MCP server for Asterisk and FreeSWITCH. Lets AI assistants inspect live PBX state.
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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.
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.
Twelve read-only tools, six per engine, each 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.
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.
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.
| 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
| 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 |
| Variable | Default | Notes |
|---|---|---|
PBX_MCP_ALLOW_WRITE | false | Unlocks call control. Read the safety section first |
PBX_MCP_TIMEOUT_MS | 10000 | Per command timeout |
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.
| 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_originate | Places a call. Write mode only |
asterisk_hangup | Kills a channel by name. Write mode only |
| 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_originate | Places a call. Write mode only |
freeswitch_hangup | uuid_kill on a channel UUID. Write mode only |
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 an allow list of inspection verbs (status, show, sofia, version and friends).
Word level scanning on FreeSWITCH. sofia is allow listed, but sofia profile internal restart is not a read. Every word in the command is checked against a list of state changing verbs, so the destructive tail gets caught even when the leading verb looks harmless.
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.
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.
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.
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:
npm install asterisk-ami-nodenpm install freeswitch-esl-nodeBoth 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.
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
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.
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:
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.
MIT. See LICENSE.
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about 1 month ago
docker pull ictinnovations/pbx-mcp