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When attempting to use the Transfer call to a sip,...
# ❓・ai-evy-support
j
@AI Evy When attempting to use the Transfer call to a sip, I get an error. I used the example in the docs: in the format of sip:username@domain (e.g. sip:user@retellai.com). The call history shows the error: SIP status code: undefined SIP error category: unknown Error: twirp error unknown: server required auth, but no username or password was provided. How would i provide username/password credentials for the transfer?
a
Thanks for tagging me @Jeremy. I'll look through relevant sources and get back to you shortly.
### Answer Based on the available documentation, the Transfer Call feature allows you to set a transfer number in SIP URI format (
sip:username@domain
) , but there is no documented option to provide username/password credentials directly within the Transfer Call tool configuration itself. [(1) ]() [(2) ]() The authentication error you're seeing suggests the destination SIP server requires credentials that aren't being passed. Potential workarounds: 1. **Custom SIP Headers**: You can add custom SIP headers to transfers (all header names must start with
X-
) , but standard SIP authentication credentials typically aren't passed this way. 2. **Configure authentication on your SIP provider side**: If you're using custom telephony with elastic SIP trunking, authentication is typically configured at the trunk level (via IP whitelisting or username/password credentials) when setting up the termination SIP URI in your provider. [(3) ]() 3. **Contact the destination SIP server administrator**: They may need to whitelist your originating IP or provide an alternative authentication method. Unfortunately, I don't see a documented way to embed SIP authentication credentials directly in the transfer call configuration. You may want to reach out to Retell support for guidance on this specific use case. Check out these potentially useful sources: - [Transfer call]() - [Custom Telephony Overview]() - [Contact Sales]() Would you like me to try providing a general answer based on SIP authentication patterns?
​ To learn more, see the sources I considered: [[1] Transfer call](https://docs.retellai.com/build/single-multi-prompt/transfer-call) [[2] Call Transfer Node](https://docs.retellai.com/build/conversation-flow/call-transfer-node) [[3] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
j
what IP are the transfers coming from?
@AI Evy would that be lk-real-ip listed in the call logs?
a
Thanks for the follow-up, @Jeremy. I'll get back to you soon.
### Answer --- I don't have specific documentation confirming whether
lk-real-ip
in the call logs represents the originating IP address for SIP transfers. The available sources mention that Retell's SIP server does not currently have a static IP address. [(1) ]() When configuring Twilio elastic SIP trunking termination, the documentation notes you would need to whitelist IP addresses or use username/password authentication, but it doesn't specify what those IP addresses are or confirm that
lk-real-ip
is the relevant field. For the specific IP addresses used for outbound SIP transfers, I'd recommend reaching out to Retell support directly: - [Contact Sales]() - [Twilio SIP Trunking Guide]() - [Custom Telephony Overview]() Would you like a general answer about SIP header fields and IP identification patterns?
​ To learn more, see the sources I considered: [[1] Twilio](https://docs.retellai.com/deploy/twilio) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
Appreciate the feedback! Feel free to ask me again with additional guidance or context.