interface will be stored in the AstDB, and therefore retained across a restart.

The autofill parameter allows Asterisk to be more efficient in the way it distributes calls to queue members, especially if there are multiple callers in the queue and multiple queue members available to receive the call. It is recommended you set autofill to yes.

Another general parameter of queues.conf is MonitorType. If set to MixMonitor, it will mix the inbound and outbound audio streams. If set to Monitor, it will revert to the older method of recording the inbound and outbound audio in separate files.

Next you can define one or more queues by placing its name inside of square brackets ([]). Within each queue, the following parameters are available:


This parameter allows you to configure which music-on-hold class (configured in musiconhold.conf) to use for the queue.


When a call is presented to a member of the queue, the prompt specified by announce will be played to that agent before the caller is connected. This can be useful for agents who are logged in to more than one queue. You can specify either the full path to the file, or a path relative to /var/lib/asterisk/sounds/.


Asterisk can use six strategies to distribute calls to agents:


The queue rings every available agent and connects the call to whichever agent answers first (this is the default).

roundrobin (deprecated)

The queue cycles through the agents until it finds one who is available to take the call. roundrobin does not take into account the workload of the agents. Also, because roundrobin always starts with the first agent in the queue, this strategy is suitable only in an environment where you want your higher-ranked agents to handle all calls unless they are busy, in which case the lower-ranked agents may get a call.


The call is presented to the agent who has not been presented a call for the longest period of time.


The call is presented to the agent who has received the least amount of calls. This strategy does not take into account the actual agent workloads; it considers only the number of calls they have taken (for example, an agent who has had 3 calls that each lasted for 10 minutes will be preferred over an agent who has had 5 calls each lasting 2 minutes).


As its name suggests, the random strategy chooses an agent at random. In a small call center, this strategy may prove to be the most fair.


The queue cycles through the list of queue members, keeping track of which member last received a call. The next time a call needs to be distributed, Asterisk will continue from this point in the list of queue members. (This strategy is known as round-robin memory). This ensures that call presentation cycles through the agents as fairly as possible.


In a call center, the service level represents the maximum amount of time a caller should ideally have to wait before being presented to an agent. For example, if servicelevel is set to 60 and the service level percentage is 80 percent, that means 80 percent of the calls that came into the queue were presented to an agent in less than 60 seconds.


If a context is assigned to a queue, the caller will be able to press a single digit to exit to the corresponding extension within the configured context, if it exists. This action takes the caller out of the queue, which means that she will lose her place in line—be aware of this when you use this feature.


The timeout value defines the maximum amount of time (in seconds) to let an agent’s phone ring before deeming the agent unavailable and placing the call back into the queue.


When a timeout occurs, the retry value specifies how many seconds to wait before presenting the call again to an available agent.


The weight parameter assigns a rank to the queue. If calls are waiting in multiple queues, those queues with the highest weight values will be presented to agents first. When you are designing your queues, be aware that this strategy can prevent a call in a lower-weighted queue from ever being answered. Always ensure that calls in lower-weighted queues eventually get promoted to higher-weighted queues to ensure that they don’t have to hold forever.


maxlen is the maximum number of calls that can be added to the queue before the call goes to the next priority of the current extension.


The announce-frequency value (defined in seconds) determines how often to announce to the caller his place in the queue and estimated hold time.


There are three possible values for this parameter: yes, no, and once. The announce-holdtime parameter determines whether or not to include the estimated hold time within the position announcement. If set to once, it will be played to the caller only once.


This parameter accepts three possible values: wav, gsm, and wav49. By enabling this option, you are telling Asterisk that you wish to record all completed calls in the queue in the format specified. If this option is not specified, no calls will be recorded.


The Monitor() application in Asterisk normally records either end of the conversation in a separate file. Setting monitor-join to yes instructs Asterisk to merge the files at the end of the call. This should be set only if the MonitorType parameter is set to Monitor.


This parameter accepts three values: yes, no, and strict. It allows you to determine whether callers can be added to a queue based on the status of the members of the queue. The strict option will not allow callers to join the queue if all members are unavailable.


This parameter determines whether you want your holding callers to be removed from the queue when the conditions preventing a caller from joining exist (i.e., when all of your agents log out and go home).


Set eventwhencalled to yes if you wish to have queue events presented on the Manager Interface.


Setting this parameter to no will generate extra information pertaining to each queue member.


If you set this parameter to yes, the amount of time the caller held before being connected will be announced to the queue member answering the call.


This parameter defines whether a delay will be inserted between the time when the queue identifies a free agent and the time when the call is connected to that agent.

member => member_name

Members of a queue can be either channel types or agents. Any agents you list here must be defined in the agents.conf file.


The purpose of the res_odbc.so module is to store configuration file information in a database and retrieve that information from the database. The res_odbc.conf file specifies how to access the table within the database. The extconfig.conf file is used to determine how to connect to the database.


The res_snmp.conf file is used to configure SNMP support in Asterisk. There are two options in the [general]. The subagent option specifies whether res_snmp should run as a subagent or a full SNMP agent. Asterisk defaults to running as a subagent. The enabled option specifies whether SNMP support in Asterisk is enabled. This defaults to no, so you’ll need to change this if you want SNMP support.


The rpt.conf file is used to configure Jim Dixon’s newest project, Jim’s Radio Repeater Application (app_rpt), allows Asterisk to communicate using VoIP via radio repeater technology. This allows people to efficiently provide large-area coverage of wireless networking and routing information to the amateur radio public through their local high-speed Internet connections.


The rtp.conf file controls the Real-time Transport Protocol (RTP) ports that Asterisk uses to generate and receive RTP traffic. The RTP protocol is used by SIP, H.323, MGCP, and possibly other protocols to carry media between endpoints.

The default rtp.conf file uses the RTP port range of 10,000 through 20,000. However, this is far more ports than you’re likely to need, and many network administrators may not be comfortable opening up such a large range in their firewalls. You can limit the RTP port range by changing the upper and lower bound limits within the rtp.conf file.

For every bidirectional SIP call between two endpoints, five ports are generally used: port 5060 for SIP signaling, one port for the data stream and one port for the Real-Time Control Protocol (RTCP) in one direction, and an additional two ports for the data stream and RTCP in the opposite direction.

UDP datagrams contain a 16-bit field for a Cyclic Redundancy Check (CRC), which is used to verify the integrity of the datagram header and its data. It uses polynomial division to create the 16-bit checksum from the 64-bit header. This value is then placed into the 16-bit CRC field of the datagram, which the remote end can then use to verify the integrity of the received datagram.

Setting rtpchecksums=no requests that the OS not do UDP checksum creating/checking for the sockets used by RTP. If you add this option to the sample rtp.conf file, it will look like this:



The say.conf file is used to configure spoken language grammar rules for a number of applications, such as SayNumber(). If you’re looking to use Asterisk in a language that isn’t currently supported, you can script support through the configuation options in this file.


The sip.conf file defines all the SIP protocol options for Asterisk. The authentication for endpoints, such as SIP phones and service providers, is also configured in this file. Asterisk uses the sip.conf file to determine which calls you are willing to accept and where those calls should go in relation to your dialplan. Many SIP-related options are configured in sip.conf, which was covered in depth in Appendix A.


Asterisk has the ability to remotely notify a SIP phone to recheck its configuration files or reboot by sending it a specially formatted, manufacturer-specific NOTIFY message (defined in sip_notify.conf). Because each of these messages is manufacturer-specific, support varies from phone to phone.


If you wish to connect to phones using Cisco’s proprietary Skinny Client Control Protocol (SCCP), you can use the skinny.conf file to define the parameters and channels that will use it. However, since the SCCP protocol is proprietary, you may find that support in Asterisk for this protocol is less than perfect but steadily improving over time.


Even though Asterisk is a modern PBX, many people still want it to behave like an old key system with shared lines. Asterisk can emulate a key system by configuring shared lines in sla.conf.


This file configures the Station Message Desk Interface. SMDI is a very useful addition to Asterisk, as it will allow it to act as a voicemail system for legacy PBXes that support the SMDI protocol.


This file is used for the configuration of Asterisk’s support for UDPTL packets. UDPTL packets are one of the transports used by T.38 faxing over IP connections.


With the advent of the Asterisk GUI, the Asterisk developers found it would be helpful to create a configuration file where user accounts can be specified, instead of having different pieces spread across a myriad of files (such as extensions.conf, sip.conf, and voicemail.conf. This file is also updated by the Asterisk GUI when new users are added to the system or when user settings are modified.


The voicemail.conf file controls the Asterisk voicemail system (called Comedian Mail). It consists of three main sections. The first, called [general], sets the general system-wide settings for the voicemail system. The second, called [zonemessages], allows you to configure different voicemail zones, which are a collection of time and time zone settings. The third and final section is where you create one or more groups of voicemail boxes, each containing the mailbox definitions. For more information on adding voicemail capabilities to your dialplan, see Chapter 6.

General Voicemail Settings

The [general] section of voicemail.conf contains a plethora of options that affect the entire voicemail system:


Lists the codecs that should be used to save voicemail messages. Codecs should be separated with the pipe character (|). The first format specified is the format used when attaching a voicemail message to an email. Defaults to wav49|gsm|wav. The reason that the voicemail might be saved in several different formats is to minimize the amount of transcoding that Asterisk does when the voicemail is played back.


Provides the email address from which voicemail notifications should be sent.


Specifies whether or not Asterisk should attach the voicemail sound file to the voicemail notification email.


Sets the maximum number of messages that may be kept in any voicemail folder.


Sets the maximum length of a voicemail message, in seconds.


Sets the minimum length of a voicemail message, in seconds.


Sets the maximum length of voicemail greetings, in seconds.


Specifies how many milliseconds to skip forward/back when the user skips forward or backward during message playback.


Indicates how many seconds of silence to allow before ending the recording.


Sets the silence threshold (what we consider “silence”—the lower the threshold is, the more sensitive it is).


Sets the maximum allowed number of failed login attempts.


Specifies which voicemail context the mailboxes defined in users.conf should be a part of. This defaults to the default voicemail context.


Supplies the full path and filename of an external program to be executed when a voicemail is left or delivered, or when a mailbox is checked. It can also be set to smdi to use SMDI for external notification. If it is smdi, the smdiport should be set to a valid port as specified in smdi.conf.


Specifies the communications port used by SMDI. This should correspond to a valid port as specified in smdi.conf. This is used when the externnotify setting is set to smdi.


Supplies the full path and filename of an external program to be executed whenever a voicemail password is changed.


If set, overrides the default introduction to the dial-by-name directory.


Defines the character set for voicemail messages.


Specifies the ADSI feature descriptor number to download to.


Sets the ADSI security lock code.


Indicates the ADSI voicemail application version number.


Causes Asterisk not to add the string [PBX]: to the beginning of the subject line of a voicemail notification email.


Changes the From: string of voicemail notification email messages.


Permits a mailbox owner to select entries from the dial-by-name directory for forwarding and/or composing new voicemail messages.


If support for ODBC voicemail storage has been compiled into Asterisk, this option allows you to specify which ODBC connection to use. ODBC connections are defined in res_odbc.conf.


This option is used in conjunction with the odbcstorage option. This option specifies which database table to use for voicemail messages.


Specifies the email subject of voicemail notification email messages.


Changes the From: string of voicemail notification pager messages.


Supplies the email body of voicemail notification email messages.


Please note that all the emailsubject, emailbody, pagersubject, and pagerbody settings can use the following variables to provide more in-depth information about the voicemail:


  • VM_DUR








Specifies the format of the date and time for outbound email notifications. For more information on the format, see the strftime(3) manpage.


Supplies the full path and filename of the program Asterisk should use to send notification emails. This option is useful if you want to override the default email program.


Skips to the next message after the user hits 7 or 9 to delete or save the current message. This can be set only globally at this time, not on a per-mailbox basis.

Voicemail Zones

As voicemail users may be located in different geographical locations, Asterisk provides a way to configure the time zone and the way the time is announced for different callers. Each unique combination is known as a voicemail zone. You configure your voicemail zones in the [zonemessages] section of voicemail.conf. Later, you can assign your voicemail boxes to use the settings for one of these zones.

Each voicemail zone definition consists of a line with the following syntax:

zonename=timezone |time_format

The zonename is an arbitrary name used to identify the zone. The timezone argument is the name of a system time zone, as found in /usr/share/zoneinfo. The time_format argument specifies how times should be announced by the voicemail system. The time_format argument is made up of the following elements:

' filename '

The filename of a sound file to play (single quotes around the filename are required)

${ VAR }

Variable substitution

A or a

The day of the week (Saturday, Sunday, etc.)

B or b or h

The name of the month (January, February, etc.)

d or e

The numeric day of the month (first, second... thirty-first)


The year

I or l

The hour, in 12-hour format


The hour, in 24-hour format—single-digit hours are preceded by “oh”


The hour, in 24-hour format—single-digit hours are not preceded by “oh”


The minute

P or p

A.M. or .P.M.


“today”, “yesterday,” or ABdY (note: not standard strftime value)


“” (for today), “yesterday”, weekday, or ABdY (note: not standard strftime value)


24-hour time, including minutes

For example, the following example sets up two different voicemail zones, one for the Central time zone in 12-hour format, and a second in the Mountain time zone, in 24-hour format:

central=America/Chicago|'vm-received' Q 'digits/at' IMp
mountain24=America/Denver|'vm-received' q 'digits/at' H 'digits/hundred' M 'hours'

Defining Voicemail Contexts and Mailboxes

Now that the system-wide settings and voicemail zones have been set, you can define your voicemail contexts and individual mailboxes.

Voicemail contexts are used to separate out different groups of voicemail users. For example, if you are using Asterisk to host voicemail for more than one company, you should place each company’s mailboxes in different voicemail contexts, to keep them separate. You might also use voicemail contexts to create per-department dial-by-name directories.

To define a new voicemail context, simply put the context name inside of square brackets, like this:


Inside a voicemail context, each mailbox definition takes the following syntax:


The mailbox argument is the mailbox number.

The password argument is the numeric code the mailbox owner must enter to access his voicemail. If the password is preceded by a minus sign (-), the password may not be changed by the mailbox owner.


If the password is set to d, then this line is assumed to provide an alternate name that can be used for this mailbox in the dial-by-name directory. In the following example, people can reach extension 123 by searching for either Robert or Bob, along with a popular misspelling of his last name:

123 => 4444,Robert Schauerhamer
123 => d,Bob Schauerhamer
123 => d,Robert Showerhammer

The email and pager_email arguments are email addresses where voicemail notifications will be sent. These may be left blank if you don’t want to send voicemail notifications via email. The message sent to the pager_email address is usually shorter and suitable for sending to a cell phone (via an email to SMS gateway) or to an alpha-numeric pager.

The options argument is a pipe-separated list of voicemail options that may be specified for the mailbox. (These options may also be set globally by placing them in the [general] section.) Valid voicemail options include:


Sets the voicemail zone from the [zonemessages] section earlier. This option is irrelevant if envelope is set to no.


Attaches the voicemail to the notification email (but not to the pager email). May be set to either yes or no.


Specifies the format of the voicemail message that should be attached to the message sent to the notification email. Ordinarily, Asterisk uses the first format specified in the format parameter listed in [general] (covered earlier), but this may also be changed on a per-mailbox basis. This option may be set only on a per-mailbox basis.

This option is often used if Windows users want wav49 attachments but Linux users want their attachments in gsm format.


Says the Caller ID information before the message.


Sets the internal context for name playback instead of extension digits when saying the Caller ID information.


Turns on/off the duration information before the message. Defaults to on.


Specifies the minimum duration to say when sayduration is on. Default is two minutes.


Specifies the context to dial out from (by choosing option 4 from the advanced menu). If not specified, dialing out from the voicemail system will not be permitted.


Specifies the context to send voicemail from (by choosing option 5 from the advanced menu). If not specified, sending messages from within the voicemail system will not be permitted.


By default, Asterisk searches only the default context if no context is specified. To have Asterisk search all contexts, set this option to yes.


Specifies the context to call back from. If not specified, calling the sender back from within the voicemail system will not be permitted.


Allows senders to review/re-record their messages before saving them. Defaults to off.


Allows senders to hit 0 before, after, or while leaving a voicemail message to reach an operator. Defaults to off.


Turns on/off envelope playback before message playback. Defaults to on. This does not affect option 3,3 from the advanced options menu.


Deletes voicemails from the server after notification is sent. This option may be set only on a per-mailbox basis; it is intended for use with users who wish to receive their voicemail messages only by email.


If your voicemail attachments to email are too quiet, you can set this option to increase the gain on the message before it is attached to the email notification.


This option only works if the sox application has been installed on your Asterisk system.


Forces new users to record their names. A new user is determined by the password being the same as the mailbox number. Defaults to no.


Forces new users to record greetings. A new user is determined by the password being the same as the mailbox number. Defaults to no.


Hides the mailbox from the dial-by-name directory. Defaults to no.


Warns users that their temporary greeting is still enabled.

You can specify multiple options by separating them with the pipe character, as shown in the definitions for mailboxes 102 and 103 below.

Here are some sample mailbox definitions:

; regular mailbox with email notification
101 => 4242,Example Mailbox,somebody@asteriskdocs.org

; more advanced mailbox with email and pager notification and a couple of
; special options
102 => 9855,Another User,another@asteriskdocs.org,pager@asteriskdocs.org,

; a mailbox with no email notification and lots of extra options
103 => 6522,John Q. Public,,,tz=central|attach=yes|saycid=yes|dialout=fromvm|


This file is used to configure Voicetronix cards with Asterisk. See the sample vpb.conf for more information.


The zapata.conf file is used to define the relationship between Asterisk and the Zaptel driver. Because zapata.conf is specific to Asterisk, it is located with the other Asterisk configuration files in /etc/asterisk/. As with zaptel.conf, the zapata.conf file contains a multitude of choices reflecting the multitude of hardware it supports, and we won’t try to list all of the options here. In this book we’ve covered only the analog interfaces to the Zaptel driver, as described in Chapter 3.


The zaptel.conf file is not located with the other Asterisk .conf files; the Zaptel driver is available to any application that can make use of it, so it makes more sense to store it in a non-Asterisk-specific directory (/etc/). zaptel.conf is parsed by the ztcfg program to configure the TDM hardware elements in your system. You configure three main elements in the zaptel.conf file:

  • A way of identifying the interfaces on the card within the dialplan

  • The type of signaling the interface requires

  • The tone language associated with a particular interface, as found in zonedata.c


Be very careful not to plug your FXS module into a telephone line. The voltage associated with the phone line, especially during an incoming call, will be much too high for the module to handle and may permanently damage it, rendering it useless!

Within the zaptel.conf file, we define the type of signaling that the channel is going to use. We also define which channels to load. The options in the configuration file are the information that will be used to configure the channels with the ztcfg command.

The actual parameters available in the zaptel.conf file are quite extensive, as a wide variety of PSTN interfaces make use of the Zaptel telephony engine. Also, as this technology is rapidly evolving, anything we write now may not be accurate by the time you read it. Consequently, we won’t try to list all of the options here.

In this book, we have focused on the Zaptel analog interfaces as provided by the Digium TDM400P card (see Chapter 3).

[164] With the advent of the new menuselect system, this best practice may no longer be necessary if you are building only the modules you need in the first place

[165] Yes, we are aware that the user interface to the channel interface is the Asterisk CLI; however, this is not usable as a telephone and therefore does not meet the criteria of an interface from the perspective of a telephone user.

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