***

title: Call Transfers
subtitle: Enable your AI agent to seamlessly transfer calls to a human.
slug: call-transfers
description: >-
Learn how to configure and manage call transfers within Synthflow, including
TEL, SIP, and dynamic routing options.
--------------------------------------

Call transfers allow your AI agent to seamlessly hand off conversations to a human when needed. In Synthflow, these handoffs are managed using the **call transfer action**.

Synthflow supports four transfer types, each suited to different use cases:

| Transfer Type                                   | Format                                | Best For                                                   |
| ----------------------------------------------- | ------------------------------------- | ---------------------------------------------------------- |
| [TEL (Phone Number)](/transfer-calls-to-humans) | E.164 format (e.g., `+14155551234`)   | Direct transfers to mobile or landline numbers             |
| [SIP](/call-transfer-to-sip)                    | SIP URI (e.g., `sip:user@domain.com`) | Transfers to PBX systems, softphones, or SIP trunks        |
| [Dynamic](/dynamic-transfers)                   | Runtime-determined                    | Transfers where the destination is fetched during the call |
| [Phonebook](/create-a-phone-book)               | Managed list                          | Multiple destinations with conditional routing             |

***

## <Icon icon="phone" /> TEL (Phone Number) Transfers

Use TEL transfers for direct handoffs to a phone number in E.164 format. This is the default transfer option in Synthflow, ideal for routing calls to mobile phones, landlines, or call center queues.

<Note>
  TEL URI transfers are confirmed to work with **Twilio** and **Telnyx**. If you're using a different carrier or a self-hosted PBX, TEL URI handling may not be supported. In that case, use a [SIP transfer](/call-transfer-to-sip) instead, as SIP URIs are more universally supported across carriers and PBX platforms.
</Note>

**Key features:**

* Supports warm and cold (blind) transfer modes
* Configurable timeout and transfer messages
* Optional background music during transfer

[Learn more about phone number transfers](/transfer-calls-to-humans)

***

## <Icon icon="server" /> SIP Transfers

Use SIP transfers when routing calls to SIP-compatible endpoints such as PBX systems, SIP phones, or softphones. SIP transfers use a SIP URI format instead of a phone number.

**Key features:**

* Works with any SIP-compatible endpoint (Asterisk, FreePBX, etc.)
* Supports warm and cold transfer modes
* Context preservation with conversation summaries

[Learn more about SIP transfers](/call-transfer-to-sip)

***

## <Icon icon="bolt" /> Dynamic Transfers

Use dynamic transfers when the destination number isn't known in advance. Your agent can fetch the target number during the call using a **custom action** or a **pre-call webhook**, enabling flexible routing based on caller input, business logic, or real-time data.

**Key features:**

* Fetch phone numbers from CRM systems or APIs during the call
* Route to on-call staff based on schedules
* Location-based or tier-based routing

[Learn more about dynamic transfers](/dynamic-transfers)

***

## <Icon icon="address-book" /> Phonebook Transfers

For scenarios where you have multiple potential transfer destinations, use **phonebooks** to manage them in one place. A phonebook allows your agent to transfer to different numbers based on conditions, all from a single action.

**Key features:**

* Centralized management of transfer destinations
* Conditional routing based on caller input or context
* Easy updates without modifying agent configuration

[Learn more about phonebooks](/create-a-phone-book)

***

## <Icon icon="shuffle" /> Transfer Modes

All transfer types support the following modes:

| Mode               | Description                                                       | Caller ID Shown          | SIP Method |
| ------------------ | ----------------------------------------------------------------- | ------------------------ | ---------- |
| **Blind (Cold)**   | Transfer immediately without speaking to the recipient            | Original caller's number | SIP REFER  |
| **Warm (Message)** | Play a private whisper message to the recipient before connecting | Synthflow agent's number | SIP INVITE |
| **Warm (Summary)** | Provide an AI-generated conversation summary to the recipient     | Synthflow agent's number | SIP INVITE |

<Note>
  The whisper message in warm transfers is heard only by the receiving agent, not the caller. This allows you to provide context like "Incoming call about a billing dispute" before the agent accepts.
</Note>

### Human Detection

When using warm transfers, enable **Human Detection** to ensure the whisper message or summary is only delivered once a live human answers. This prevents the briefing from playing to voicemail, IVR queues, or empty air.

You can configure a **Human Detection Timeout** to control how long the AI waits for a live human before treating the transfer as failed.

### Feature Comparison

| Feature                 | Blind (Cold) | Warm (Message)       | Warm (Summary)   |
| ----------------------- | ------------ | -------------------- | ---------------- |
| Recipient hears context | No           | Yes (custom message) | Yes (AI summary) |
| Caller ID passthrough   | Yes          | No                   | No               |
| Hold music for caller   | Yes          | Yes                  | Yes              |
| Human detection         | No           | Yes                  | Yes              |
| Retry on failure        | No           | Yes                  | Yes              |

***

## <Icon icon="id-card" /> Caller ID Behavior

How caller ID is displayed depends on the transfer mode and your telephony provider:

* **Blind (cold) transfers:** Use SIP REFER, which passes through the original caller's number. The receiving party sees who is actually calling.
* **Warm transfers:** Use SIP INVITE to create a new call leg, so the receiving party sees your Synthflow agent's phone number.

<Note>
  Caller ID passthrough for cold transfers requires your telephony provider to support SIP REFER. Most major providers (Twilio, Telnyx, Vonage) support this, but verify with your provider if you experience issues.
</Note>

***

## <Icon icon="triangle-exclamation" /> Transfer Failure Handling

When a transfer fails (timeout, busy, no answer), behavior depends on the transfer mode:

| Scenario                 | Blind Transfer | Warm Transfer                            |
| ------------------------ | -------------- | ---------------------------------------- |
| Recipient doesn't answer | Call ends      | Agent can retry or continue conversation |
| Number is busy           | Call ends      | Agent can retry or continue conversation |
| Invalid number           | Call ends      | Agent informed, can try alternate        |

For critical transfers, use **warm transfer mode** to allow graceful failure recovery. You can also configure:

* **Timeout duration:** How long to wait before considering the transfer failed
* **Fallback behavior:** What the agent should say or do if the transfer fails

***

## <Icon icon="music" /> Hold Experience

While the caller waits for the transfer to connect, you can customize their experience:

* **Hold music:** Enable background music during the transfer wait time
* **Transfer announcement:** Optionally announce "Please hold while I transfer you" before the music starts

Configure these options in the transfer action's settings.