I’ve been waiting for email support response since Jan 31 regarding a critical SIP Trunk outbound issue.
PROBLEM:
Outbound calls via Custom Telephony fail. tcpdump on my SIP server shows NO packets from Retell, even after whitelisting CIDR 18.98.16.120/30 as instructed by support.
CONFIG:
Termination URI: 160.251.184.117:5060
Transport: UDP
Outbound Agent: Configured
ERROR:
SIP status code: undefined
Error: twirp error unknown: canceled
INVESTIGATION:
I ran tcpdump during outbound call attempts. No packets from any Retell IP arrived at my server. Only keepalive traffic from my upstream provider was visible.
My SIP provider test account expires Feb 6. I urgently need someone to check if SIP INVITE is actually being sent from Retell’s side.
Call IDs:
call_c4194400d6f782d76b67d94f4da
call_d453704f407c91f07bc9f822bda
Has anyone successfully made outbound calls with Custom Telephony (SIP Trunk)? Any advice appreciated.
Based on your configuration and the error you’re experiencing, here are several troubleshooting steps:
Transport Protocol Issue
Your configuration shows UDP transport, but there’s a documented case where switching from UDP to TCP resolved similar outbound SIP issues(1). The Retell SIP server supports TCP, UDP, and TLS(2).
To use TCP, append the transport method to your termination URI: sip:160.251.184.117:5060;transport=tcp(2).
Termination URI Format
Verify your Termination URI format is correct. The standard format should be the SIP server URL(2). Your current format 160.251.184.117:5060 may need the sip: prefix: sip:160.251.184.117:5060(2).
IP Whitelisting
You’ve whitelisted 18.98.16.120/30, which is correct for all regions(2). However, if you’re in the United States, additional IP blocks may be relevant: 143.223.88.0/21 and 161.115.160.0/19(2).
Firewall Configuration
A similar issue was resolved by allowing incoming TCP traffic from Retell to the custom telephony server(1). Ensure your firewall rules permit inbound traffic from Retell’s IP ranges on the appropriate port and protocol.
Disconnection Reasons
When outbound calls show dial_failed, it represents cases where SIP error codes are not available or the error is unknown(3). The twirp error unknown: canceled suggests the connection attempt was terminated before establishing.
Authentication
If you’re using SIP trunk authentication, ensure the username and password are correctly configured(4). Some providers require specific SIP headers for authentication(4).
Next Steps
Switch to TCP transport and test
Verify firewall rules allow inbound traffic from Retell’s IPs
To correctly import or use a phone number in Retell AI, you need to use the E.164 format, which includes the country code prefixed with a plus sign (+), followed by the number with no spaces or special characters(1)(2)(3).
For example, a valid E.164 format would be "+14157774444"(1)(2)(3).
When importing a phone number via the Import Phone Number API, the phone_number parameter must be in E.164 format(1). Similarly, when updating a phone number, the path parameter phone_number should also follow the E.164 format(2).
If you’re experiencing issues with number format:
Ensure the number includes the + prefix followed by the country code(1)(2)(3)
Remove any spaces or special characters from the number(1)(2)(3)
For example, a US number should look like "+12137771234" rather than "(213) 777-1234" or "2137771234"(3)
Thank you for your suggestion. I have now re-registered the phone number in E.164 format (+81504560613) and configured both Inbound and Outbound agents.
However, the issue persists:
SIP status code: undefined
Error: twirp error unknown: canceled
I am still running tcpdump on my server with firewall DISABLED, and I still receive ZERO packets from any Retell IP address.
When using Retell AI with custom telephony providers via SIP trunking, calls will appear in your telephony provider’s call history because the calls are routed through your provider’s infrastructure(1). This applies whether you’re using Elastic SIP Trunking (the recommended method) or dialing to SIP URI(1).
With Elastic SIP Trunking, you connect Retell’s VoIP with PSTN through your existing telephony provider, so the call traffic flows through their system and will be logged in their records(1)(2).
However, Retell AI also maintains its own comprehensive call tracking. You can retrieve detailed call information through the List Calls API, which provides call status, transcripts, recordings, analytics, and more(3). This gives you visibility into your AI agent calls within Retell’s platform regardless of what appears in your telephony provider’s logs(3).