The asterisk role installs and configures the
Asterisk PBX.
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) |
asterisk_external_ip |
External IP address of the Asterisk server (see documentation) | |
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 | Email subject line for new voicemail notifications |
asterisk_voicemail_email_body |
see default vars | 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) |
|
asterisk_password_salt |
Salt used for generating SHA-512 password hashes | |
asterisk_sip_trunks |
[] |
Upstream SIP trunks (see format below) |
asterisk_sip_extensions |
[] |
Local SIP extensions (see format below) |
asterisk_queues |
[] |
Local call queues (see format below) |
asterisk_ari_users |
[] |
User accounts for the Asterisk REST Interface (see format below) |
asterisk_voicemail_contexts |
{} |
Dictionary of voicemail contexts (see format below) |
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) |
username |
SIP account username | |
password |
SIP account password |
Note that incoming calls from a SIP trunk will be assigned the dialplan context
from-TRUNKNAME.
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 |
direct_media |
yes | Allow direct media traffic (disable for NAT) |
qualify_frequency |
30 | Interval between qualify attempts (seconds) |
qualify_frequency |
30 | Interval between qualify attempts (seconds) |
qualify_timeout |
3.0 | Qualify timeout (seconds) |
The asterisk_queues variable describes your call 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) |
retry |
5 | Queue retry time in seconds (see documentation) |
weight |
Relative weight of queue (see documentation) | |
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) |
leave_when_empty |
no | Drop callers from queue when no members are available (see documentation) |
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 |
The asterisk_ari_users variable describes user accounts for the Asterisk REST Interface.
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 |
The asterisk_voicemail_contexts variable describes your voicemail contexts.
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) |
This role exports the following variables:
| Variable | Description |
|---|---|
asterisk_archive_shell |
Shell command for generating tarball of Asterisk data |
Example playbook:
- name: configure asterisk pbx
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()