pbx-mcp

pbx-mcp

An MCP server that lets an AI assistant inspect and control Asterisk and FreeSWITCH.

Ask "which extensions are offline right now?" or "why is my SIP trunk not registering?" and get a real answer from the live switch, not a guess.

Developed by Tahir Almas at ICT Innovations, the team behind ICTCore, ICTContact, ICTDialer, ICTFax and ICTPBX. The AMI and ESL clients in this repo are the same protocol groundwork those products run on.

Why

Debugging a PBX means memorising two very different command sets. Asterisk speaks AMI and a CLI with hundreds of verbs. FreeSWITCH speaks ESL with its own vocabulary. If you run both, you're context switching all day.

pbx-mcp puts a single, well described tool surface in front of both, so your assistant can go from "calls are failing" to sofia status gateway without you spelling out each step.

New here? The user guide walks through PBX setup, client config, worked examples and troubleshooting. This README is the quick reference.

What your assistant sees

Twelve read-only tools, six per engine, each one described well enough that the model picks the right one without being told. This is the server running under the MCP Inspector:

The four write tools, asterisk_originate, asterisk_hangup, freeswitch_originate and freeswitch_hangup, stay hidden until you set PBX_MCP_ALLOW_WRITE=true.

Install

npm install -g pbx-mcp

Or run it straight from npx, which is what most MCP client configs do:

npx -y pbx-mcp

You need Node 18 or newer.

Docker

There's a prebuilt image if you'd rather not put Node on the machine that talks to your PBX.

docker run -i --rm \
  -e ASTERISK_AMI_HOST=10.0.0.10 \
  -e ASTERISK_AMI_USERNAME=mcp \
  -e ASTERISK_AMI_PASSWORD=change-me \
  ghcr.io/ictinnovations/pbx-mcp

The same image is on Docker Hub as ictinnovations/pbx-mcp if that registry is an easier pull for you.

Three things to know:

• -i is not optional. The server speaks MCP over stdio, so without stdin attached the container starts and then sits there saying nothing, which looks exactly like a broken server.
• There is no port to publish. Nothing listens.
• Your PBX has to be reachable from inside the container. If Asterisk runs on the Docker host itself, swap the IP above for host.docker.internal on Mac and Windows, or add --network host on Linux.

The image runs as a non-root user and, like every other way of running this, starts read only.

Configure

Everything comes from environment variables. Set the Asterisk block, the FreeSWITCH block, or both. The server only registers tools for what you've actually configured, so an Asterisk-only shop never sees a FreeSWITCH tool.

Asterisk

Variable	Default	Notes
ASTERISK_AMI_HOST	(required to enable)	Hostname or IP of the Asterisk box
ASTERISK_AMI_PORT	5038	AMI port from manager.conf
ASTERISK_AMI_USERNAME		AMI user
ASTERISK_AMI_PASSWORD		AMI secret
ASTERISK_AMI_TLS	false	Set true if tlsenable=yes

Your manager.conf user needs at least read = system,call,command and write = command. Add originate only if you plan to turn on write mode.

[mcp]
secret = change-me
read = system,call,command
write = command

FreeSWITCH

Variable	Default	Notes
FREESWITCH_ESL_HOST	(required to enable)	Hostname or IP of the switch
FREESWITCH_ESL_PORT	8021	Inbound ESL port
FREESWITCH_ESL_PASSWORD	ClueCon	From event_socket.conf.xml

Shared

Variable	Default	Notes
PBX_MCP_ALLOW_WRITE	false	Unlocks call control. Read the safety section first
PBX_MCP_TIMEOUT_MS	10000	Per command timeout

Provisioning (Asterisk only, opt-in)

Variable	Default	Notes
PBX_MCP_ALLOW_PROVISION	false	Registers the six trunk and extension tools. Independent of PBX_MCP_ALLOW_WRITE
PBX_MCP_TRUNK_ALLOW	(unset)	Comma-separated IPv4 CIDRs and hostnames a trunk may point at. Unset means asterisk_trunk_create is refused
PBX_MCP_CONTEXT_ALLOW	(unset)	Comma-separated dialplan contexts new objects may use. Unset means every create is refused
PBX_MCP_PJSIP_FILE	pjsip_mcp.conf	The one include file provisioning writes to. A bare pjsip_*.conf name (never pjsip.conf, manager.conf or any other file)

Claude Desktop

Add this to claude_desktop_config.json:

{
  "mcpServers": {
    "pbx": {
      "command": "npx",
      "args": ["-y", "pbx-mcp"],
      "env": {
        "ASTERISK_AMI_HOST": "10.0.0.10",
        "ASTERISK_AMI_USERNAME": "mcp",
        "ASTERISK_AMI_PASSWORD": "change-me",
        "FREESWITCH_ESL_HOST": "10.0.0.11",
        "FREESWITCH_ESL_PASSWORD": "ClueCon"
      }
    }
  }
}

The same shape works for any MCP client that speaks stdio. There's a copy in examples/claude_desktop_config.json.

To run the container instead of npx, keep the env block and point command at Docker:

{
  "mcpServers": {
    "pbx": {
      "command": "docker",
      "args": [
        "run", "-i", "--rm",
        "-e", "ASTERISK_AMI_HOST",
        "-e", "ASTERISK_AMI_USERNAME",
        "-e", "ASTERISK_AMI_PASSWORD",
        "ghcr.io/ictinnovations/pbx-mcp"
      ],
      "env": {
        "ASTERISK_AMI_HOST": "10.0.0.10",
        "ASTERISK_AMI_USERNAME": "mcp",
        "ASTERISK_AMI_PASSWORD": "change-me"
      }
    }
  }
}

Each -e NAME with no value forwards that variable from env into the container, which keeps the secrets out of the argument list.

Tools

Asterisk

Tool	What it does
asterisk_status	Version, uptime, active calls and calls processed
asterisk_channels	Every live channel with caller ID, state, bridge, duration and dialplan position. Optional substring filter
asterisk_endpoints	PJSIP endpoints with device state and contact count. Falls back to chan_sip peers on older installs
asterisk_dialplan	Dumps a context, or one extension inside a context
asterisk_cli	Any CLI command, subject to the safety policy below
asterisk_hangup_preview	Shows which live channels a hangup would drop, without dropping them
asterisk_originate	Places a call. Write mode only
asterisk_hangup	Kills a channel by name. Write mode only

FreeSWITCH

Tool	What it does
freeswitch_status	Version, uptime, current and maximum sessions
freeswitch_channels	Every live call leg from show channels. Optional substring filter that keeps the header row
freeswitch_registrations	Registered users on a Sofia profile, with contact URI, user agent and expiry
freeswitch_sofia_status	Every SIP profile and gateway, including whether trunks are registered upstream
freeswitch_api	Any API command, subject to the safety policy below
freeswitch_hangup_preview	Shows which live legs a hangup would drop, without dropping them
freeswitch_originate	Places a call. Write mode only
freeswitch_hangup	uuid_kill on a channel UUID. Write mode only

Safety

A PBX is not a scratch pad. Reloading a profile drops registrations, and an originate spends real money on a live trunk. So the default posture is read-only and the guards are layered:

Read-only by default. asterisk_cli accepts an allow list of inspection prefixes (core show, pjsip show, dialplan show, queue show and friends). freeswitch_api accepts the same kind of list (status, show, sofia status, db list and friends), matched on the start of the command so the subcommand counts.

The FreeSWITCH list allows subcommands, it does not deny scary words. sofia status reads, sofia profile internal restart isn't on the list, so it's refused. This used to work the other way around, scanning each word against a list of state changing verbs, and that only ever catches the words somebody thought of. conference 3001 kick all walked straight through it. Reported by Electrical-Place-458 on r/mcp.

Write tools aren't registered at all in read-only mode. asterisk_originate and the other three never appear in tools/list unless you set PBX_MCP_ALLOW_WRITE=true. A model can't call a tool it can't see.

The hangup preview tools are always available. asterisk_hangup_preview and freeswitch_hangup_preview show which live channels a hangup would drop without touching them, so you can see the blast radius before setting the write flag, or catch a typo in a channel name before running the real thing. They're read-only by contract but exercise the same matching the write path uses.

Shell metacharacters are rejected on both transports before a command is sent.

AMI header injection is blocked. Every field that lands in an AMI action is checked for carriage returns and newlines, so a caller ID string can't smuggle in an extra header.

Output is clamped to 20,000 characters. One show channels on a busy switch won't flood the context window.

Even with all that, give the AMI user the narrowest permission set that answers your questions, and put the PBX behind a firewall rather than on the public internet.

Provisioning trunks and extensions (Asterisk, opt-in)

Six more tools create, list and delete PJSIP trunks and SIP extensions through AMI UpdateConfig. They are registered only when PBX_MCP_ALLOW_PROVISION=true, and they are separate from write mode: enabling one does not enable the other.

Tool	What it does
asterisk_trunk_create	IP-authenticated trunk: endpoint, aor and identify. Fields: name, host, port (5060), transport (udp, tcp, tls), codecs, context, dry_run
asterisk_trunk_list / asterisk_trunk_delete	List or remove trunks this server created
asterisk_extension_create	A SIP endpoint phones register to (endpoint, aor, auth), not a dialplan extension. Fields: number, context, codecs, dry_run. The generated password is returned once
asterisk_extension_list / asterisk_extension_delete	List or remove extensions this server created. Passwords are never listed

Codecs are ulaw, alaw, g722, g729, opus. Names and numbers match ^[A-Za-z0-9_-]{1,32}$.

One-time setup

1. Create the managed file, empty, beside pjsip.conf (a missing include is an error), and let Asterisk write to it:

   touch /etc/asterisk/pjsip_mcp.conf
   

2. Add this single line at the end of pjsip.conf. Nothing else in pjsip.conf is ever touched:

   #include pjsip_mcp.conf
   

3. Give the AMI user the extra permissions provisioning needs. config is for UpdateConfig/GetConfig, command for the reload and verify steps, originate only if you also use write mode:

   [mcp]
   secret = change-me
   deny = 0.0.0.0/0.0.0.0
   permit = 192.0.2.0/24
   read = system,call,command
   write = command,originate,config
   

4. Trunks reference a transport named transport-udp, transport-tcp or transport-tls; define the ones you will use in pjsip.conf.

How it works

• Everything goes into the managed file with UpdateConfig (NewCat, Append, RenameCat, DelCat), then module reload res_pjsip.so, then pjsip show endpoint mcp-<name>. If Asterisk does not report the endpoint, the change is removed again and the call returns an error.
• Every object is named mcp-<name>. List and delete only act on mcp- sections of the managed file, and a trunk cannot be deleted as an extension or the other way round.
• Create refuses a name that already exists. dry_run=true returns the exact config block and makes no AMI calls; for an extension the password is shown as a placeholder.
• All provisioning calls are serialized in-process, because UpdateConfig is read-modify-write. Run only one pbx-mcp provisioner per managed file.
• An extension registers as mcp-<number>, not the bare number. PJSIP matches the endpoint and its AOR by the registering username, and every object has to carry the mcp- prefix. The create response states the username.
• The password is 24 random URL-safe characters from crypto.randomBytes. It is returned only in the create response, which is the only place it is ever shown or logged; it does exist in the managed file on the PBX, as any SIP password must.

Toll fraud: read this before enabling

A SIP trunk lets calls leave your PBX, and a registrable extension with a context that reaches an outbound route lets anyone who learns the password do the same. Stolen SIP credentials are a leading cause of large telephony bills.

• Point PBX_MCP_CONTEXT_ALLOW at contexts that cannot dial out to the PSTN unless you really mean it. The context is the only thing that decides what a registered phone or trunk may call.
• Keep PBX_MCP_TRUNK_ALLOW as narrow as you can. Hostnames are compared literally and never resolved, so DNS cannot widen the list. IPv6 is not supported.
• An empty allowlist refuses everything: provisioning fails closed.
• Treat the assistant that holds these tools as holding the ability to change who can reach your PBX. Review what it creates (*_list, or dry_run first), and do not expose SIP (5060) or AMI (5038) to the internet.
• Prefer a lab or staging PBX when trying this out.

Example flow

Using documentation addresses (RFC 5737) only. Environment:

PBX_MCP_ALLOW_PROVISION=true
PBX_MCP_TRUNK_ALLOW=192.0.2.0/24,198.51.100.0/24
PBX_MCP_CONTEXT_ALLOW=from-trunk,from-internal

1. Preview a trunk with asterisk_trunk_create and dry_run=true:

   { "name": "carrier1", "host": "192.0.2.10", "port": 5060, "transport": "udp",
     "codecs": ["ulaw", "alaw"], "context": "from-trunk", "dry_run": true }
   

   which returns

   [mcp-carrier1]
   type=endpoint
   transport=transport-udp
   context=from-trunk
   disallow=all
   allow=ulaw,alaw
   aors=mcp-carrier1
   
   [mcp-carrier1]
   type=aor
   contact=sip:192.0.2.10:5060
   
   [mcp-carrier1]
   type=identify
   endpoint=mcp-carrier1
   match=192.0.2.10
   

2. Run it again without dry_run. A host outside the allowlist, for example 203.0.113.5, is refused.

3. Create an extension: { "number": "1001", "context": "from-internal" }. The reply gives the SIP username mcp-1001 and the password, once.

4. Point a phone at the PBX with those credentials, then check it with asterisk_endpoints.

5. asterisk_extension_list and asterisk_trunk_list show what exists; asterisk_extension_delete with 1001 and asterisk_trunk_delete with carrier1 remove it again.

Tested against Asterisk 22 (certified 22.8). The PJSIP options used (identify, max_contacts, remove_existing, auth_type=userpass) and the UpdateConfig features (RenameCat, catfilter) are standard, but re-check them on other Asterisk versions. TCP and TLS trunks are generated but were not exercised against a live peer.

Build from source

git clone https://github.com/ictinnovations/pbx-mcp.git
cd pbx-mcp
npm install
npm run build
npm start

The AMI and ESL clients have no third party dependencies. Both protocols are just framed text over TCP, and hand rolling them keeps the install small and the behaviour predictable. The only runtime dependencies are the MCP SDK and Zod.

The protocol clients, on their own

If you want to talk to a PBX from your own Node code and don't need MCP at all, the two clients underneath this server are published separately. Same protocol work, no MCP SDK, no Zod, nothing:

• asterisk-ami-node - Asterisk Manager Interface client. npm install asterisk-ami-node
• freeswitch-esl-node - FreeSWITCH Event Socket client, inbound mode. npm install freeswitch-esl-node

Both are zero dependency, TypeScript, ESM and CommonJS, Node 18 or newer, and tested against mock switches so you can run the suite without a PBX.

Layout

src/
  index.ts          entry point, config to transport wiring
  ami.ts            Asterisk Manager Interface client
  esl.ts            FreeSWITCH Event Socket Layer client
  config.ts         environment config and the command safety policy
  tools/
    asterisk.ts     Asterisk tool definitions
    freeswitch.ts   FreeSWITCH tool definitions
    format.ts       text table and truncation helpers

Contributing

Issues and pull requests are welcome. If you're adding a tool, describe it the way you'd describe it to a colleague who has never seen your dialplan. The model picks tools from those descriptions, so a vague one is a broken one.

About

Built and maintained by Tahir Almas, founder of ICT Innovations.

ICT Innovations has shipped open source and commercial telephony since 2005. If pbx-mcp is useful to you, the wider stack behind it might be too:

• ICTCore - open source telephony framework, the base for the products below
• ICTPBX - white label multi tenant IP PBX, with a free community edition on GitHub
• ICTContact - contact center and unified communications
• ICTDialer - auto and predictive dialer
• ICTFax - open source fax server
• asterisk-ami-node and freeswitch-esl-node - the protocol clients from this repo, published on their own

Questions about the commercial products go through the ICT Innovations support portal. Questions about pbx-mcp itself belong in GitHub issues, where everyone can read the answer.

License

MIT. See LICENSE.