Hello Vapi Support Team, We are currently using assistant-based warm transfer (experimental) with our SIP trunk infrastructure (FreePBX / Asterisk – PJSIP), and we are experiencing a reproducible audio issue after transfer. This issue is impacting production and has been reproduced multiple times by our SIP provider. We are not using Twilio numbers. According to your documentation, assistant-based warm transfer (experimental) is the only supported method in our configuration. Relevant documentation: https://docs.vapi.ai/calls/assistant-based-warm-transfer Observed behavior: SIP signaling remains active. Both parties are connected. Hold music stops as expected. The assistant announces that the transfer has been completed. However, no audio flows between the caller and the advisor. Both parties are connected but cannot hear each other. Important: Audio works perfectly before the transfer phase. The issue appears strictly after the final bridge. The behavior is consistent and reproducible. In earlier tests, the assistant sometimes announced: “could not transfer the call” This behavior no longer occurs. Currently, the assistant confirms that the call has been transferred, but the final audio path is not established. Our SIP provider confirms that: The signaling layer remains active. The issue only occurs after the bridge phase. The behavior does not occur when performing direct transfers. The behavior does not occur in outbound-call → warm transfer scenarios. The issue appears specific to this call flow. They suspect the problem may occur during the final media handling phase after transfer. We would appreciate clarification on the following points: How does assistant-based warm transfer manage media during the final bridge? Does Vapi remain media-anchored during and after the transfer? Are there specific requirements or limitations when using SIP trunks instead of Twilio numbers? Is this a known limitation of the experimental feature? We can provide SIP traces if needed. This issue is currently blocking part of our production deployment. Thank you.

Thanks for reaching out about your SIP Trunking setup! You can find step-by-step troubleshooting help here: 👉 [Vapi SIP Trunking Guide](https://docs.vapi.ai/advanced/sip/sip-trunk) If that doesn’t resolve it, please reply with: • Your Call ID • The exact error message or behavior (e.g. 401 Unauthorized, timeout, etc.) • Whether it’s happening on inbound, outbound, or both trunks • Any recent configuration changes Once we have that info, we can help pinpoint the issue faster.

Call id : 019c4c54-59c2-711d-91b3-c71a298db946 We don’t have error like said before It happend on both trunks