Telxi · SIP Trunk Integration
Connecting your Telxi SIP Trunk to Retell AI
Use Retell's recommended Elastic SIP Trunking to run your Telxi numbers on Retell voice agents.
This guide shows how to connect a Telxi SIP trunk to Retell AI using Retell's recommended Elastic SIP Trunking method, so your voice agents can make and receive calls over the PSTN with your Telxi numbers. It covers both authentication methods supported by Telxi — Digest (username/password) and IP-based.
Architecture at a glance
Telxi connection details
Retell connection details
When using IP-based authentication, send these IP blocks to support@telxi.com for whitelisting.
Authentication methods
Username & password
Telxi authenticates with SIP credentials from portal.telxi.com → SIP trunks. Preferred, since Retell's SIP server has no single static IP.
Authorise by source IP
Email Retell's published IP blocks to support@telxi.com for whitelisting; then configure only the SIP host and port on Retell.
Prerequisites
- A Retell AI account (dashboard.retellai.com).
- An active Telxi SIP trunk with at least one DID, in E.164 format.
- Your SIP username/password (Digest), or Retell's IP blocks to whitelist (IP-based).
Get your trunk details from Telxi
What you do on the Telxi side depends on the authentication method:
- Digest (recommended): Sign in at portal.telxi.com → SIP trunks, select your trunk, and copy the SIP username and SIP password. Nothing else is configured on the Telxi side — you'll enter these in Retell in Step 2.
- IP-based: Send the public signaling IP that Retell will use to support@telxi.com to be whitelisted. Telxi configures the whitelist for you; you then only need the SIP host (
sip.telxi.com) and port (5060) when configuring Retell. Retell sends from published IP ranges (18.98.16.120/30,143.223.88.0/21,161.115.160.0/19) — provide the relevant address(es) to support.
For inbound calls, give Telxi support the destination to route your DID to — Retell's SIP server sip:sip.retellai.com (append ;transport=tcp if inbound doesn't connect). Telxi sets up the routing on their side.
Import your number into Retell
- In the Retell dashboard, open Phone Numbers.
- Click + and choose Connect to your number via SIP trunking.
- Enter:
- Phone number — your Telxi DID in E.164 format (e.g.
+1XXXXXXXXXX). - Termination URI — your Telxi SIP server,
sip.telxi.com(port5060). - SIP username and SIP password — from portal.telxi.com → SIP trunks (Digest). Leave blank if using IP-based auth with Retell's IPs whitelisted by Telxi.
- Phone number — your Telxi DID in E.164 format (e.g.
- Save. The number now appears in Retell and can be assigned to an agent.
Assign an agent and test
- Assign the imported number to the agent that should handle its calls.
- Inbound test: call your Telxi DID from any external line; the call should reach your Retell agent. If it doesn't connect, check the origination routing with Telxi and append
;transport=tcpto the Retell SIP URI. - Outbound test: trigger an outbound call from the agent; confirm it routes through Telxi. If it fails, re-check the termination URI and credentials/whitelist.
Securing the connection (optional)
- Use TLS 1.2+ for SIP signaling:
sip:sip.retellai.com;transport=tls. SRTP media encryption requires TLS transport. - To validate Retell's server certificate over TLS, add Amazon's root CA to your trust store: G2-RootCA1.pem.
- mTLS (mutual TLS) is supported for enterprise setups; contact Retell support to enable it for your account.
Call transfer (SIP REFER)
Retell's transfer feature works over SIP trunking. For a cold transfer that shows the transferee's number as caller ID (SIP REFER), enable SIP REFER and PSTN transfer on your Telxi trunk.
Alternative: Dial to SIP URI
If elastic SIP trunking doesn't fit your setup, Retell also supports Dial to SIP URI: your system calls Retell's Register Phone Call API, receives a call_id, and dials sip:{call_id}@sip.retellai.com within 5 minutes. Note that Retell's built-in transfer feature is not available with this method (you implement transfers yourself). Most Telxi customers should use elastic SIP trunking above.
Troubleshooting
| Symptom | Likely cause | What to check |
|---|---|---|
| Inbound call doesn't connect | Origination wrong / SDP dropped | Confirm Telxi routes the DID to sip:sip.retellai.com; try ;transport=tcp. |
| Outbound call doesn't connect | Termination / auth issue | Verify termination URI sip.telxi.com and the SIP username/password, or that Retell's IPs are whitelisted. |
401 Unauthorized | Auth mismatch | Re-copy credentials from the portal, or have Retell's IP blocks whitelisted via support@telxi.com. |
| No audio after connect | Codec / encryption mismatch | Ensure PCMU/PCMA/G.722 enabled; for SRTP use TLS transport on both sides. |
| Transfer fails | SIP REFER disabled | Enable SIP REFER + PSTN transfer on the Telxi trunk. |
Reference
- Retell custom telephony: docs.retellai.com/deploy/custom-telephony
- Retell Twilio / Telnyx / Vonage SIP guides (use as references): docs.retellai.com/deploy/twilio
- Telxi portal: portal.telxi.com