​ To learn more, see the sources I considered: [[1] Call Transfer Node](https://docs.retellai.com/build/conversation-flow/call-transfer-node) [[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 --- Yes, you can configure a SIP URI in the Transfer node. [(1) ]() When setting up the transfer target, you can set the transfer number to be either: - A number in E.164 format, or a SIP URI in the format of

sip:username@domain

(e.g.,

sip:user@retellai.com

) - A dynamic variable that gets substituted at runtime **Important limitation**: If you're using the "dial to SIP URI" method for custom telephony (where Retell doesn't directly control the telephony) , you will not be able to use Retell's transfer call feature. [(2) ]() In that case, you would need to implement your own transfer logic using a custom function. For transfers to work properly with SIP URIs, you should use the Elastic SIP Trunking method, which is the recommended approach for custom telephony integration. [(2) ]() [(8) ]()