Spec Driven Development
MCP Server that facilitates spec-driven development workflows, not just Vibe Coding.
作者提供的说明(英文原文)
Inspect and control Asterisk (AMI) and FreeSWITCH (ESL) PBX servers from your AI assistant.
这三项永远不会合并成一个分数:意见、测量和人气是不同的东西。
我们还没有为它写通俗的介绍。下面是我们从注册中心读取到的事实。
选择你使用的 AI 应用,然后复制设置。
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标为机密的值是你账户的钥匙。千万不要分享给别人,也不要粘贴到聊天里。
你需要
你用的是哪个应用?
在上方选择你的应用。我们会记住它,下次按钮会显示“安装到 …”。
打开 Claude Desktop → Settings → Developer → Edit Config,把下面的内容粘贴到 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"
}
}
}
}在终端中运行这条命令
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运行前请先填写空着的值。
粘贴到项目中的 .cursor/mcp.json,或 ~/.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"
}
}
}
}浏览器会先询问是否打开应用,然后应用会显示设置并请你确认。密钥的值会留空,由你自己填写。
粘贴到项目中的 .vscode/mcp.json
{
"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"
}
}
}
}浏览器会先询问是否打开应用,然后应用会显示设置并请你确认。密钥的值会留空,由你自己填写。
这个服务器在你的电脑上运行,而 ChatGPT 只能连接互联网上的服务器。请选择其他应用。
你的选择通过 Cookie 保存在这台设备上,无需账户。
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这些设置是根据作者在注册中心的条目生成的。如果 README 的写法不同,请以 README 为准。
它在你的电脑上运行;安装后,工具列表会显示在你的 AI 应用中。
我们检查的是任何人都能看到的信息;我们没有阅读代码。不适用的检查项不计入总数。 · 检查于 2026年11月8日
注册名称已通过 GitHub 账户 ictinnovations 验证,该账户拥有此仓库
我们还没有从 GitHub 读取这个仓库
我们还没有从 GitHub 读取这个仓库
它在你的电脑上运行,没有互联网端点
在你的电脑上运行,改用本地设置
它在你的电脑上运行;我们只能读取互联网端点的工具列表
与注册中心里这个分类中所有活跃的服务器比较。只有当发布者拥有该仓库时,星标数才计入。
它在 1 项适用的基础检查中通过了 1 项。我们没有阅读代码。在添加密钥或令牌之前,请先看看这个服务器要求哪些权限,并使用只具备所需权限的密钥。
你需要填写:ASTERISK_AMI_PASSWORD, FREESWITCH_ESL_PASSWORD
它在你的电脑上运行。你的 AI 应用会在需要时启动它,数据不会经过作者的服务器(除非它自己调用在线服务)。
GitHub 仓库最后更新于–,最新版本(0.1.4)于13天前进入注册中心。
支持 MCP 的应用,例如 Claude Desktop、Claude Code、Cursor 和 VS Code。你的电脑上还需要有 Node.js。
同一分类中有:Spec Driven Development, Remote Desktop Commander, Cape Winelands Visit Planner
同一分类中其他作者的服务器,按星标数排序。
如何看卡片:↓ 每周下载量i★ GitHub 星标i三大支柱:评价 · 检查 · 使用情况i更新 = 最新版本发布到注册中心的时间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.
Cape Winelands wine farm finder and day-route planner
按原文从作者的仓库复制。PopNx MCP 没有审查或背书其内容。HTML、图片和脚本已被移除。链接会离开本站。
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.只显示了开头部分;其余内容请在仓库中阅读