SIP Trunking | Atlas | Documentation

SIP trunking is the bring-your-own-carrier option for phone numbers. Instead of Atlas provisioning a number for you through Twilio, you keep your number with your existing carrier and point that carrier’s SIP trunk at the Atlas platform. Calls travel over your trunk, and transfers go out to SIP URIs instead of dialed phone numbers.

Use it when you already own numbers you can’t move, when your carrier contract or compliance rules require you to keep the trunk, or when you want calls to stay inside a PBX you already run.

SIP trunking is rolled out per account. If you don’t see the SIP Trunking option in the phone number modal, contact support to have it enabled.

Network requirements

Atlas sends SIP signalling and media from a fixed set of IP addresses. Allowlist them on your firewall, SBC, or PBX so traffic can pass between Atlas and your trunk, in the form:

198.51.100.10/32
198.51.100.11/32

The addresses above are examples. Ask support for the current Atlas SIP addresses for your account — they carry SIP traffic only.

What you’ll need

Collect the following from your carrier or PBX administrator before you start:

What	Notes
SIP phone number	The DID that terminates on your trunk, in full E.164 format.
SIP address	The hostname of your SIP server, for example `sip:pbx.example.com`.
SIP IP address	The IP that hostname resolves to. Your carrier may need the Atlas IPs allowlisted on their side too.
Outbound username	Digest authentication username for outbound calls.
Outbound password	Digest authentication password for outbound calls.

A complete set looks like this:

Number:    +17785550123
SIP:       sip:pbx.example.com -> 203.0.113.10
Username:  dd943127ae
Password:  a7fcb7b2f6k7c187

IP-based authentication only works reliably when your carrier gives you a dedicated termination URI or SIP server. On a shared endpoint, supply outbound authentication credentials instead.

Connect your trunk

Open the phone number step

Go to the Integrations step of the campaign wizard, or the Integrations step during onboarding, and start adding a phone number.

Choose SIP Trunking as the provider

In the phone number modal, pick SIP Trunking from the provider list instead of Twilio.

Enter your trunk details

Fill in the fields below and save. Atlas registers the number against your trunk and links it to the campaign.

Field reference

Field	Required	Description
Phone number	Yes	The DID that terminates on your trunk, in E.164 format (for example `+14155550123`).
Termination SIP URI	Yes	The gateway address Atlas sends signalling to — the hostname or IP your carrier gave you (for example `sip.example.com`).
Outbound auth username	No	Digest authentication username for outbound calls. Leave blank only if your carrier authenticates by IP.
Outbound auth password	No	Digest authentication password for outbound calls. Leave blank only if your carrier authenticates by IP.

Once saved, the number appears wherever your other numbers do, labelled SIP Trunking. Twilio-only features — A2P 10DLC registration and TrustHub verification badges — don’t apply to SIP numbers and are hidden for them.

Call transfers over SIP

Campaigns on a SIP trunk use a different transfer form than Twilio call forwarding: destinations are SIP URIs rather than dialed numbers, so the agent hands the call back onto your trunk instead of placing a new outbound call.

Each transfer destination is one row:

Field	Required	Description
Who	Yes	The label the agent uses for this destination, such as `Sales` or `Billing`.
SIP URI	Yes	Destination in `sip:user@host` form, for example `sip:sales@pbx.example.com`. Each URI can only be used once per campaign.
Phone number	No	Carried alongside the URI for your own records.
Transfer message	No	Played to the caller during the transfer. Falls back to the campaign’s shared forwarding message.
SIP headers	No	Custom headers sent with the transfer. See below.

Add a row per destination with Add another transfer, then save. You can also import destinations from another campaign — only rows that already have a SIP URI are brought over.

Saving replaces every transfer destination on the campaign: the existing rows are deleted and the ones on screen are re-created. Don’t save a partially filled form expecting your old rows to survive.

SIP headers

SIP headers are free-form key/value pairs attached to the transfer, useful for passing context to your PBX or carrier — a queue name, an account reference, a routing hint. Add as many as your destination expects; Atlas sends them exactly as written, so match the header names your PBX routes on.

Troubleshooting

Symptom	Likely cause
The SIP Trunking option isn’t in the provider list	The feature isn’t enabled on your account — contact support.
Calls don’t reach the agent	Your carrier isn’t routing the DID to the Atlas gateway, the termination SIP URI is wrong, or the Atlas SIP IPs aren’t allowlisted.
Calls are rejected at setup	Authentication mismatch — your carrier expects digest credentials but the fields were left blank, or the source IP isn’t allow-listed on a shared endpoint.
A transfer destination won’t save	The SIP URI isn’t in `sip:user@host` form, or it duplicates another row on the campaign.
Transfers connect but land in the wrong place	The destination URI or its SIP headers don’t match what your PBX routes on.