> ## Documentation Index > Fetch the complete documentation index at: https://plivo.com/docs/llms.txt > Use this file to discover all available pages before exploring further. # Transfer to a Human Agent by DID Forward > Transfer an AI-handled call to a human agent's phone number with Dial XML and the Number element. > [!NOTE] > > * **What:** Transfer a call from your AI agent to a human agent's phone number with `` and ``. > * **Watch out:** On a `` with `keepCallAlive="true"`, [close your WebSocket](#if-your-call-is-on-a-live-audio-stream) when you transfer. The simplest transfer. When your AI agent decides to transfer, return this XML. Plivo dials the phone number in ``: ```xml theme={null} Connecting you to a human agent now. +14155551234 ``` ## How a DID Forward Works ![DID forward in five steps. 1: the caller is on a call with your AI agent. 2: your app returns Dial with Number to Plivo. 3: Plivo calls the phone number. 4: the human agent answers. 5: the caller now talks to the human agent.](https://mintcdn.com/plivo/9zIRdvx4doyoicBb/images/transfer-to-human-agent/did-forward-sequence-light.svg?fit=max\&auto=format\&n=9zIRdvx4doyoicBb\&q=85\&s=5d967fbd27014698000817d0a846d48d) * Plivo originates an outbound call from your AI agent's leg to the human agent's phone number * When the human agent answers, the two legs are bridged * The caller and the human agent are connected, and your AI agent leaves the call ## Trade-offs of a DID Forward **Pros** * **Works with any phone** - mobile, landline, or a softphone with a DID * **Simple to set up** - needs nothing from your network team **Cons** * **Outbound call cost** - billed for the full duration of the transferred call * **Agents need a real phone number** * **Less flexibility** - no custom SIP headers, no routing to specific agent IDs or queues * **No warm transfer** - your AI agent can't brief the human agent before the caller is connected. For that, use [Option 3: Multiparty Call](/docs/voice/use-cases/transfer-to-human-agent/multiparty-call/) > [!TIP] > > A DID forward is also the fastest way to prove a transfer problem is *not* your application. If a transfer to a phone number connects but a SIP transfer to the same team doesn't, the fault is on the SIP path — the firewall or the credentials — not in your XML. *** ## If Your Call Is on a Live Audio Stream This applies when your AI agent runs over [audio streaming](/docs/voice/xml/audio-streaming/) and the call is on a `` with `keepCallAlive="true"`. With `keepCallAlive="true"`, [the `` element runs exclusively and subsequent XML executes only after the stream disconnects](/docs/voice/xml/audio-streaming/). Your transfer XML is subsequent XML, so it does not run while the stream is still open. If the WebSocket never closes, the transfer never runs. > [!WARNING] > > Close your bot's WebSocket when you decide to transfer. Keep `keepCallAlive="true"` — closing the socket is what lets the call continue to your transfer XML, not removing the attribute. ![When your AI agent closes its WebSocket. Too early: the call ends; fix by adding Wait right after Stream. On time: the transfer works. Stays open: the transfer never runs; fix by closing the WebSocket when you transfer, or use Option 3.](https://mintcdn.com/plivo/9zIRdvx4doyoicBb/images/transfer-to-human-agent/websocket-close-window-light.svg?fit=max\&auto=format\&n=9zIRdvx4doyoicBb\&q=85\&s=4146a9d4b85d1ce4c46c8e9e2fc29f59) Add a `` right after `` in your answer XML. If your AI agent closes its WebSocket before your transfer arrives, the call waits for it instead of ending: ```xml theme={null} wss://your-ai-agent.example.com/stream ``` Set `length` to cover the time between your AI agent closing its WebSocket and your server calling [Transfer a Call](/docs/voice/api/calls/#transfer-a-call). See the [Wait](/docs/voice/xml/routing/#wait) reference for its attributes. If you stop the stream through the API rather than closing the socket, **trigger the transfer first, then send** `DELETE /v1/Account/{auth_id}/Call/{call_uuid}/Stream/`. Stopping the stream first lets the leg continue to the end of the XML document it is already running, which can end the call instead of transferring it. `streamTimeout` defaults to `86400` seconds. Set it to a realistic ceiling for your call shape so a stream that never closes cannot hold a call open. If transfers still drop the call or never run, use [Option 3: Multiparty Call](/docs/voice/use-cases/transfer-to-human-agent/multiparty-call/) instead. In a Multiparty Call, the caller's call doesn't run ``, so nothing waits for the WebSocket to close. *** ## Server-Initiated Handoff Instead of waiting for your answer URL to return Dial XML, you can trigger the transfer programmatically with the [Transfer a Call API](/docs/voice/api/calls/#transfer-a-call), which points a live call leg at a new URL. That URL returns the same `` XML shown above. For SIP auth forward, include `sipAuthUsername` and `sipAuthPassword` on ``. For a warm transfer, where your AI agent briefs the human agent before the caller is connected, use [Option 3: Multiparty Call](/docs/voice/use-cases/transfer-to-human-agent/multiparty-call/#warm-transfer-with-a-multiparty-call). *** ## Related * [Transfer to Human Agent](/docs/voice/use-cases/transfer-to-human-agent/) — When to transfer, and how to choose an option * [Option 2: SIP Auth Forward](/docs/voice/use-cases/transfer-to-human-agent/sip-auth-forward/) — Transfer to a SIP endpoint with credentials * [Option 3: Multiparty Call](/docs/voice/use-cases/transfer-to-human-agent/multiparty-call/) — Cold or warm transfer in a Multiparty Call * [Troubleshoot Transfer to Human Agent](/docs/voice/use-cases/transfer-to-human-agent/troubleshooting/) — Test a transfer and fix failed ones * [Dial XML](/docs/voice/xml/routing/) — Complete reference for `` and ``