Patch MCP

com.patchmcp由 patchmcp.com 发布官方发布者v0.2.0更新于 Oct 24, 2026

AI phone calls from your verified number, with answers, transcripts and recordings.

已验证Streamable HTTP可网页运行效率与工作流沟通与协作

概览

AI 生成的概览

通过语音代理用你本人已验证的号码拨打真实电话,并返回通话状态、文字记录与结构化结果。

可以做什么
  • 拨号前估算线路可用性、每分钟用量与剩余额度
  • 用已验证号码按指定目标发起外呼,由实时语音代理完成通话
  • 查询通话状态与结构化结果,可选择短暂等待状态变化
  • 列出近期通话及状态与结果,并翻页查看更早记录
  • 提前结束进行中的通话,以及把通话设为公开或私密
  • 读取账户绑定的本机号码(E.164 格式)及其来电显示验证状态
适用场景
适合需要助手真正替用户打电话并回报通话内容的场景,例如订位、咨询问题、催查订单。也适合回顾历史通话或中止不再需要的通话。
运行要求
远程 streamable HTTP 端点,无需安装包或本地运行时。需要已认证的账户,并绑定一个已验证、用作主叫显示的号码。涉及任何非美国/加拿大号码的通话需要付费套餐,用量按额度计费。
安装前请注意
会拨打真实且计费的电话并消耗计量额度,建议先做估算。通话默认私密,但一旦设为公开,其目标、电话号码、文字记录、结果与录音(包括进行中的实时音频)将对持有收听链接的任何人可见,仅在用户明确要求时公开。提前结束通话仍会保留部分文字记录与结果。

安装

在 SourceWeft 中

  1. 打开 控制台中的 Patch MCP,将其添加到工作区。
  2. 为需要使用其工具的对话启用该服务。

Web executable,通过 Streamable HTTP。 远程服务在工作区中配置后即可从网页运行时运行。

交给智能体安装

把这段提示词粘贴到 Claude Code、Codex、Cursor 等智能体,或 SourceWeft 对话中。智能体会读取该服务器的安装指南,先向你展示发布者、端点和登录方式,经你同意后再把它添加到你的客户端。密钥不会写进任何命令。

阅读 https://sourceweft.com/mcp/com.patchmcp%2Fpatch/install.md,并按其中说明把这个 MCP 服务器接入我的客户端。修改任何配置之前,先告诉我它的发布者、端点和登录方式,等我确认。不要把 API 密钥写进命令或我会提交的文件;问我密钥应从哪里获取。

查看智能体所依据的安装指南

其他 MCP 客户端

把它添加到你客户端的 mcpServers 配置中。

{
  "mcpServers": {
    "patch": {
      "type": "http",
      "url": "https://www.patchmcp.com/mcp"
    }
  }
}

工具

7

estimate_call

Estimate call

Check route availability, estimated one-minute usage, and remaining allowance before dialing. Calls in Central America, Europe, North America, and South America. Calls involving any non-US/Canadian number require a paid plan. Uses the verified caller ID and destination to price the route. Uses cached routing data or dialed prefixes without a paid lookup. Does not place a call, send a code, or reserve plan usage. Estimates expire and are rechecked before dialing.

输入参数to必填

place_call

Place call

Place the call FROM the authenticated user's own verified phone number to accomplish a specific objective (book a table, ask a question, chase an order). A real-time voice agent handles the call, including automated phone menus and hold queues. Its default opening says who it is calling for and states the call's purpose. Users can customize the introduction and conversation style under Account → Call agent instructions; saved instructions apply to new calls, with the objective, context and other call details added by the server. The first spoken turn to a human must still include the recording disclosure in the resolved call language (English: "This call is being recorded.") after the introduction and before the first business question or request. Returns immediately with a call_id and resolved locale, language, and voice; the call proceeds in the background and typically takes 2–5 minutes — poll get_call (pass wait_seconds to long-poll) until the status is terminal. The response includes a listen_url: a private web link where the user can hear the call live, watch the transcript, and stop the call — offer it to the user. Calls in Central America, Europe, North America, and South America, subject to route availability; never for emergencies. Calls involving a non-US/Canadian number require a paid plan from the first call. Rates vary by caller ID and destination; use estimate_call to check available minutes. Paid usage follows telephone, AI and any lookup costs, including unsuccessful attempts, and can update after the call. Calls can be placed at any time of day, any day of the week, and the user is responsible for fair use; one call runs at a time per account and volume is governed by the plan — rejections return a machine-readable `error` code with retry guidance, so relay the message to the user rather than retrying blindly. The voice agent receives the name of the person the call is for: if the account has no name on file the call is rejected with `user_name_req

输入参数to必填localecontextobjective必填user_namecallee_name

get_call

Get call

Get the status and, once terminal, the structured resolution of a call started with place_call. Returns instantly by default. Pass wait_seconds (max 50) to have the server hold the request and return early on any status change — fewer round trips while a call is in progress. Terminal statuses: completed, no_answer, busy, voicemail, failed, rejected, canceled, declined_recording, opted_out. rejected means safety screening declined the objective before dialing; error.message and resolution.summary explain why. Terminal calls include a `resolution` {outcome: achieved|partially_achieved|not_achieved|no_conversation, summary, commitments[], follow_ups[], key_facts[]} — read it before the transcript; it is the answer. Completed calls also carry the transcript (may be marked transcript_truncated) and a recording_url the user can listen to (private links expire after ~1 hour). The resolution can lag the terminal status by a few seconds while it is extracted. Calls include the resolved locale, language, voice, and is_public; public calls return token-free listen and recording URLs that remain accessible while public. Use set_call_visibility only when the user explicitly asks to publish or make private. Every call includes a listen_url — the call page for the user (live audio + transcript while in progress; recording + transcript once ended). ended_by='user' on a canceled call means the user deliberately stopped it from that page — don't retry. recording_status=processing means the recording is still being prepared by the carrier (typically ready 15–90s after the call ends) — call get_call again, or pass wait_seconds on an already-terminal call to wait for the recording_url.

输入参数call_id必填wait_seconds

list_calls

List calls

List the authenticated user's recent calls, newest first, with statuses and outcomes. Useful to recover a lost call_id or review call history. Pass the returned next_cursor as cursor to fetch the next page of older calls. next_cursor is null when no older calls remain. Omit cursor to start at the newest calls. limit defaults to 20 (1–100) on every request.

输入参数limitcursor

set_call_visibility

Set call visibility

Make a call owned by the authenticated user public or private. Calls are private by default. Only publish when the user explicitly asks: a public call exposes its objective, phone numbers, transcript, result and audio to anyone with its permanent, token-free listen_url, including live audio while in progress. Set is_public to false to disable token-free access again; existing private signed links remain valid until they expire. Public visitors cannot stop the call or change visibility.

输入参数call_id必填is_public必填

end_call

End call

End an in-progress call early. The voice agent wraps up politely ("I'll let you go, thanks!") and hangs up; the call is force-terminated if wrap-up takes more than ~10 seconds. Use when the user changes their mind or the call is no longer needed. The partial transcript and resolution are still produced.

输入参数call_id必填

get_phone_number

Get connected phone number

Get the phone number connected to the authenticated user's account in E.164 format, along with its caller_id_status (verified or unverified). This is the user's own number used as outbound caller ID, not the service's SMS sending number. Takes no arguments and does not place a call or send a text.

无需输入参数。

版本历史

1
  1. v0.2.0最新Oct 11, 2026

评分

已满足适用检查项的 83%

该评分衡量此收录条目的完整程度,依据注册中心条目、端点与公开来源计算得出。它不是对服务器本身的评价,也不表示服务器安全或当前可用:可连通性是带日期的检查结果,评分也不包含安装量或用户评价。

  • 近 30 天内可连通必需25 分
  • 已声明认证方式必需10 分
  • 已列出工具必需15 分
  • 描述必需15 分
  • 图标10 分
  • 文档15 分

    请声明包含 README 的 GitHub 仓库,或在 OAuth 受保护资源元数据中提供 resource_documentation。

  • 许可证不适用

    托管服务按其提供方的条款使用,因此这里不要求许可证。此检查项不计入百分比。