BYOC SIP-trunk import + inbound SIP questions for caller ID and header passthrough

Hi Retell team,

We are Proto Health, and we are currently building an AI calling integration for a pilot the week of May 4.

To meet our pilot requirements, patients must see the clinic’s own number as the caller ID. Additionally, the clinic’s PBX must be able to transfer inbound calls to a Retell agent while preserving patient context in SIP headers. We have several technical questions regarding outbound BYOC and inbound SIP that are blocking our integration decisions before the April 25 trunk provisioning cutoff:

Outbound BYOC Caller ID

1. Does Retell perform number-ownership verification (LOA, CNAM lookup) for BYOC SIP-trunk imports, or is trust based on the trunk accepting the call?

2. On outbound calls via an imported trunk, does Retell set the P-Asserted-Identity / From headers, or pass through the trunk’s settings?

3. Can we dial out from a US-imported number with a different display-only override or CNAM?

4. What is the current production turnaround time for Verified/Branded Caller ID business-profile approval in the US?

5. For the “Dial to SIP URI” path (registerPhoneCall), does the caller ID reflect our upstream trunk’s setting (e.g., Twilio), or does Retell override it?

6. Are there specific concurrency or rate limits for imported SIP trunks compared to Retell-provisioned numbers?

Inbound SIP

7. Does Retell accept inbound SIP INVITEs from a third-party PBX? If so, what endpoint and authentication methods (IP allowlist, digest auth, mTLS) are supported?

8. If a third-party PBX transfers a call with custom SIP headers (e.g., X-Patient-ID), are these exposed to the agent runtime as dynamic variables for the LLM?

9. Can a Retell agent hand off a call mid-conversation to an arbitrary SIP URI? Does a built-in transfer tool exist, and does the handback preserve the original caller’s PSTN identity?

Since these features are blocking, we would appreciate your feedback soon so we can finalize our provisioning.

Hey @manish

I’ve answered some of your questions, and I’ve forwarded the remaining ones to our team for further details.

  1. Caller ID is controlled by your trunk. For Twilio, you configure caller ID via SIP Header Manipulation policies on the Termination side (replace the From header). Retell does not override it on BYOC outbound.

  2. Yes — on Twilio you can verify a display number and use a Header Manipulation rule to replace the From header so outbound from the imported trunk displays that number.

  3. For Dial‑to‑SIP URI (registerPhoneCall), Retell does not make/receive the call itself — your upstream system dials to sip:{call_id}@sip.retellai.com, so caller ID reflects your upstream trunk settings.

  4. Yes. Origination SIP URI is sip:sip.retellai.com (with ;transport=tcp|udp|tls). SRTP requires TLS. Retell’s SIP SBC IP blocks: 18.98.16.120/30, 143.223.88.0/21, 161.115.160.0/19. For auth beyond IP/credentials (e.g., mTLS, static IP), Retell points to telephony partners like Jambonz or Cloudonix.

  5. Yes. Any inbound header starting with X-/x- is auto-extracted into call.custom_sip_headers and added to dynamic variables (prefix stripped), directly usable by the LLM. For pre-setting dynamic variables, Twilio-style sip.h.x-... format is also supported.

  6. Yes — the built-in Transfer Call tool supports a SIP URI destination (sip:user@domain), warm or cold. You can configure caller ID to show the original user’s number (provider must support caller-ID override; Twilio supports both warm/cold, Telnyx cold-only via SIP REFER).

Thank You

Hi @manish

Here are the answers to your remaining questions :

  1. Number ownership verification for BYOC SIP-trunk imports: Retell does not perform number-ownership verification (LOA, CNAM lookup) for BYOC SIP-trunk imports - it is trust-based. Identity verification (KYC) is required for Retell-provisioned numbers. For imported/BYO telephony numbers, KYC verification is required for outbound calls.

  2. Current turnaround time for Verified/Branded Caller ID approval in the US: The official timeframe is 1-2 weeks for approval, but it could take longer. The process involves external carrier validation which is managed by third parties and can experience delays

  3. There are no additional concurrency or rate limits specific to imported SIP trunks compared to Retell-provisioned numbers. Custom Telephony (SIP trunks) has a default CPS (Calls Per Second) limit of 1 and concurrency limit of 20. You can purchase more concurrency and then increase your CPS limit.

Thank You