Connecting FreePBX to Telnyx — SIP trunk setup walkthrough

Telnyx is one of the most popular SIP-trunk carriers for FreePBX® deployments — API-first, credit-based pricing, strong US/Canada coverage. This article walks the full setup: creating a SIP Connection in the Telnyx portal, configuring the FreePBX trunk, adding outbound and inbound routes, and the first test call. Assumes you've already followed Configuring a new FreePBX install and have extensions registered.

1. Sign up for Telnyx and load credit

Create an account at telnyx.com. The free trial gives you a small credit; for production usage you'll need to add a payment method. Telnyx is post-paid for established accounts, prepaid until then.

You'll also need to complete the Regulatory Profile under Account → Regulatory Requirements before you can buy DIDs. Takes 5 minutes; required by FCC for US numbers.

2. Create a SIP Connection

In the Telnyx portal: Voice → SIP Connections → Add SIP Connection.

• Connection Type: Credentials (simpler than IP auth and works behind NAT). IP auth is an option if you have a static IP — slightly more secure since no password to leak, but less convenient.
• Connection Name: anything memorable, e.g. lylix-freepbx
• Save. Telnyx generates a username and asks you to set a password (long random string — use a password manager).

Once created, click the connection and note these settings for later:

• Username
• Password
• SIP URI (something like sip.telnyx.com)

Connection settings to verify

On the SIP Connection's Settings tab:

• Outbound Voice Profile: create one if you don't have one (Voice → Outbound Voice Profiles → Add). Set a daily/monthly spending limit so a misconfiguration can't run away with your bank account.
• SIP Region: pick US East 1 (Ashburn) — same US East Coast location as LYLIX, lowest latency.
• DTMF Type: RFC 2833 (default).
• Codec preferences: G.711 (ulaw for US, alaw for EU). Skip G.729 unless you really need bandwidth savings and your customers don't mind paying the licensing.

3. Buy a DID number

In the Telnyx portal: Numbers → Search & Buy Numbers. Pick a region, area code, search, buy. Once you own the number, go to Numbers → My Numbers, click the DID, and on the Voice tab set the Connection to the SIP Connection you created.

This is the routing: inbound calls to this DID get sent over SIP to the FreePBX server registered against the connection.

4. Add the trunk in FreePBX

In FreePBX: Connectivity → Trunks → Add Trunk → Add SIP (chan_pjsip) Trunk.

General tab:

• Trunk Name: telnyx
• Outbound CallerID: "Your Company" <15551234567> using your Telnyx DID in E.164 format (no plus, 11 digits for US/Canada)

pjsip Settings → General tab:

• Username: the Telnyx connection username
• Secret: the connection password
• Authentication: Outbound
• Registration: Send
• SIP Server: sip.telnyx.com (or the region-specific one Telnyx shows in the connection details)
• SIP Server Port: 5060

pjsip Settings → Advanced tab:

• From Domain: sip.telnyx.com
• From User: your Telnyx connection username
• DTMF Mode: RFC 4733
• Match (Permit): leave blank unless using IP auth, in which case put the Telnyx peering IPs here

Click Submit, then Apply Config.

5. Verify trunk registration

Wait ~10 seconds, then check Reports → Asterisk Info → PJSIP Endpoints. Your telnyx trunk should show Available with the Telnyx SIP server's IP.

If it shows Unavailable, the most common fixes:

• Username/password mismatch — copy/paste again, check no leading/trailing whitespace.
• Firewall — your FreePBX firewall might be blocking Telnyx's IPs. Telnyx publishes its peering ranges; add them to FreePBX's Connectivity → Firewall → Networks as Trusted.
• NAT — if your VPS has a private IP behind a NAT (rare on LYLIX VPS — yours has a public IP), Telnyx sees one IP, the SIP packet says another. LYLIX VPSes don't have this problem; if you do, check your Settings → Asterisk SIP Settings → External Address matches your public IP.

6. Add an outbound route

Connectivity → Outbound Routes → Add Outbound Route:

• Route Name: telnyx-us
• Route CID: leave blank (uses trunk default)
• Trunk Sequence: telnyx

Dial Patterns tab — patterns are what FreePBX sends to the trunk. For US/Canada 10-digit dialing with auto-add of the leading 1:

prepend	prefix	match pattern
1		NXXNXXXXXX
		1NXXNXXXXXX

This lets users dial either 5551234567 or 15551234567 and both work. For international, add patterns starting with 011 or use E.164 (+ + country code).

Submit, Apply Config.

7. Add an inbound route for your DID

Connectivity → Inbound Routes → Add Inbound Route:

• Description: anything descriptive
• DID Number: 15551234567 (E.164 without the +, matching what Telnyx sends)
• Set Destination: pick an extension, IVR, or ring group

Submit, Apply Config.

8. Test outbound, then inbound

From your registered softphone:

• Outbound: dial your mobile. You should hear ringback, then your mobile rings, then two-way audio.
• Inbound: call your DID from your mobile. The destination extension should ring.

Watch Reports → Asterisk Logfiles live while testing. Useful Telnyx response codes:

• 200 OK — call established. Good.
• 403 Forbidden — auth failure or DID not owned.
• 404 Not Found — Telnyx doesn't recognize the destination number (check the dialed digits).
• 503 Service Unavailable — Telnyx-side issue, usually transient.
• 603 Decline — call declined; often a regulatory or fraud-detection rejection (international, premium-rate).

Recommended Telnyx-side hardening

• Spending limit on the Outbound Voice Profile. Hard cap your monthly exposure.
• Geographic restrictions on the profile — block calls to high-toll-fraud destinations (Cuba, North Korea, satellite ranges, premium-rate numbers).
• IP auth instead of credentials if you can — eliminates the password-leak risk. Requires your VPS to have a static IP (which LYLIX VPSes do).
• SIP TLS instead of UDP — encrypts signaling in transit. Set it up after you have a Let's Encrypt cert on the FreePBX server.

If something doesn't work

Telnyx has a useful Debugging → SIP Call Flow tool that shows the SIP exchange for a recent call — INVITE, BYE, provisional responses, the whole thing. Filter to the time of your failed test call; you'll usually see exactly where the negotiation broke down.

If you're still stuck, open a LYLIX ticket with the test date/time and we can dig into the FreePBX side. Telnyx's own support is fast and good for carrier-side questions.

FreePBX®, Asterisk®, and FreePBX modules are registered or common-law trademarks of Sangoma Technologies Corporation; LYLIX hosts FreePBX and Asterisk but is not affiliated with Sangoma. Telnyx is a trademark of Telnyx LLC.