To install Asterisk on a server, simply use the following:

include asterisk

This will install a plain version of Asterisk without any extra features enabled.

Users that are upgrading (e.g. switching or merging to current master) should consult the section named Upgrade notices near the end.

In order to use this module, you need the stdlib module from:

https://github.com/puppetlabs/puppetlabs-stdlib

You should also make sure that augeas is installed since it is used to enable the service in /etc/default/asterisk.

Some good references to consult when it comes to Asterisk configuration are:

• The most up to date reference and documentation for all configurations can be found in upstream default configuration files. They can also be found in the asterisk-doc debian package.
• Online version of "Asterisk: The Definitive Guide" 3rd edition. This is definitely the best and most up to date documentation available. A must read for anyone that is configuring a PBX with Asterisk. Consult this reference if you need more information about any options that can be configured with this module. The web site mentions a 4th edition was released but it is not available online: http://asteriskdocs.org/
• A good reference for VoIP and Asterisk (some information might be outdated): http://www.voip-info.org/
• The Asterisk project wiki: https://wiki.asterisk.org/

The main class has a couple of parameters that determine what is managed and how general configuration is set.

• $manage_service is a boolean that determines whether puppet will ensure that the service is running. Default value is true.

• $manage_package is a boolean that determines whether puppet will ensure that the package is installed. Default value is true.

• $package_name can be used to override the name of the package that installs Asterisk. Default value is "asterisk".

• $service_name can be used to override the name of the Asterisk service. Default value is "asterisk".

• $confdir can be used to override the path to the Asterisk configuration. Default value is "/etc/asterisk".

• $purge_confdir is a boolean that enables or disables (default) removal of all files inside confdir that are not managed by puppet. This makes sure that the state of your service is well known.

• $iax_general is a hash of global options for IAX2. See section IAX2 Options.

• $sip_general is a hash of global options for SIP. See section SIP Options.

• $voicemail_general is a hash of global options for voicemail. See section Voicemail Options.

• $extensions_general is a hash of global options for extensions. See section Extensions Options.

  Options section.

• $features_general and $featuremap are detailed in the Features Options section.

• $queues_general is detailed in the Queues Options section.

• $modules_autoload, $modules_noload, $modules_load and $modules_global are detailed in the Modules section.

• $manager_enable, $manager_port and $manager_bindaddr are detailed in the Manager Options section.

Asterisk has lots and lots of configuration variables that can be set in different files.

As you will see in some of the following configuration sections, some configuration files will be customizable through option hashes. The format of those hashes is always the same and looks like the following, where xyz would match the name of the configuration file:

$xyz_general = {
  'configuration-option1' => 'value1',
  'allow'                 => ['list-value1', 'list-value2'],
  #[...]
}

In order to simplify the module, we're actually not validating that the options passed in are valid ones and expect this validation to be done by the user.

We encourage users to use strings as hash keys as in the example above since some Asterisk options have dashes in their name and dashes are prohibited in puppet DSL symbols.

Some options should always be arrays: the option can be specified in the configuration file more than once to declare more values. Those options will always be set in the hashes that define default values (see in each section below) as arrays either containing a number of strings, or being empty. The module enforces that those options be arrays since it needs to iterate over them in templates. Empty arrays mean that the option should not appear in the configuration file.

Default values are taken from Debian's default configuration files.

Keys that are present in the option hash paramters to the asterisk class will override the default options (or set new ones for options that are not present in the default option hash). This lets you use all the default values but change only a couple of values.

Most of the defined types that drop a configuration file in a .d directory can either take a puppet source specification (of the form 'puppet:///modules/...' or a textual content.

The puppet source specification is always used with the source parameter and textual content with the content parameter.

When using a puppet source specification the user has complete control over the contents of the configuration file. When textual content is used, the contents will usually be added after a line that defines a configuration section (e.g. a line of the form '[section]').

source and content are always mutually exclusive.

The asterisk::iax defined type helps you configure an IAX2 channel. source or content can be used with this type.

asterisk::iax { '5551234567':
  source => 'puppet:///modules/site_asterisk/5551234567',
}

The asterisk::registry::iax defined type is used to configure an IAX2 registry. All parameters (except for ensure) are mandatory. For example:

asterisk::registry::iax { 'providerX':
  server => 'iax.providerX.com',
  user   => 'doyoufindme',
  pass   => 'attractive?',
}

If you are using the IAX2 protocol, you'll want to set some global configuration options. For passing in settings, you need to send a hash to the asterisk class with the iax_general parameter.

Here is the default hash with the default values, as defined in params.pp:

$iax_general = {
  'allow'             => [],
  'disallow'          => ['lpc10'],
  'bandwidth'         => 'low',
  'jitterbuffer'      => 'no',
  'forcejitterbuffer' => 'no',
  'autokill'          => 'yes',
  'delayreject'       => 'yes',
}

You can configure SIP channels with the asterisk::sip defined type. source and content can be used with this type.

asterisk::sip { '1234':
  ensure  => present,
  secret  => 'blah',
  context => 'incoming',
}

You can also use the template_name argument to either define a template, or make the channel definition inherit from a template.

To define a template, set template_name to '!':

asterisk::sip { 'corporate_user':
  context       => 'corporate',
  type          => 'friend',
  # ...
  template_name => '!',
}

If inheriting from a template, set template_name to the name of the template from which the channel is inheriting options.

asterisk::sip { 'hakim':
  secret        => 'ohnoes!',
  template_name => 'corporate_user',
}

The defined type asterisk::registry::sip lets you configure a SIP registry. The server and user paramters are mandatory.

asterisk::registry::sip { 'providerX':
  server => 'sip.providerX.com',
  user   => 'doyoufindme',
}

Password, authuser, port number and extension are optional parameters. If you define authuser, you must specify a password.

asterisk::registry::sip { 'friends_home':
  server    => 'home.friend.com',
  port      => '8888',
  user      => 'me',
  password  => 'myselfandI',
  authuser  => 'you',
  extension => 'whatsupfriend',
}

If you are using the SIP protocol, you'll want to set some global configuration options. For passing in settings, you need to send a hash to the asterisk class with the sip_general parameter.

Here is the default hash with the default values, as defined in params.pp:

$sip_general = {
  'disallow'         => [],
  'allow'            => [],
  'domain'           => [],
  'localnet'         => [],
  'context'          => 'default',
  'allowoverlap'     => 'no',
  'udpbindaddr'      => '0.0.0.0',
  'tcpenable'        => 'no',
  'tcpbindaddr'      => '0.0.0.0',
  'transport'        => 'udp',
  'srvlookup'        => 'yes',
  'allowguest'       => 'no',
  'alwaysauthreject' => 'yes',
}

If you want to enable SIP encryption, you can set the following settings in the sip_general parameter to the asterisk class:

$sip_option = {
  'transports'          => ['tls'],
  'encryption'          => 'yes',
  'tlsenable'           => 'yes',
  # Change the following two values to the full paths where you're placing your
  # own certificat and CA files, respectively.
  'tlscertfile'         => '/etc/ssl/somecert.crt',
  'tlscafile'           => '/etc/ssl/someca.crt',
  # Only set this to 'yes' if you can't possibly get a verifiable certificate.
  'tlsdontverifyserver' => 'no',
}

Note: the 'transports' option needs to be an array, so even though you only enable 'tls' as a transport, you need to enclose the string inside an array.

With the defined type asterisk::voicemail you can configure a voicemail. The context and password parameters are mandatory:

asterisk::voicemail { '3000':
  context   => 'some_context',
  password  => '5555',
  user_name => 'Bob Bobby',
  email     => '[email protected]',
}

You can also use the optional 'pager_email' parameter to set the email that should receive a page about new voice messages.

And finally, the argument 'options' can take a hash of voicemail options like the following:

asterisk::voicemail { '3001':
  context  => 'blah',
  password => '112233',
  options  => { 'attach' => 'yes', 'delete' => 'yes' },
}

Voicemail can be configured through a set of options in the [general] context. To set those options, you can pass values as a hash to the voicemail_general parameter to the main class.

Here is the default hash with the default values, as defined in params.pp:

$voicemail_general = {
  'format'           => 'wav49|gsm|wav',
  'serveremail'      => 'asterisk',
  'attach'           => 'yes',
  'skipms'           => 3000,
  'maxsilence'       => 10,
  'silencethreshold' => 128,
  'maxlogins'        => 3,
  # This is not really the default value for emailbody but it makes more
  # sense to be a bit more verbose by default.
  'emailbody'        => 'Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just ${IF($["${VM_CIDNUM}" = "${ORIG_VM_CIDNUM}"]?left:forwarded)} a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID} <${VM_CIDNUM}>, on ${VM_DATE},\n${IF($["${VM_CIDNUM}" = "${ORIG_VM_CIDNUM}"]?so:(originally sent by ${ORIG_VM_CALLERID} on ${ORIG_VM_DATE})\nso)} you might want to check it when you get a chance.  Thanks!\n\n\t\t\t\t--Asterisk\n',
  'emaildateformat'  => '%A, %B %d, %Y at %r',
  'pagerdateformat'  => '%A, %B %d, %Y at %r',
  'sendvoicemail'    => 'yes',
}

Extensions can be set with the asterisk::extensions defined type. source or content can be used with this type.

asterisk::extensions { 'incoming':
  ensure  => present,
  content => template('site_asterisk/extensions/incoming.erb'),
}

Some global options can be set for extensions. You can achieve that by passing a hash to the extensions_general parameter to the asterisk class.

Here is the default hash with the default values, as defined in params.pp:

$extensions_general = {
  'static'          => 'yes',
  'writeprotect'    => 'no',
  'clearglobalvars' => 'no',
}

Note that by default no global variables (e.g. values set in the [globals] context) are set. To set global variables, you can use an asterisk::extensions resource with a context value of "globals".

To define an agent you can use the asterisk::agent defined type. The ext, password and agent_name parameters are mandatory.

To define a static agent:

asterisk::agent { 'joe':
  ext        => '1001',
  password   => '123413425',
  agent_name => 'Joe Bonham',
}

You can also assign a static agent to one or more agent groups with the groups parameter. This parameter is a list of group names:

asterisk::agent { 'cindy':
  ext        => '1002',
  password   => '754326',
  agent_name => 'Cindy Rotterbauer',
  groups     => ['1']
}

Static agents have some disadvantages compared to dynamic agents. For example, once assigned to a queue they cannot logout of that queue. For more information on how to setup dynamic agents, see:

• https://www.voip-info.org/asterisk-cmd-agentlogin
• https://www.voip-info.org/asterisk-cmd-addqueuemember
• https://www.voip-info.org/asterisk-cmd-removequeuemember

Global options in the [agents] context can be set by passing a hash to the agents_global parameter to the asterisk class. By default this parameter doesn't define any global options.

For creating agents, it is recommended to use the asterisk::agent defined type.

Features let you configure call parking and special numbers that trigger special functionality. The asterisk::feature defined type helps you configuring such features. The options parameter is mandatory.

Define features that are contained within feature group "myfeaturegroup":

$ft_options = {
  'pausemonitor'   => '#1,self/callee,Pausemonitor',
  'unpauseMonitor' => '#3,self/callee,UnPauseMonitor',
}
asterisk::feature { 'myfeaturegroup':
  options => $ft_options,
}

A special section in the features configuration file, namely [applicationmaps] lets you define global features. The asterisk::feature::applicationmap defined type helps you configure such a global feature. The feature and value parameters are mandatory:

asterisk::feature::applicationmap { 'pausemonitor':
  feature => 'pausemonitor',
  value   => '#1,self/callee,Pausemonitor',
}

Some global feature options can be configured, like the default parkinglot, via the features_general parameter to the asterisk class.

Here is the default hash with the default values, as defined in params.pp:

$features_general = {
  'parkext' => '700',
  'parkpos' => '701-720',
  'context' => 'parkedcalls',
}

A special context, featuremap, lets you configure global features. By default, no feature is configured. You can pass a hash to the features_featuremap parameter to the asterisk class to configure features in this context.

Another special context, applicationmap, lets you configure dynamic features. To set entries in this context, you should use the asterisk::feature::applicationmap defined type. Note also that for dynamic features to work the DYNAMIC_FEATURES channel variable must be set by listing features enabled in the channel, separated by '#'.

To configure additional feature contexts, you can use the asterisk::feature defined type.

Asterisk can put call in queues, for example when all agents are busy and the call cannot get connected. To create a queue, you can use the asterisk::queue defined type:

asterisk::queue { 'frontline':
  ensure   => present,
  stragegy => 'rrmemory',
  members  => [
    'SIP/reception',
    'SIP/secretary',
  ],
  maxlen   => 30,
  timeout  => 20,
  retry    => 10,
}

Call queues have lots of options and can interact with agents. Because of this we will not detail all of the parameters here. Please refer to the manifests/queue.pp file for the complete list of supported parameters. Also, for an in-depth coverage of call queueing, see: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-ACD.html

For queues some global configurations and default values can be set in the [general] context. You can set options by passing a hash to the queues_general parameter to the asterisk class.

Here is the default hash with the default values, as defined in params.pp:

$queues_general = {
  'persistentmembers' => 'yes',
  'monitor-type'      => 'MixMonitor',
}

Configuring Asterisk modules is key to implementing your features right. Four parameter to the asterisk class offer you the possibility to customize what modules are loaded or not on your PBX. Default values for the parameters were taken from the default config file in Debian.

• modules_autoload: a boolean value (defaults to true) that decides whether or not Asterisk will try to automatically load required modules even though they are not explicitely marked as needing to be loaded in the modules.conf file.

• modules_noload: an array of strings of explicitely unwanted modules that won't load even though modules_autoload is true. Specifying an array to this parameter overrides the default list so make sure to include all unwanted modules. The default array is the following:

  $modules_noload = [
    'pbx_gtkconsole.so',
    'pbx_kdeconsole.so',
    'app_intercom.so',
    'chan_modem.so',
    'chan_modem_aopen.so',
    'chan_modem_bestdata.so',
    'chan_modem_i4l.so',
    'chan_capi.so',
    'chan_alsa.so',
    'cdr_sqlite.so',
    'app_directory_odbc.so',
    'res_config_odbc.so',
    'res_config_pgsql.so'
  ]

• modules_load: an array of strings of explicitely wanted modules. Specifying an array to this parameter overrides the default list so make sure to include all wanted modules. The default array is the following:

  $modules_load = ['res_musiconhold.so']

• modules_global: a hash of options that should be set in the [global] context. These options let you customize behaviours for modules that are loaded.

Asterisk can expose an interface for managing the PBX. This interface can be offered to different users with different permissions. You can configure read and write access to certain features of the PBX for each user.

The asterisk::manager defined type helps you configure a manager access. The secret parameter is mandatory. By default, the resource name is used as the manager name:

asterisk::manager { 'nagios':
  secret => 'topsecret1234',
  read   => ['all'],
  write  => ['system', ' call', ' log', ' verbose', ' command', ' agent', ' user'],
}

Here's a paranoid version of the above configuration, with minimal network access, but the option to run system commands and trigger calls:

asterisk::manager { 'nagios':
  secret => 'topsecret1234',
  read   => ['system', 'call'],
  write  => ['system', 'call'],
}

Here, we permit remote management to two other systems on an internal network:

asterisk::manager { 'robocall':
  secret => 'robotsdeservesomeloveafterall',
  permit => ['10.10.10.200/255.255.255.0', '10.20.20.200/255.255.255.0'],
  read   => ['system', 'call', 'log'],
  write  => ['system', 'call', 'originate'],
}

To override the manager name, you can use the manager_name parameter:

asterisk::manager { 'sysadmin':
  secret       => 'nowyouseemenowyoudont',
  read         => ['all'],
  write        => ['all'],
  manager_name => 'surreptitioustyrant',
}

Asterisk maintains a service on a port through which you can inspect asterisk's state and issue commands to the PBX. You can control on which IP and port it binds to and if it is enabled at all with three parameters to the asterisk class.

• manager_enable: a boolean value that decides whether or not the manager is in function. Defaults to true.

• manager_port: an integer value that specifies on which port the manager will listen. Default value is 5038.

• manager_bindaddr: a string that contains the IP address on which the manager should bind. Default value is 127.0.0.1.

By default, no user access is configured. If you want to enable users to interact with the manager, you should declare asterisk::manager resources.

Dahdi is a set of kernel modules combined with an asterisk module that let people interact with Digium cards to send and receive calls from the POTS. To enable dahdi, use the following:

  include 'asterisk::dahdi'

To include any language sounds, you can use the following (in this example, we're installing french and spanish sounds):

  asterisk::language {
    ['fr-armelle', 'es']:
  }

Valid languages strings are the following (these are all based on debian package names for now -- either asterisk-prompt-X or asterisk-Y. the language strings that start with core-sounds enable you to install language sounds in a specific encoding to avoid the need for asterisk to recode it while feeding it to a device):

• de
• es-co
• fr-armelle
• fr-proformatique
• it-menardi
• it-menardi-alaw
• it-menardi-gsm
• it-menardi-wav
• se
• es
• core-sounds-en
• core-sounds-en-g722
• core-sounds-en-gsm
• core-sounds-en-wav
• core-sounds-es
• core-sounds-es-g722
• core-sounds-es-gsm
• core-sounds-es-wav
• core-sounds-fr
• core-sounds-fr-g722
• core-sounds-fr-gsm
• core-sounds-fr-wav
• core-sounds-ru
• core-sounds-ru-g722
• core-sounds-ru-gsm
• core-sounds-ru-wav

• The module used to manage files under /etc/asterisk/file.conf.d for all values of "file" that were managed. Things have been moved to /etc/asterisk/file.d, so before upgrading you should remove all .conf.d directories (all files under the old dirs will be automatically recreated in the new directories).

• The defines that were previously named asterisk::context::xyz (or transitorily asterisk::snippet::xyz) are now named asterisk::xyz. Users will need to adjust their manifests to upgrade.

• The queues_monitor_type and queues_monitor_format parameters to the default class were removed in favor of using quoted strings in the options array. Users who used those two options need to place their values in the $queues_general hash with 'monitor-type' and 'monitor-format' strings as keys, respectively. To ensure that 'monitor-type' is not present in the config file, simply leave it out (as opposed to the previous behaviour of the option that required an empty string for this).

• Some default values were removed and some others were modified to be closer to default Debian config files. You should verify that new values or variables that disappear won't have an impact on your setup.

Contributions are highly welcomed, more so are those which contribute patches with tests. Or just more tests! We have rspec-puppet tests. When [contributing patches](Github WorkFlow), please make sure that your patches pass tests:

user@host01 ~/src/bw/puppet-composer (git)-[master] % rake tests
....................................

Finished in 2.29 seconds
36 examples, 0 failures

Types:

• asterisk::mwi

This module is licensed under the GPLv3 , feel free to redistribute, modify and contribute changes.

A copy of the GPLv3 license text should be included with the module. If not, check out the github repository at https://github.com/lelutin/puppet-asterisk or one of its clones.

The license text can also be downloaded from:

https://www.gnu.org/licenses/gpl-3.0.txt