Asterisk

Open source telephony toolkit and IP PBX with dialplan, queues and PSTN interconnect.

Provider: default
Kind: Contact center
Category: Messaging
Official docs: https://docs.asterisk.org/Getting-Started/Beginning-Asterisk/

Properties · 6 set by this component

property	type	values	required	kind default	this component
callRecording	boolean	—	no	false	▸ false
channelChat	boolean	—	no	false	▸ false
channelDriver · component property	enum	pjsip \| iax2 \| dahdi \| websocket	no	pjsip	inherited
channelVoice	boolean	—	no	true	▸ true
emergencyCalling	boolean	—	no	false	inherited
managed	boolean	—	no	false	▸ false
maxConcurrentContacts	integer	—	no	—	inherited
outboundDialing	boolean	—	no	false	▸ false
pstnInterconnect · component property	boolean	—	no	false	inherited
queueStrategy · component property	enum	ringall \| leastrecent \| fewestcalls \| random \| rrmemory \| linear \| wrandom	no	ringall	inherited
recordingRetentionDays	integer	—	no	—	inherited
routing	enum	fifo \| skills-based \| attribute-based	no	fifo	▸ fifo

Rules that apply · 0

Every rule whose selector covers this component's kind or its category. Whether a rule fires depends on the design: its properties, its edges and its groups.

No rule reads this kind yet. That is a gap, not a guarantee.

Use it in a design

nodes:
  - { id: "asterisk", component: "asterisk" }

Other Contact center products · 13

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Source

1# yaml-language-server: $schema=../../schema/component.schema.json
2# Asterisk: self-hosted open source telephony toolkit and IP PBX written in C, speaking SIP and
3# connecting to PSTN/POTS as well as VoIP phones ("Beginning Asterisk"). Callers wait in app_queue
4# in arrival order and the member-selection strategy defaults to ringall, which rings every
5# available member until one answers (queues.conf sample). Recording is not automatic: a dialplan
6# has to invoke MixMonitor, so nothing is recorded out of the box.
7schemaVersion: "1"
8type: "component"
9id: "asterisk"
10kind: "contact-center"
11provider: "default"
12docs: "https://docs.asterisk.org/Getting-Started/Beginning-Asterisk/"
13defaults:
14  managed: false
15  channelVoice: true
16  channelChat: false
17  routing: "fifo"
18  callRecording: false
19  outboundDialing: false
20properties:
21  queueStrategy:
22    type: "enum"
23    values: ["ringall", "leastrecent", "fewestcalls", "random", "rrmemory", "linear", "wrandom"]
24    default: "ringall"
25  channelDriver:
26    type: "enum"
27    values: ["pjsip", "iax2", "dahdi", "websocket"]
28    default: "pjsip"
29  pstnInterconnect:
30    type: "boolean"
31    default: false

The same page as markdown, for an agent: /catalogue/components/asterisk.md