​ To learn more, see the sources I considered: [[1] Transfer call](https://docs.retellai.com/build/single-multi-prompt/transfer-call) [[2] Warm Transfer vs Cold Transfer - The AI Advantage in Call Centers | Retell AI](https://www.retellai.com/blog/effortless-handoffs-with-retell-ais-warm-transfer-feature) [[3] Agent Transfer](https://docs.retellai.com/build/single-multi-prompt/transfer-agent) [[4] Agent transfer, MCP Client, Static IP and more](https://www.retellai.com/changelog/agent-transfer-mcp-client) ​

### Answer Based on the available documentation, the AI agent does not stay on the call after a transfer is completed. Here's what happens with each transfer type: **Cold Transfer**: The call is transferred to a destination number and that's it—the AI agent's involvement ends. [(1) ]() **Warm Transfer**: The AI agent can perform actions before connecting the caller to the transfer destination, such as: - Detect if the other side is human - Leave private whisper messages to the receiving agent (not heard by the user) - Deliver a three-way introduction message However, once the transfer is completed and the caller is connected to the human agent, the AI agent does not remain on the call. [(2) ]() If you need the AI to stay involved during a call while switching handling responsibilities, you might consider Agent Transfer instead. This transfers the conversation from one AI agent to another (not to a human) , and the destination agent inherits the full conversation history. [(3) ]() [(4) ]() Could you share more about your use case? That would help determine the best approach for your needs.

Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.

### Answer Based on the available sources, Retell AI's built-in transfer features do not support keeping the AI agent on the call alongside two human participants in a persistent three-way conference. The warm transfer feature allows for a three-way handoff message where the caller can hear the AI briefing the receiving agent before the connection completes, but the AI then exits the call. [(1) ]() A community member in Discord noted: "Not right now. It can only create a conference call and then remove itself. Both transferee and recipient can hear what the AI says." They also mentioned: "You would have to custom code this using call conferencing on Twilio." [(2) ]() So to achieve a true 2-person + 1-AI-agent call where the agent stays active, you would likely need to build a custom solution using your telephony provider's conferencing capabilities (like Twilio's call conferencing API) rather than relying on Retell's native transfer functions. Check out these potentially useful sources: - [Warm Transfer vs Cold Transfer - The AI Advantage in Call Centers | Retell AI]() - [Transfer call documentation]() Would you like a general answer about how such a custom implementation might work?

​ To learn more, see the sources I considered: [[1] Warm Transfer vs Cold Transfer - The AI Advantage in Call Centers | Retell AI](https://www.retellai.com/blog/effortless-handoffs-with-retell-ais-warm-transfer-feature) [[2] Discord message from @ahmad_48064 on 2025-03-27](https://discord.com/channels/1186424376300679209/1186462578310910112/1354844951560716560) ​

Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.

### Answer The available Retell AI documentation doesn't provide specific code examples for building a custom Twilio conference call that keeps the AI agent active alongside two human participants. However, there are a few approaches mentioned that could help you get started: Using Cloudonix as a telephony partner: One option discussed involves using Cloudonix, which has a "fork" API that can split call legs and connect them independently into a conference call. [(1) ]() As described in a video tutorial: "A fork basically says, okay, I got leg A, I got leg B, so let's split them apart, and each one goes into a new application. Now, the new application can be connected both into a conference call, but now I can control each one independently." Using Dial to SIP URI method: If you need more control over telephony, you can use the Dial to SIP URI method, which gives you full control over telephony functionalities. [(2) ]() With this method, "you will need to handle the transfer call/end call logic yourself, meaning that you will need to write a custom function which internally interacts with your telephony provider." Check out these potentially useful sources: - [Custom Telephony Overview]() - [Smarter Call Transfers with Alex & Nir (Cloudonix) ]() - [Cloudonix Partner Page]() For detailed Twilio-specific conferencing code, you would need to refer to Twilio's own documentation on their Conference API, as Retell AI's documentation doesn't include those implementation details. Would you like a general answer on how conference calls typically work with SIP-based systems?

​ To learn more, see the sources I considered: [[1] Smarter Call Transfers with Alex & Nir (Cloudonix)]() [[2] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) ​

Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.

### Answer RingCentral is not explicitly covered in the available Retell AI documentation. However, you can connect RingCentral numbers to Retell AI using one of two general methods for custom telephony providers: *Method 1: Elastic SIP Trunking (Recommended) * [(1) ]() If RingCentral supports elastic SIP trunking, you would: - Set up a SIP trunk in RingCentral and configure your number to point to it - Configure origination to point to Retell's SIP server:

sip:sip.retellai.com

- Import that number to Retell via the dashboard or the Import Number API Method 2: Dial to SIP URI If RingCentral doesn't support elastic SIP trunking or you have a more complex setup: - Use the Register Phone Call API to obtain a

call_id

- Dial the call to the SIP URI:

sip:{call_id}@sip.retellai.com

Note: With this method, you won't be able to use Retell's built-in transfer call feature and would need to implement your own transfer logic. Connection Details: - Retell SIP server URI:

sip:sip.retellai.com

- IP block for whitelisting:

18.98.16.120/30

(All regions) ,

143.223.88.0/21

(certain US traffic) ,

161.115.160.0/19

(certain US traffic) - Supported transport methods: TCP (recommended) , UDP, TLS You can use the Twilio or Telnyx guides as references for the general setup process with your RingCentral account. Check out these potentially useful sources: - [Custom Telephony Overview]() - [Twilio SIP Trunking Guide]() - [Telnyx SIP Trunking Guide]()

​ To learn more, see the sources I considered: [[1] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) ​

Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.