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Asterisk
========
Description
-----------
The `asterisk` role installs and configures the
[Asterisk](https://www.asterisk.org/) PBX.
Variables
---------
This role **accepts** the following variables:
Variable | Default | Description
---------------------------------------|----------------------------------------------------|------------
`asterisk_local_nets` | `['10.0.0.0/8', '172.16.0.0/12', 192.168.0.0/16']` | Networks local to the Asterisk server (see [documentation](https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Configuration_res_pjsip#Asterisk17Configuration_res_pjsip-transport_external_media_address))
`asterisk_external_ip` | | External IP address of the Asterisk server (see [documentation](https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Configuration_res_pjsip#Asterisk17Configuration_res_pjsip-transport_external_media_address))
`asterisk_timezone` | `{{ timezone }}` | Timezone used for voicemail metadata
`asterisk_locale` | `en_US.UTF-8` | Locale used for voicemail prompts
`asterisk_public_fqdn` | `{{ ansible_fqdn }}` | Public FQDN of the Asterisk server (used for SIP TLS client certificate)
`asterisk_from_domain` | `{{ email_domain }}` | Domain used in SIP `From` header
`asterisk_rtp_port_start` | 10000 | Minimum port used for RTP dynamic port range
`asterisk_rtp_port_end` | 10999 | Maximum port used for RTP dynamic port range
`asterisk_sip_port` | 5060 | Source port for SIP connections
`asterisk_sip_tls_port` | 5061 | Source port for SIP TLS connections
`asterisk_http_port` | 8088 | Listening port for the Asterisk HTTP server
`asterisk_https_port` | 8089 | Listening port for the Asterisk HTTP server (HTTPS)
`asterisk_voicemail_formats` | `['wav49', 'gsm', 'wav']` | Audio formats used for voicemail recordings
`asterisk_mail_from` | `asterisk-noreply@{{ email_domain }}` | Email address used for new voicemail notifications
`asterisk_voicemail_email_subject` | see [default vars](defaults/main.yml) | Email subject line for new voicemail notifications
`asterisk_voicemail_email_body` | see [default vars](defaults/main.yml) | Email body used for new voicemail notifications
`asterisk_voicemail_email_date_format` | `%A, %B %d, %Y at %r` | Email date format used for new voicemail notifications
`asterisk_voicemail_min_password` | 4 | Minimum voicemail password length
`asterisk_voicemail_max_message_count` | 100 | Maximum number of messages in each voice mailbox
`asterisk_voicemail_max_message_secs` | 300 | Maximum length of voicemail message (seconds)
`asterisk_voicemail_max_greeting_secs` | 60 | Maximum length of voicemail greeting (seconds)
`asterisk_voicemail_max_failed_logins` | 3 | Maximum number of failed voicemail logins before lockout
`asterisk_dialplan` | | Raw dialplan to place in `extensions.conf` (see [documentation](https://wiki.asterisk.org/wiki/display/AST/Contexts%2C+Extensions%2C+and+Priorities))
`asterisk_password_salt` | | Salt used for generating SHA-512 password hashes
`asterisk_sip_trunks` | `[]` | Upstream SIP trunks (see [format](#asterisk_sip_trunks) below)
`asterisk_sip_extensions` | `[]` | Local SIP extensions (see [format](#asterisk_sip_extensions) below)
`asterisk_queues` | `[]` | Local call queues (see [format](#asterisk_queues) below)
`asterisk_ari_users` | `[]` | User accounts for the Asterisk REST Interface (see [format](#asterisk_ari_users) below)
`asterisk_voicemail_contexts` | `{}` | Dictionary of [voicemail contexts](https://wiki.asterisk.org/wiki/display/AST/Configuring+Voice+Mail+Boxes) (see [format](#asterisk_voicemail_contexts) below)
### asterisk\_sip\_trunks
The `asterisk_sip_trunks` variable describes your upstream SIP trunks. It should
contain a list of dictionaries of the following format:
Key | Default | Description
-------------------|---------|------------
`name` | | Name of the trunk (no spaces)
`transport` | `udp` | Network transport
`host` | | Remote SIP host (comma-separated `host:port`, or list)
`codecs` | | Allowed codecs (comma-separated, or list)
`media_encryption` | `"no"` | Force media encryption (see [documentation](https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-endpoint_media_encryption))
`username` | | SIP account username
`password` | | SIP account password
Note that incoming calls from a SIP trunk will be assigned the dialplan context
`from-TRUNKNAME`.
### asterisk\_sip\_extensions
The `asterisk_sip_extensions` variable describes your internal SIP extensions
(AKA your internal phone numbers). It should contain a list of dictionaries of
the following format:
Key | Default | Description
-------------------|------------------|------------
`name` | | Name of the extension (usually a 3- or 4-digit number)
`codecs` | | Allowed codecs (comma-separated, or list)
`context` | | Dialplan context for inbound calls
`mailbox` | | Mailbox name (comma-separated, or list)
`cid_name` | | Caller ID name
`cid_number` | `{{ ext.name }}` | Caller ID number
`username` | `{{ ext.name }}` | Extension SIP username
`password` | | Extension SIP password
`max_contacts` | 1 | Maximum simultaneous logins
### asterisk\_queues
The `asterisk_queues` variable describes your [call queues](https://wiki.asterisk.org/wiki/display/AST/Building+Queues).
It should contain a list of dictionaries of the following format:
Key | Default | Description
-------------------------|------------------|------------
`name` | | Name of the queue (no spaces)
`music_class` | | Set hold music
`strategy` | | Ring strategy
`context` | | Dialplan context used when someone "dials out" of the queue
`timeout` | 15 | Queue timeout in seconds (see [documentation](https://github.com/asterisk/asterisk/blob/master/configs/samples/queues.conf.sample))
`retry` | 5 | Queue retry time in seconds (see [documentation](https://github.com/asterisk/asterisk/blob/master/configs/samples/queues.conf.sample))
`weight` | | Relative weight of queue (see [documentation](https://github.com/asterisk/asterisk/blob/master/configs/samples/queues.conf.sample))
`maxlen` | | Maximum number of callers in queue
`announce_holdtime` | no | Periodically inform callers of estimated hold time
`announce_position` | no | Periodically inform callers of their position in the queue
`announce_frequency` | 0 | Frequency (in seconds) to play holdtime announcement
`min_announce_frequency` | 15 | Minimum time (in seconds) between holdtime announcements
`periodic_announce` | | Sounds to play periodically (comma-separated, or list)
`monitor_format` | | Audio formats used to record calls (list)
`join_empty` | yes | Allow callers to join queue when no members are available (see [documentation](https://github.com/asterisk/asterisk/blob/master/configs/samples/queues.conf.sample))
`leave_when_empty` | no | Drop callers from queue when no members are available (see [documentation](https://github.com/asterisk/asterisk/blob/master/configs/samples/queues.conf.sample))
`ring_in_use` | yes | Send calls to queue members even when their device state is "in use"
`members` | [] | List of local extensions having membership in the queue
### asterisk\_ari\_users
The `asterisk_ari_users` variable describes user accounts for the [Asterisk REST Interface](https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573).
It should contain a list of dictionaries of the following format:
Key | Default | Description
------------|---------|------------
`name` | | Username
`password` | | Password
`read_only` | yes | Limit account to read-only requests
### asterisk\_voicemail\_contexts
The `asterisk_voicemail_contexts` variable describes your [voicemail contexts](https://wiki.asterisk.org/wiki/display/AST/Configuring+Voice+Mail+Boxes).
It should contain a mapping of context names to dictionaries of the following format:
Key | Default | Description
-----------|---------|------------
`address` | | Mailbox address (usually a 3- or 4-digit number)
`password` | | Initial mailbox password (usually a numeric PIN)
`name` | | Human-readable mailbox name
`email` | | Email address for new voicemail notifications (or list)
### Exports
This role **exports** the following variables:
Variable | Description
--------------------|------------
`asterisk_data_dir` | Path to asterisk data files (voicemails, etc)
Usage
-----
Example playbook:
````yaml
- hosts: asterisk_servers
roles:
- role: asterisk
vars:
asterisk_local_nets:
- 192.168.1.0/24
asterisk_external_ip: 203.0.113.42
asterisk_public_fqdn: pbx.example.com
asterisk_password_salt: foobar
asterisk_sip_trunks:
- name: upstream-provider
host: 'sip.example.com:5060'
username: myusername
password: s3cret
vault_asterisk_sip_extensions:
- name: 7001
context: from-home
mailbox: 7000@default
cid_name: Living Room
password: s3cret
- name: 7002
context: from-home
mailbox: 7000@default
cid_name: Kitchen
password: s3cret
- name: 7003
context: from-home
mailbox: 7000@default
cid_name: Office
password: s3cret
asterisk_queues:
- name: home
strategy: ringall
retry: 1
timeout: 30
members:
- 7001
- 7002
- 7003
asterisk_ari_users:
- name: nagios
readonly: yes
password: s3cret
asterisk_voicemail_contexts:
default:
- address: 7000
password: 1234
name: Doe Family
email:
- johndoe@example.com
- janedoe@example.com
asterisk_dialplan: |
[from-upstream-provider]
; Ring all house phones for incoming PSTN calls, if no answer send to voicemail.
exten => _X.,1,Queue(home,nr,,,25)
same => n,Answer(500)
same => n,Voicemail(7000@default,su)
same => n,Hangup()
[from-home]
; local voicemail access
exten => *99,1,Answer(500)
same => n,VoiceMailMain(7000@default,s)
same => n,Hangup()
; pstn
exten => _+1NXXNXXXXXX,1,Set(CALLERID(all)=John Doe <5555555555>)
same => n,Dial(PJSIP/${EXTEN}@upstream-provider)
same => n,Hangup()
````
|
