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Wixzel Phone does not sell phone numbers or minutes. You connect a SIP trunk from a carrier you choose, and your carrier bills you directly at your rates. That is a deliberate trade. It means one more step before your first call, and it means we are never between you and your telephony provider — no markup, no resale margin, and no reason for us to care which carrier you pick.

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The IP to allowlist

Calls are placed from 95.216.218.102.Nearly every carrier authorises outbound calls by source IP, so this address has to be on your trunk’s allowlist before anything will connect. Do this first: a trunk that is correct in every other respect still fails without it, and the failure looks like a fault at our end rather than a missing permission.Every trunk returns it as platform_ip, so you can read it from the API rather than trusting this page to stay current:

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Where inbound calls should be sent

Send inbound calls to sip:95.216.218.102:5090.This is the other half of a trunk, and it is separate configuration in a different part of your carrier’s console from the allowlist above. The allowlist tells them to accept calls from us; this tells them where to deliver calls to us. Twilio calls the field the Origination URI; other carriers call it a termination point, a destination SIP URI, or just the host. It is the same string.Getting this wrong fails in the quietest way this product has: you buy a number, point it at an agent, and it simply never rings. Nothing appears in your logs or ours, because the call never arrives — the carrier had nowhere to deliver it.Every trunk returns it as origination_uri, so read it from the API rather than trusting this page to stay current:

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What you need

Any SIP provider will do. From them you need:
  • a host — sip.carrier.example
  • credentials — username and password, if they use registration
  • your numbers, and whether inbound calls are delivered to that trunk
  • an IP allowlist entry for 95.216.218.102, or a credential list if you would rather authenticate with a username and password

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Connecting it

transport is udp, tcp or tls.
The password is encrypted at rest and is never returned by the API again — including to you. Store it wherever you keep the rest of your carrier credentials.

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Registering a number

A number must belong to a trunk before it can place a call:
Give it an inbound_agent_id and calls to that number are answered automatically by that agent.

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Carrier-side setup

Your carrier needs to know where to send traffic, and — more often the cause of a silent failure — needs to agree that you are allowed to place the call at all. Work through all of these before blaming the API:
1

Allowlist the Wixzel Phone IP

Wixzel Phone authenticates to your carrier by IP unless you gave the trunk a username and password. If the IP is not on your carrier’s allowlist, your INVITE is rejected before it is ever rated, and nothing appears on your bill to hint at why.Calls originate from 95.216.218.102. Add that to whatever your carrier calls its IP access control list, and see the carrier notes below for where that setting lives on the common providers.
This is the single most common reason a trunk that works elsewhere does not work here. A trunk configuration that is correct in every other respect — same host, same port, same credentials — still fails if it was allowlisted for a different server. Moving between deployments means updating the allowlist at every carrier.
2

Verify the caller ID

The number in from_number_id has to be one your carrier has authorised you to present. Carriers routinely accept the call, log it, then drop it when the presented caller ID is not provisioned on the account.
3

Complete KYC and any destination registration

Some markets gate outbound calling behind identity checks. In India in particular, an account that has not finished KYC — and, for many providers, DLT registration — can hold a positive balance and still have every outbound call cleared immediately. The portal shows the calls; it does not always show the reason.
4

Check the balance and the destination is enabled

A prepaid balance is necessary but not sufficient. Confirm the specific destination range is enabled on your account, not just that there is credit.

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Twilio

Twilio authorises outbound calls by source IP, and rejects anything else instantly.
1

Open the trunk's termination settings

Elastic SIP Trunking → your trunk → Termination → Authentication.
2

Add the platform IP to an IP Access Control List

Create or edit an IP ACL and add 95.216.218.102. A Credential List works instead, if you would rather authenticate with a username and password — put those on the trunk’s username and password fields and Wixzel Phone will use them in place of IP auth.
3

Check the Termination SIP URI matches the trunk host

The host on your Wixzel trunk must be the same your-domain.pstn.twilio.com shown in the console.
4

Confirm the caller ID is a number on that account

Twilio rejects a from_number_id it does not own.
5

Set the Origination URI, for inbound

Elastic SIP Trunking → your trunk → Origination, then add an Origination SIP URI of sip:95.216.218.102:5090 — or whatever GET /v1/sip-trunks reports as origination_uri.Outbound works without this; inbound does not, and does not complain. Attach your numbers to the trunk in the same console, then give the number an inbound_agent_id here so there is an agent to answer them.
A missing IP ACL entry shows up as an immediate rejection, roughly a second after the call is placed, with hangup cause 21 (Call Rejected). It does not ring and it does not time out.

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Wholesale carriers

Most non-Twilio carriers also authorise by IP, and additionally care about number formatting and account verification. If calls are accepted and then dropped rather than rejected outright, work through the troubleshooting below — particularly send_plus and the KYC step above.
An open SIP port is scanned for toll fraud within hours of appearing. Only endpoints matching a trunk you have configured are accepted — there is no anonymous endpoint — but keep your carrier’s own fraud controls on regardless.
Trunk changes take effect immediately: creating, updating or deleting a trunk rewrites the telephony configuration and reloads it, so there is nothing to restart.

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When a call does not connect

A call that is accepted by the API and then fails looks like this: POST /v1/calls returns 202 with status: "queued", and a later GET /v1/calls shows status: "failed", a short duration_seconds, and cost_micros: null. cost_micros: null is the important part. You are not charged for a call that never connected — no answer means no billable seconds, and the credit reserved at admission is released. A failed call costs you nothing, which also means the absence of a charge is not evidence of a problem on our side. Work down this list in order.
This is the single most informative step, and it is easy to skip. Your carrier’s portal will show whether the call reached them at all, what caller ID they saw, what destination they parsed, and how they cleared it.Three outcomes, and each points somewhere different:
  • The call is not listed. It never reached your carrier. The IP allowlist, the trunk host or the firewall is wrong.
  • Listed, answered, with a duration and a cost. The call worked at the telephony layer. If the caller heard nothing, the problem is media, not signalling — see the audio note below.
  • Listed, zero answered, zero cost, a short duration. The carrier accepted your INVITE and then cleared the call. This is a carrier-side permission problem far more often than a configuration one: caller ID not provisioned, KYC incomplete, or the destination not enabled. Nothing on the Wixzel Phone side can fix it, and the fastest route is your carrier’s support with the timestamp and the call id from their log.
400 number_not_on_trunk means the number has no sip_trunk_id. Calls are placed over your trunk, so a number that is not on one has nowhere to go.
If your carrier answers 100 Trying and then 408 Request Timeout a few seconds later, they could not route the destination. The most common reason is the leading +.Wixzel Phone dials sip:+15551234567@your-carrier by default, which is what Twilio and Telnyx expect. Many wholesale carriers cannot parse a + in the request URI: they accept the INVITE, fail to route it, and time out. The call shows in their portal as received and never answered, which looks like a fault at your end.Send bare digits instead:
Setting a dial_prefix implies this: a tech prefix is matched literally, so the + is always dropped when one is configured.
A 403 is a permission decision, usually about who you claim to be rather than who you are dialling. Check that the number behind from_number_id is one your carrier has authorised you to present.Carriers commonly reject a call with no caller ID at all. If you are testing from a softphone or an Asterisk CLI rather than through the API, confirm you are actually sending one — an INVITE with From: "Anonymous" earns a 403 from most carriers regardless of how well the account is configured.
A call that fails in roughly a second, with hangup cause 21 (Call Rejected) and no ringing, was refused by the carrier before it was routed. On Twilio this is almost always the Termination IP Access Control List: add 95.216.218.102 to it.The distinguishing feature is the timing. An unauthorised source is rejected immediately, because the carrier does not need to look anything up to decide. A call that is accepted and then fails several seconds later is a different problem — routing, formatting or account permissions — and the entries above cover those.This is worth checking first whenever the same trunk works from another server. Identical trunk settings do not carry the allowlist with them.
Normal Clearing (Q.850 cause 16) on a call that was never answered is not as reassuring as it sounds. It means the far end ended the call deliberately rather than erroring — a rejection can be dressed as a normal hangup. Read it as “something upstream chose to end this”, not as “the call completed”.A duration of a few seconds with no answer is usually the carrier clearing the call, not a person declining it. A human declining a call typically takes longer than the ring itself.
Not a failure — an absence. If the Origination URI is missing from your carrier’s console the call never leaves them, so there is nothing in our logs, no hangup cause, and no failed call to inspect. Outbound working is no evidence either way: the two directions are configured separately.Check the carrier has sip:95.216.218.102:5090 (or whatever GET /v1/sip-trunks reports as origination_uri) as the destination for that number, that the number is attached to the trunk on their side, and that it has an inbound_agent_id here so something answers it.
If calls connect but there is no audio, the usual cause is RTP not reaching the server — check that your firewall allows the media port range, not just the signalling port. Signalling on 5060 and media on the RTP range are separate holes in the firewall, and opening only the first produces exactly this symptom.