Patch MCP

com.patchmcp由 patchmcp.com 發布官方發布者v0.2.0更新於 Oct 24, 2026

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

已驗證Streamable HTTP可網頁執行效率與工作流溝通與協作

概覽

AI 產生的概覽

透過語音代理以你本人已驗證的號碼撥打真實電話,並回傳通話狀態、逐字稿與結構化結果。

可以做什麼
  • 撥號前估算線路可用性、每分鐘用量與剩餘額度
  • 以已驗證號碼依指定目標撥出電話,由即時語音代理完成通話
  • 查詢通話狀態與結構化結果,可選擇短暫等待狀態變化
  • 列出近期通話及狀態與結果,並翻頁查看更早紀錄
  • 提前結束進行中的通話,以及把通話設為公開或私人
  • 讀取帳號綁定的本機號碼(E.164 格式)及其來電顯示驗證狀態
適用情境
適合需要助理真正替使用者打電話並回報通話內容的情境,例如訂位、詢問問題、催查訂單。也適合回顧歷史通話或中止不再需要的通話。
執行需求
遠端 streamable HTTP 端點,不需安裝套件或本機執行環境。需要已驗證的帳號,並綁定一個已驗證、作為發話號碼顯示的電話號碼。涉及任何非美國/加拿大號碼的通話需要付費方案,用量依額度計費。
安裝前請注意
會撥打真實且計費的電話並消耗計量額度,建議先做估算。通話預設為私人,但一旦設為公開,其目標、電話號碼、逐字稿、結果與錄音(包括進行中的即時音訊)將對持有收聽連結的任何人可見,僅在使用者明確要求時才公開。提前結束通話仍會保留部分逐字稿與結果。

安裝

在 SourceWeft 中

  1. 開啟 儀表板中的 Patch MCP,將其新增到工作區。
  2. 為需要使用其工具的對話啟用該服務。

Web executable,透過 Streamable HTTP。 遠端服務在工作區中設定後即可從網頁執行環境執行。

交給代理程式安裝

把這段提示詞貼上到 Claude Code、Codex、Cursor 等代理程式,或 SourceWeft 對話中。代理程式會讀取該伺服器的安裝指南,先向你展示發布者、端點和登入方式,經你同意後再把它新增到你的客戶端。金鑰不會寫進任何命令。

閱讀 https://sourceweft.com/mcp/com.patchmcp%2Fpatch/install.md,並按其中說明把這個 MCP 伺服器接入我的客戶端。修改任何設定之前,先告訴我它的發布者、端點和登入方式,等我確認。不要把 API 金鑰寫進命令或我會提交的檔案;問我金鑰應從哪裡取得。

檢視代理程式所依據的安裝指南

其他 MCP 客戶端

把它新增到你客戶端的 mcpServers 設定中。

{
  "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.

輸入參數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

輸入參數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.

輸入參數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.

輸入參數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.

輸入參數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.

輸入參數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.

無需輸入參數。

版本歷史

1
  1. v0.2.0最新Oct 11, 2026

評分

已滿足適用檢查項的 83%

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  • 近 30 天內可連通必需25 分
  • 已宣告認證方式必需10 分
  • 已列出工具必需15 分
  • 描述必需15 分
  • 圖示10 分
  • 文件15 分

    請宣告包含 README 的 GitHub 儲存庫,或在 OAuth 受保護資源後設資料中提供 resource_documentation。

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