​ To learn more, see the sources I considered: [[1] Call transfer node in conversation flow](https://docs.retellai.com/build/conversation-flow/call-transfer-node) [[2] Transfer call tool (single & multi-prompt agents)](https://docs.retellai.com/build/single-multi-prompt/transfer-call) [[3] 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) [[4] Warm Transfers](https://discord.com/channels/1186424376300679209/1289365041010835518) ​

### Answer During a warm transfer, while the agent is dialing and speaking with the destination (e.g., the clinic) before connecting the caller, the caller hears the configured on-hold music. The default is a standard ringtone, and this can be changed to other options like relaxing sound or uplifting beats [(1) ]() [(2) ]() [(3) ]() . There is a known limitation reported on Discord: in some setups, the caller has been able to hear what the AI agent says to the destination (e.g., the whisper/debrief message) instead of the on-hold music playing during that period. This was reported as an issue that the Retell team placed on their roadmap without a short-term fix at the time [(4) ]() . Since this comes from a Discord discussion rather than official documentation, treat it as anecdotal rather than a documented guarantee of current behavior.

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

​ To learn more, see the sources I considered: [[1] Connect Retell voice agents to custom telephony](https://docs.retellai.com/deploy/custom-telephony) [[2] Avaya](https://docs.retellai.com/deploy/avaya) ​

### Answer I don't have specific information in these sources about troubleshooting dropped calls for an inbound agent on custom telephony. What I can point to from these sources related to call connectivity on custom SIP setups: - If using elastic SIP trunking, inbound issues are typically tied to origination settings in your SIP trunking provider — checking your provider's logs and possibly opening a ticket with them is recommended [(1) ]() . - Retell will not throw errors if your SIP configuration is incorrect; it only becomes apparent when a call is actually made, so misconfigurations can silently cause call issues . - For the "Dial to SIP URI" method, the call must be dialed to the SIP URI within 5 minutes of calling the Register Phone Call API — if the call isn't connected within that window, it disconnects with

registered_call_timeout

. - If using Avaya SBCE, common call-drop symptoms and fixes are documented, such as calls dropping around 30 seconds being caused by SIP re-INVITE being blocked mid-call (fix: allow mid-call SIP signaling through the firewall) , and one-way/no audio issues tied to RTP port blocking or NAT/topology hiding problems [(2) ]() . None of these sources describe a general "calls dropping" issue for inbound agents outside the custom telephony/SIP context (e.g., Twilio-native setup) , so I don't want to guess at settings not documented here. Would you like me to give a general answer based on my own knowledge instead, since these sources don't fully cover your scenario?

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