### Answer When using SIP trunking with custom providers like Bandwidth/Sansay, transfer calls route through your configured termination URI [(1) ]() . When you import a number into Retell and specify the termination SIP URI, that same URI is used for outbound calls including transfers [(2) ]() [(3) ]() . The transfer SIP INVITE is sent to the termination URI you configured during number import [(1) ]() . For transfers to work properly with custom SIP trunks, you need to ensure your SIP trunking provider allows SIP REFER and PSTN transfer if you're using cold transfers with the transferee's number showing as caller ID [(4) ]() . If you're experiencing issues with transfers not reaching your SIP proxy, verify that: - Your termination URI is correctly configured without spaces [(3) ]() - Your SIP trunk accepts traffic from Retell's IP ranges (18.98.16.120/30) - Authentication credentials (if using credential-based auth) are properly set

​ To learn more, see the sources I considered: [[1] Call transfer on custom LLM and custom Sip trunk (without Twilio)](https://discord.com/channels/1186424376300679209/1308509166762987520) [[2] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) [[3] Twilio](https://docs.retellai.com/deploy/twilio) [[4] Genesys](https://docs.retellai.com/deploy/genesys) ​

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