Table of Contents

Asterisk PBX

Asterisk is a free, open-source framework for building communications applications — most commonly a PBX (Private Branch Exchange) for Voice over IP (VoIP) telephony. Maintained by Sangoma, it turns a server into a phone system that can register SIP phones, connect to telephone-service providers, route calls, run voicemail and IVR menus, bridge conferences, and much more.

This guide focuses on running Asterisk in containers with Docker, covering deployment, PJSIP configuration, the dialplan, SIP endpoints and provider trunks, security (toll-fraud prevention, TLS/SRTP), monitoring, and production best practices.

Important

Telephony is a prime target for toll fraud — attackers who compromise a PBX can place expensive calls at your expense. Never expose Asterisk to the internet without the hardening in Security. Treat a public-facing PBX as a high-value asset.

Versions and Editions

Track Notes
Asterisk 20 (LTS) Long-Term Support release — the recommended default for new production systems; security fixes through ~2027.
Asterisk 22 (Standard) Newest standard release with the latest features; shorter support window.
Certified Asterisk Commercially supported, extra-stability builds from Sangoma.
FreePBX / distros Web GUI (FreePBX) layered on top of Asterisk for admins who prefer not to edit config files by hand.
Note

Use the modern PJSIP channel driver (chan_pjsip) throughout this guide. The legacy chan_sip driver is deprecated and was removed in Asterisk 21, so any current configuration should be PJSIP-based.

Core Concepts

Concept Meaning
Channel A single call leg — a connection between Asterisk and an endpoint (a phone, a trunk). Identified like PJSIP/6001-00000001.
Endpoint A configured SIP entity (a desk phone, softphone, or trunk) defined in pjsip.conf.
Dialplan The call-routing logic in extensions.conf — what Asterisk does when a call enters a context.
Context A named grouping of dialplan rules that provides isolation between, for example, internal users and inbound provider calls.
Trunk A connection to a telephone-service provider (ITSP) that carries calls to/from the PSTN.
AOR "Address of Record" — where a PJSIP endpoint can be reached (its registered contacts).
AMI / ARI The Asterisk Manager Interface (TCP control/events) and Asterisk REST Interface (HTTP/WebSocket API) for integrations.

Architecture

flowchart LR
    Phone[SIP Phones / Softphones] -- SIP + RTP --> AST
    subgraph AST[Asterisk PBX]
      PJSIP[chan_pjsip] --> Dialplan[Dialplan<br/>extensions.conf]
      Dialplan --> Apps[Applications<br/>Dial, Voicemail, Queue, IVR]
    end
    AST -- SIP trunk --> ITSP[Telephony Provider / PSTN]
    AST -. AMI/ARI .-> Apps2[Integrations / CRM]

Signaling uses SIP (default UDP/TCP 5060, TLS 5061); the actual audio flows separately as RTP over a UDP port range (default 10000–20000). Both must be reachable, which makes networking — especially NAT — the trickiest part of a container deployment (see Installation and Best Practices).

In This Section

  • Installation and Deployment — running Asterisk with Docker, config volumes, and the SIP/RTP networking model
  • Configuration — core config files and PJSIP transports, endpoints, auth, and AORs
  • Dialplan — call routing in extensions.conf — contexts, extensions, applications, and patterns
  • Endpoints and Trunks — registering SIP phones and connecting provider trunks with inbound/outbound routing
  • Queues and IVR — building auto-attendant menus and ACD call queues with agent management
  • Security — toll-fraud prevention, TLS/SRTP, ACLs, fail2ban, and securing AMI/ARI
  • Monitoring and Troubleshooting — the Asterisk CLI, logging, CDR/CEL, and diagnosing SIP/RTP issues
  • Best Practices — NAT and RTP handling, codecs, high availability, and backups

Resources

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