Hey folks, quick question on BYO SIP trunk routing.
I’m seeing outbound calls end with endedReason=customer-busy and sipStatus=603 (Decline), but the call record shows phoneCallTransport=pstn and phoneCallProvider=vapi.
Expected: this call should have gone out over my BYO SIP trunk (TLS 5061) to my SBC, with caller ID formatted as E.164 (+971 …).
From/To: +971…80 → +971…033 (redacted)
Any idea why it’s routing via PSTN instead of the SIP trunk, and what I need to do to ensure outbound actually egresses via BYO SIP?
Also, is there a way to get the raw SIP INVITE (headers + SDP) Vapi sent (PCAP like artifact.pcapUrl, or an internal trace) so I can verify From / P-Asserted-Identity formatting?
Calling to UAE numbers via a dedicated SBC in our UAE office.
Best,
Ahmed