Hi Retell Support, We are integrating a Custom SIP Trunk (Exotel / Veero Communications) into our workspace. We are encountering a bizarre bridging issue where calls initiated via the Retell Dashboard work perfectly, but calls initiated via the create-phone-call API ring the destination device but fail to bridge audio, timing out with a dial_no_answer / sip request timed out error when answered. Here are the details from our testing today: 1. Successful Call (Via Retell Dashboard UI): Call ID: call_73bfa02f2243539a4f5...61f Result: Audio bridged instantly, AI spoke, call successful. 2. Failed Call (Via create-phone-call API): Call ID: call_dced4e64eafca7ef33e5c76f4b5 Result: Destination phone rings. When answered, there is dead air for 20 seconds, then Retell drops the call with Error: twirp error unknown: sip request timed out. Payload Used: from_number: "02240611290", to_number: "+919769061947" Our SIP provider claims their trunk is working fine because the dashboard call succeeded. Could your engineering team please compare the outbound SIP INVITE traces for these two Call IDs? Since the to_number and from_number are identical in both cases, we suspect there is a difference in the SIP Headers, codec negotiation, or authorization tags between a Dashboard-initiated call and an API-initiated call that is causing the provider's firewall to block the 200 OK signal on the API route. Any insight into what differs in the SIP trace between these two calls would be hugely appreciated! Best regards, Sanjay Singh