Spec Driven Development
MCP Server that facilitates spec-driven development workflows, not just Vibe Coding.
Description from the maker (original English)
Inspect and control Asterisk (AMI) and FreeSWITCH (ESL) PBX servers from your AI assistant.
These three are never merged into one score: opinions, measurements and popularity are different things.
We have not written a plain-language summary for this one yet. Below are the facts we read from the registry.
Pick the AI app you use, then copy the settings.
ASTERISK_AMI_HOST, ASTERISK_AMI_PORT, ASTERISK_AMI_USERNAME, ASTERISK_AMI_PASSWORD, ASTERISK_AMI_TLS, FREESWITCH_ESL_HOST, FREESWITCH_ESL_PORT, FREESWITCH_ESL_PASSWORD, PBX_MCP_ALLOW_WRITE, PBX_MCP_TIMEOUT_MSValues marked secret are keys to your account. Never share them or paste them into a chat.
You need
Which app do you use?
Pick your app above. We remember it, so next time the button says “Install in …”.
Open Claude Desktop → Settings → Developer → Edit Config, and paste this into claude_desktop_config.json
{
"mcpServers": {
"pbx": {
"command": "npx",
"args": [
"-y",
"pbx-mcp"
],
"env": {
"ASTERISK_AMI_HOST": "your-value",
"ASTERISK_AMI_PORT": "your-value",
"ASTERISK_AMI_USERNAME": "your-value",
"ASTERISK_AMI_PASSWORD": "your-value",
"ASTERISK_AMI_TLS": "your-value",
"FREESWITCH_ESL_HOST": "your-value",
"FREESWITCH_ESL_PORT": "your-value",
"FREESWITCH_ESL_PASSWORD": "your-value",
"PBX_MCP_ALLOW_WRITE": "your-value",
"PBX_MCP_TIMEOUT_MS": "your-value"
}
}
}
}Run this in your terminal
claude mcp add pbx --env ASTERISK_AMI_HOST=your-value --env ASTERISK_AMI_PORT=your-value --env ASTERISK_AMI_USERNAME=your-value --env ASTERISK_AMI_PASSWORD=your-value --env ASTERISK_AMI_TLS=your-value --env FREESWITCH_ESL_HOST=your-value --env FREESWITCH_ESL_PORT=your-value --env FREESWITCH_ESL_PASSWORD=your-value --env PBX_MCP_ALLOW_WRITE=your-value --env PBX_MCP_TIMEOUT_MS=your-value -- npx -y pbx-mcpFill in the empty values before running it.
Paste into .cursor/mcp.json in your project, or ~/.cursor/mcp.json
{
"mcpServers": {
"pbx": {
"command": "npx",
"args": [
"-y",
"pbx-mcp"
],
"env": {
"ASTERISK_AMI_HOST": "your-value",
"ASTERISK_AMI_PORT": "your-value",
"ASTERISK_AMI_USERNAME": "your-value",
"ASTERISK_AMI_PASSWORD": "your-value",
"ASTERISK_AMI_TLS": "your-value",
"FREESWITCH_ESL_HOST": "your-value",
"FREESWITCH_ESL_PORT": "your-value",
"FREESWITCH_ESL_PASSWORD": "your-value",
"PBX_MCP_ALLOW_WRITE": "your-value",
"PBX_MCP_TIMEOUT_MS": "your-value"
}
}
}
}Your browser asks to open the app, then the app shows the settings and asks you to confirm. Key values are left empty for you to fill in.
Paste into .vscode/mcp.json in your project
{
"servers": {
"pbx": {
"type": "stdio",
"command": "npx",
"args": [
"-y",
"pbx-mcp"
],
"env": {
"ASTERISK_AMI_HOST": "your-value",
"ASTERISK_AMI_PORT": "your-value",
"ASTERISK_AMI_USERNAME": "your-value",
"ASTERISK_AMI_PASSWORD": "your-value",
"ASTERISK_AMI_TLS": "your-value",
"FREESWITCH_ESL_HOST": "your-value",
"FREESWITCH_ESL_PORT": "your-value",
"FREESWITCH_ESL_PASSWORD": "your-value",
"PBX_MCP_ALLOW_WRITE": "your-value",
"PBX_MCP_TIMEOUT_MS": "your-value"
}
}
}
}Your browser asks to open the app, then the app shows the settings and asks you to confirm. Key values are left empty for you to fill in.
This one runs on your computer, and ChatGPT can only connect to servers on the internet. Pick another app.
Your choice stays on this device in a cookie; no account needed.
ASTERISK_AMI_HOST envASTERISK_AMI_PORT envASTERISK_AMI_USERNAME envASTERISK_AMI_TLS envFREESWITCH_ESL_HOST envFREESWITCH_ESL_PORT envPBX_MCP_ALLOW_WRITE envPBX_MCP_TIMEOUT_MS envThese settings are generated from the maker’s registry entry. If the README says otherwise, follow the README.
Runs on your computer; the tool list appears in your AI app after you install it.
We check information anyone can see; we have not read the code. Checks that do not apply are left out of the count. · Checked Nov 8, 2026
The registry name was verified with the GitHub account ictinnovations, which owns the repo
We have not read this repo from GitHub yet
We have not read this repo from GitHub yet
This one runs on your computer; there is no internet endpoint
Runs on your computer and uses local settings instead
Runs on your computer; we can only read tool lists from internet endpoints
Compared with every active server in this category in the registry. Stars count only when the publisher owns the repo.
It passed 1 of 1 applicable basic checks. We have not read the code. Before adding a key or token, see what access the server asks for and use a key with only the permissions you need.
You need to fill in: ASTERISK_AMI_PASSWORD, FREESWITCH_ESL_PASSWORD
It runs on your computer. Your AI app starts it when needed, and data does not pass through the maker’s servers (unless it calls an online service itself).
The GitHub repo last changed –, and the latest version (0.1.4) reached the registry 13 days ago.
Apps that support MCP, such as Claude Desktop, Claude Code, Cursor and VS Code. You also need Node.js on your computer.
In the same category: Spec Driven Development, Remote Desktop Commander, Thth
Servers in the same category from other makers, sorted by stars.
Reading a card:↓ Downloads per weeki★ GitHub starsiThree pillars: reviews · checks · usageiUpdated = latest version published to the registryi
MCP Server that facilitates spec-driven development workflows, not just Vibe Coding.
Remote Desktop Commander: use files, terminal and processes on your own computers from any AI app.
Call before you post: ships a draft to Threads/Bluesky/Mastodon only after a person approved it.
Copied from the maker’s repo as written. PopNx MCP has not reviewed or endorsed it. HTML, images and scripts are removed. Links lead off this site.
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 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.
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.
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.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.
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 |
| 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) |
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.
| 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 |
| 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 |
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.
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}$.
pjsip.conf (a missing include is an error), and let Asterisk write to it: touch /etc/asterisk/pjsip_mcp.conf
pjsip.conf. Nothing else in pjsip.conf is ever touched: #include pjsip_mcp.conf
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
transport-udp, transport-tcp or transport-tls; define the ones you will use in pjsip.conf.Only the beginning is shown; read the rest in the repo