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I am trying to connect an inbound phone number fro...
# ❓・ai-evy-support
y
@AI Evy I am trying to connect an inbound phone number from NetGSM (Turkey) via SIP trunking. NetGSM sends inbound calls as SIP INVITE using IP-based authentication (no REGISTER, no username/password). The calls reach NetGSM successfully but are immediately disconnected (1 second), which suggests the INVITE is being rejected on your side. Could you please confirm: Do you support inbound SIP calls via IP authentication (no REGISTER)? If yes, what SIP ingress domain or IP should NetGSM send calls to? Do you require IP whitelisting or any specific headers?
a
Thanks for tagging me @yuzu_gulmeyen. I'll look through relevant sources and get back to you shortly.
### Answer Based on the documentation, here's what I can confirm for your NetGSM SIP trunking setup: SIP Ingress Domain for Inbound Calls NetGSM should send inbound SIP INVITE requests to Retell's SIP server: 
sip:sip.retellai.com
[(1) ]() For specifying transport method, you can use: - TCP (Recommended) : 
sip:sip.retellai.com;transport=tcp
- UDP: 
sip:sip.retellai.com;transport=udp
- TLS: 
sip:sip.retellai.com;transport=tls
IP Whitelisting Retell provides these IP blocks for traffic that NetGSM may need to whitelist: - 
18.98.16.120/30
(All regions) - 
143.223.88.0/21
(United States only) - 
161.115.160.0/19
(United States only) Note: Some non-US traffic might come from IPs not listed above, but will be migrated to the first IP block by 12/03/2025. *Regarding IP-Based Authentication (No REGISTER) * The documentation describes two integration methods—Elastic SIP Trunking and Dial to SIP URI. For the Dial to SIP URI method, you would call the Register Phone Call API to obtain a 
call_id
, then construct the SIP URI as: 
sip:{call_id}@sip.retellai.com
The immediate disconnection you're experiencing suggests the call may not be properly registered with Retell before the INVITE arrives. The documentation doesn't explicitly confirm support for inbound calls via pure IP authentication without prior registration through the API. Recommendation If you're having trouble with the standard Elastic SIP Trunking approach, consider checking out Retell's partner [Jambonz]() , which supports static IP addresses and mutual TLS, and has a dedicated Slack support channel for Retell users.
Feel free to tag @AI Evy with additional questions. I also appreciate feedback, it helps me improve.
​ To learn more, see the sources I considered: [[1] Custom Telephony Overview](https://docs.retellai.com/deploy/custom-telephony) ​
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