Spec Driven Development
MCP Server that facilitates spec-driven development workflows, not just Vibe Coding.
作者による説明(英語の原文)
Inspect and control Asterisk (AMI) and FreeSWITCH (ESL) PBX servers from your AI assistant.
この3つを1つのスコアにまとめることはしません。意見、測定、人気はそれぞれ別のものだからです。
このサーバーのやさしい解説はまだ書いていません。以下はレジストリから読み取った事実です。
使っているAIアプリを選んで、設定をコピーしてください。
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_MS「秘密」と表示された値は、あなたのアカウントの鍵です。絶対に人に教えたり、チャットに貼り付けたりしないでください。
必要なもの
どのアプリを使っていますか?
上でアプリを選んでください。選んだアプリを覚えておくので、次からはボタンに「…にインストール」と表示されます。
Claude Desktop → Settings → Developer → Edit Config を開き、claude_desktop_config.json にこれを貼り付けます
{
"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"
}
}
}
}ターミナルでこれを実行します
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-mcp動かす前に、空欄の値を入力してください。
プロジェクトの .cursor/mcp.json、または ~/.cursor/mcp.json に貼り付けます
{
"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"
}
}
}
}ブラウザがアプリを開いてよいか確認し、そのあとアプリが設定を表示して確認を求めます。キーの値は空欄のままなので、自分で入力してください。
プロジェクトの .vscode/mcp.json に貼り付けます
{
"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"
}
}
}
}ブラウザがアプリを開いてよいか確認し、そのあとアプリが設定を表示して確認を求めます。キーの値は空欄のままなので、自分で入力してください。
これはパソコンで動くサーバーですが、ChatGPTがつなげるのはインターネット上のサーバーだけです。ほかのアプリを選んでください。
選んだアプリは、この端末のCookieに保存されます。アカウントは不要です。
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 envこの設定は、作者のレジストリ登録情報から作成しています。READMEと違う場合は、READMEに従ってください。
パソコンで動くサーバーです。インストールすると、ツール一覧がAIアプリに表示されます。
誰でも見られる情報をチェックしており、コードは読んでいません。当てはまらない項目は数に入れていません。 · 2026年11月8日にチェック
レジストリ名は、リポジトリを所有するGitHubアカウント ictinnovations で確認済みです
このリポジトリはまだGitHubから読み取っていません
このリポジトリはまだGitHubから読み取っていません
これはパソコンで動くため、インターネット上のエンドポイントはありません
パソコンで動き、代わりにローカルの設定を使います
パソコンで動くサーバーです。ツール一覧を読み取れるのは、インターネット上のエンドポイントがある場合だけです
レジストリでこのカテゴリにある有効なすべてのサーバーと比較しています。スターは、公開者がリポジトリの所有者である場合だけ数えます。
当てはまる基本チェック1項目のうち1項目に合格しました。コードは読んでいません。キーやトークンを追加する前に、サーバーがどんなアクセスを求めているかを確認し、必要な権限だけを持つキーを使ってください。
入力が必要なもの:ASTERISK_AMI_PASSWORD, FREESWITCH_ESL_PASSWORD
あなたのパソコンで動きます。必要なときにAIアプリが起動し、データは作者のサーバーを通りません(サーバー自体がオンラインサービスを呼び出す場合を除きます)。
GitHubリポジトリの最終更新は–、最新バージョン(0.1.4)がレジストリに届いたのは12 日前です。
Claude Desktop、Claude Code、Cursor、VS CodeなどのMCP対応アプリで使えます。パソコンにNode.jsも必要です。
同じカテゴリには:Spec Driven Development, Remote Desktop Commander, Tethered Trust
同じカテゴリで、ほかの作者が作ったサーバー。スターの多い順。
カードの見方:★ GitHubスターi3つの柱:レビュー · チェック · 利用状況i更新=最新バージョンをレジストリに公開した日i
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.
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作者のリポジトリから書かれたとおりにコピーしたものです。PopNx MCPは内容を確認・推奨していません。HTML、画像、スクリプトは取り除いています。リンクはこのサイトの外へ移動します。
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.冒頭だけを表示しています。続きはリポジトリで読めます