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I'm getting an error intermittently when trying to...
# ❓・ai-evy-support
v
@AI Evy I'm getting an error intermittently when trying to transfer a call "error: Error transferring call, SIP status code: 486 SIP error category: user_busy Error: twirp error unknown: INVITE failed: sip status: 486: Busy here" I am using E.164, Cold Transfer SIP Invite and displaying the users phone number on the caller ID>
a
Thanks for tagging me @vteck101. I'll look through relevant sources and get back to you shortly.
### Answer The SIP 486 error ("Busy here") indicates the transfer destination number is currently busy or on another call—this is a standard telephony response, not a Retell configuration issue. [(1) ]() The
dial_busy
disconnection reason represents cases where the number dialed is busy. ## Troubleshooting Steps 1. Verify the destination can accept calls - The transfer destination may be on another call or have call waiting disabled. Try the transfer when you know the destination line is free. 2. Check call history and detailed logs - The logs contain the disconnection reason, SIP error code, and error message, which can help confirm the root cause. 3. Consider the caller ID configuration - Since you're using "User's Number" as the displayed caller ID with cold transfer (SIP INVITE) , note that caller ID override depends on your telephony provider supporting it. If caller ID override is not supported, the transfer would fail. [(2) ]() 4. Test with Agent's Number as caller ID - Try temporarily switching to "Retell Agent's number" as the caller ID to isolate whether the issue is related to caller ID override. 5. If using custom telephony - Check with your telephony provider to see what error occurred on their side, and verify your termination settings are configured correctly. [(1) ]() , [(3) ]() Since this error is intermittent and specifically returns "Busy here," the most likely cause is that the destination number is genuinely busy at those moments. If the destination should be available, the caller ID override configuration is worth investigating.
​ To learn more, see the sources I considered: [[1] Debug outbound connection issues](https://docs.retellai.com/reliability/debug-outbound-call) [[2] Call transfer failures](https://community.retellai.com/t/call-transfer-failures/600) [[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.
v
@AI Evy I also get this error "error: Error transferring call, SIP status code: undefined SIP error category: unknown Error: twirp error unknown: max auth retry attemps reached"
a
Thanks for the follow-up, @vteck101. I'll get back to you soon.
### Answer The "max auth retry attempts reached" error indicates an authentication failure with your SIP trunk configuration. [(1) ]() This error occurs when your telephony provider requires authentication credentials (username/password) , but either: - No credentials were provided when importing the phone number - The credentials provided are incorrect ## Troubleshooting Steps 1. Verify SIP trunk credentials - If you're using custom telephony (e.g., Twilio) , ensure your Retell phone number is configured with the correct username and password that matches your SIP trunk authentication settings. 2. Re-import the number with correct auth - If the configuration is incorrect, delete the imported number and re-import it with the proper authentication credentials. [(2) ]() 3. Check with your telephony provider - Confirm what authentication credentials are required for your SIP trunk and ensure they match what you've configured in Retell. 4. Review detailed logs - Check the call detailed log for the specific call IDs where this occurred—it contains the disconnection reason, SIP error code, and error message that can help identify the root cause. A similar issue was reported where transfers failed with the error "server required auth, but no username or password was provided"—the resolution was to configure the associated Retell phone number with the correct SIP trunk credentials. [(1) ]()
​ To learn more, see the sources I considered: [[1] URGENT: Newly created call transfers not working](https://discord.com/channels/1186424376300679209/1440817497203802243) [[2] Debug outbound connection issues](https://docs.retellai.com/reliability/debug-outbound-call) ​
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.