Connect a VoIP provider

Connect a provider you already use (Twilio or Telnyx by API key; SignalWire, Vonage, Plivo or any SIP trunk over SIP), or open a new Telnyx account with guided setup.

Production traffic should run on numbers you own through your own VoIP provider. The dashboard's Connections → VoIP page (/dashboard/voip-connections) connects your provider account; once connected, you import numbers that support inbound and outbound calling.

Click New connection and pick one of two routes: Connect a provider you already use, or Set up a new Telnyx account.

Connect a provider you already use#

Pick the provider your numbers are with:

For Twilio and Telnyx, paste the credentials and click Connect & Test. ThunderPhone verifies them and configures the provider side for you. For Telnyx, ThunderPhone adds one SIP connection of its own to your Telnyx account (named thunderphone-production-<org>-outbound-<id>) and places outbound calls through it, so Telnyx bills those calls to you. It uses the outbound voice profile of the Telnyx connection your numbers are on, so that profile's allowed countries and spend limits apply. If that connection has no outbound voice profile and your account has exactly one enabled profile, ThunderPhone uses that one. To use a different one, assign it to the connection in the Telnyx portal and save the ThunderPhone connection again. If the account has no enabled profile, or several, assign one to the connection in the Telnyx portal and save the ThunderPhone connection again. Until then, outbound calls fail with outbound_trunk_unavailable; inbound calls keep working.

When Telnyx refuses an outbound call (for example SIP 403 Forbidden), the refusal comes from your own Telnyx account. The usual causes are no available credit, a caller number that is not on that Telnyx account, or an outbound voice profile that does not allow the destination country.

SIP setup for SignalWire, Vonage and Plivo#

These three open the manual SIP form with the steps for that provider at the top, a link to the provider's own setup guide, and an example SIP domain:

• SignalWire: create SIP credentials for outbound calls, enter the SIP domain, username and password, and give each number a SWML script that connects incoming calls to the ThunderPhone SIP endpoint.
• Vonage: in the SIP Trunking dashboard, create a trunk with outbound credentials, enter its domain (for example your-trunk.sip-us.vonage.com) and the credentials, add the ThunderPhone SIP endpoint as the trunk's inbound SIP URI, and link your numbers to the trunk.
• Plivo: create an outbound trunk with a credential and enter its Termination SIP Domain (<trunk-id>.zt.plivo.com), username and password; create an inbound trunk whose Primary URI is the ThunderPhone SIP endpoint followed by ;transport=tcp; then set each number's Application Type to that trunk.

The result is a manual SIP connection (provider: "manual" in the API), tested and verified exactly like one entered by hand. The connection card shows the provider's name, for example SignalWire · SIP.

Set up a new Telnyx account#

No provider yet? The guided flow walks you through opening a Telnyx account and connecting it:

1. Create the account

   Click Open Telnyx Signup, create your account, and check "I've created my account."

2. Get an API key

   Click Open API Keys Page, generate a key, and paste it into API Key (v2). Optionally name the connection.

3. Connect & Test

   Click Connect & Test — ThunderPhone verifies the credentials and provisions what it needs on the Telnyx side before saving.

What ThunderPhone sets up in Telnyx#

Unless you enter your own Connection ID, both Telnyx routes (guided setup and pasting an API key) leave the same thing in your Telnyx account: one SIP connection named thunderphone-<your organization id>. It is an FQDN connection whose FQDN record points at ThunderPhone's SIP endpoint, and every number you import is assigned to it. Telnyx sends calls for those numbers straight to ThunderPhone over SIP.

• Its webhook fields are empty on purpose. No webhook, Call Control application, TeXML application or registered SIP phone is involved.
• Don't move imported numbers to another connection or application, and don't register a device against it.
• If you entered your own Connection ID when connecting, ThunderPhone uses that connection instead and never changes it; it must be an FQDN connection pointing at the ThunderPhone SIP endpoint.

Before August 30, 2026, the thunderphone-… connection ThunderPhone created was a Credential connection, which cannot receive calls. If yours still shows as a Credential connection, contact support and we will convert it.

Manual SIP configuration#

The SignalWire, Vonage and Plivo tiles open this form, and so does I'll enter every setting myself under Other provider. Any SIP trunk works, including DIDWW or a PBX you run yourself. Enter the connection name, SIP domain / URI, optional username and password, port, and transport; Advanced Settings adds media encryption (SRTP). The form also shows the SIP address your trunk must send inbound calls to: sip.thunderphone.com, on port 5060 for UDP or TCP, or 5061 for TLS. Over TLS, the SIP signaling stays encrypted all the way to LiveKit Cloud, where ThunderPhone handles the call; audio never passes through sip.thunderphone.com, so SRTP works as before. Address each call to the dialed number in E.164 form, with the same number, including the +, in both the request line and the To header: for example sip:+14155550142@sip.thunderphone.com. The + is what lets ThunderPhone handle the call in the right region (see the note under IP allowlisting). Trunks set up earlier with a <your-project>.sip.livekit.cloud address keep working while Allowed Source IPs is empty; you don't need to change them. ThunderPhone turns those settings into a dedicated pair of SIP trunks on LiveKit Cloud: outbound calls from the connection's numbers are placed to your SIP domain with your credentials, and inbound calls to those numbers are accepted from any address unless you fill in the optional Allowed Source IPs field. Only fill it in if your carrier publishes fixed signaling IPs: a wrong or incomplete list makes ThunderPhone reject your carrier's incoming calls. Two buttons do the heavy lifting:

• Generate AI Setup Guide — writes provider-specific instructions for the trunk-side configuration you need.
• Test & Save Connection — checks that the SIP endpoint resolves to a public address and, for TCP/TLS, accepts a connection on the chosen port (UDP hosts get the DNS check only), then saves. It cannot place a call yet because the connection has no numbers; the real proof comes at import time (below).

Editing a connection later re-runs the same check on your changed settings before anything is saved, then rebuilds the trunk and re-verifies every number on the connection with a short test call in the background — the numbers show as Provisioning until their test call succeeds. Changing only Allowed Source IPs on a Connected connection is different: it never affects outgoing calls, so ThunderPhone updates the list on the inbound trunk right away and the numbers stay Active. Their incoming call test resets to not tested; run it again, or place a real call through your carrier, to confirm the new list lets calls in. The test fails when its call would be refused under the list. Stored passwords are never shown back; leave the password blank to keep it, or tick "Remove the stored password" for an IP-authenticated trunk.

IP allowlisting and trunk authentication#

Some carriers (DIDWW, Telnyx IP-auth connections, most PBXs behind a firewall) only accept SIP traffic from addresses you allowlist, or authenticate outbound calls by source IP. ThunderPhone uses two sets of addresses.

SIP edge. Inbound calls you send to sip.thunderphone.com go through ThunderPhone's SIP edge, which carries SIP signaling only. The first INVITE goes to sip.thunderphone.com; the rest of the call's signaling (ACK, hang-up, transfers) runs between your carrier and the edge server that took the call, sip-1 or sip-2. If your firewall filters SIP signaling, allow all three, on UDP and TCP port 5060 and TLS port 5061:

LiveKit Cloud. The audio (RTP) of every call, and the SIP signaling of outbound calls, run between your carrier and LiveKit Cloud, never through the SIP edge. Allow every range below, for both SIP signaling and media:

LiveKit applies these ranges to Realtime, SIP signaling and media and Webhooks, and they cover calls handled in its Canada, EU, India, Japan and US regions. Region names for a pinned endpoint: canada, eu, india, japan, us.

A few rules of thumb for the trunk itself:

• ThunderPhone does not register. It never sends SIP REGISTER to your carrier, so configure the trunk for IP-based and/or credential (username + password) authentication, not registration.
• Outbound calls should use credentials. Enable username/password authentication on the carrier side and enter the same values in the manual SIP form. Treat the IP allowlist as a second layer, not the only one — outbound calls to destinations outside the covered regions may originate from other addresses.
• Inbound is the reverse direction. The optional Allowed Source IPs field in the form is where you enter your carrier's signaling addresses so ThunderPhone only accepts calls from them. Enter IP addresses or CIDR ranges; hostnames never match. Leave it blank unless your carrier publishes a fixed list. ThunderPhone checks the list on every inbound call against the address its SIP edge sees, so once the list is set, send your calls to sip.thunderphone.com. A call from an address not on the list, or one sent straight to a <your-project>.sip.livekit.cloud address, is refused: it fails or connects and ends straight away.

The ranges are published by LiveKit at Regions, regional endpoints, and static IPs; ThunderPhone checks that page weekly and updates this guide when it changes.

After connecting#

Import numbers from Phone Numbers → Add number → Connect VoIP provider (or straight from the connection). Each imported number is verified before it goes active. For a manual SIP connection that verification places a short (a few seconds) test call from your number, through your SIP trunk, to a ThunderPhone number — so the username, password, transport and your carrier's outbound routing are all exercised for real. Your carrier bills that call like any other. If the carrier rejects it, the number stays in Provisioning and the phone-numbers page shows the carrier's answer (for example "rejected ThunderPhone's credentials (SIP 401)"); fix the connection and verify again. If your carrier accepts the trunk but its routing plan does not cover ThunderPhone's test number (DIDWW answers "Destination number blacklisted by routing plan"), the number shows as Verified · carrier accepted: it is usable, but calls to other destinations were not tested. That call proves your trunk can place calls as the number; it is not proof that you own the number, and it does not test calls to the number; the optional incoming test below does. Verified numbers support inbound and outbound — which unlocks outbound campaigns and programmatic outbound calls.

Incoming call test#

Verify number opens a dialog that can also test incoming calls. The test places a call to your number, checks that the call reaches ThunderPhone, then hangs up without starting your agent:

• Numbers with their own outbound SIP trunk (manual SIP numbers, and Twilio numbers once ThunderPhone has set up outbound calling in your Twilio account) first call themselves: ThunderPhone dials the number from the number, through that trunk. The dialog tells you before it places the call. The call goes out through your carrier and, if your inbound routing is right, comes straight back to ThunderPhone. When it comes back under the number's own caller ID, that one call proves both that your carrier accepts ThunderPhone's calls as the number and that calls to the number reach ThunderPhone, in any country. Some carriers refuse or don't connect a call from a number to itself; that says nothing about your routing, so ThunderPhone then calls from one of its own numbers instead. If your number forwards somewhere else, that phone may ring twice: first showing the number's own caller ID, then ThunderPhone's test number.
• Other numbers (and the fallback above) are called from one of ThunderPhone's own numbers. ThunderPhone can only place these calls to US and Canadian numbers for now; for other numbers the test reports that it couldn't run, with your carrier's answer if the self-call was refused.

If your carrier changes the caller ID on the way, ThunderPhone can't recognise the call, so your agent may answer it for a moment. The incoming test is optional. Skip it and the number still works; the phone-numbers page then shows Incoming not tested, and you can run it later with Test or the number's Test incoming calls menu item. Right after an import, the phone-numbers page also asks you to test incoming calls on the new numbers. Nothing is dialed until you confirm in the dialog.

If your provider sends the number somewhere else, whatever it points at may ring for up to 8 seconds per call before ThunderPhone hangs up. If something answers first (a Twilio app that answers and then forwards, a phone menu), the test waits for the forwarded call to reach ThunderPhone: up to 8 seconds on the self-call and 15 seconds on a call from a ThunderPhone number. The result stays on the number (Incoming passed or failed) until you change the connection or re-import the number, which resets it to not tested. Changing a connection re-verifies outgoing calls in the background but never calls your number unasked.

Troubleshooting inbound calls#

The connection's Test button checks credentials, and verification checks outbound dialing. To check inbound, run the incoming call test above, or call the number from a phone and look for the call in Call History. Browser Talk tests and bot-to-bot simulations don't go through your carrier, so they don't show whether inbound works. A simulation that targets the phone number (SIP loopback) does dial it through your carrier, so an answered one does exercise inbound routing.

• You hear your own mobile carrier's message (for example "Welcome to Verizon" or "the number you have dialed is not in service"): the call failed before it reached ThunderPhone. Check that the number is active in your provider account and assigned to the ThunderPhone connection or trunk.
• Busy signal or the call rings out: the provider accepted the call but could not deliver it. With Telnyx, check the number's Connection in the Telnyx portal: it should be the thunderphone-… FQDN connection described above, or your own FQDN connection pointing at the ThunderPhone SIP endpoint. With Twilio, the number should be attached to the ThunderPhone SIP trunk rather than a Voice webhook or TwiML app.
• Your carrier's logs show 404 No trunk found (manual SIP): the call reached ThunderPhone but matched no trunk. Check that the number is imported on this connection and that the trunk points at sip.thunderphone.com. If it points at a <your-project>.sip.livekit.cloud address, Allowed Source IPs must be empty: with a list set, only calls sent to sip.thunderphone.com are accepted.
• 404 Not Found from sip.thunderphone.com (manual SIP): the call was not addressed to a phone number, or the request line and the To header name different numbers. Send the dialed number in both, as above.
• 403 Forbidden from sip.thunderphone.com: the request line names another host. Address the call to sip.thunderphone.com, for example sip:+14155550142@sip.thunderphone.com.
• The call reaches ThunderPhone but the wrong agent answers, or nobody does: check the number's inbound agent on Phone Numbers.

The in-app copilot can check a number for you: ask it why calls to the number fail. For Telnyx connections it also looks up, read-only and for organization admins, which connection Telnyx assigns the number to.

Next steps#