How Outbound Calling Works — SIP Endpoint vs SIP Outgoing Trunk

Voice

This article explains how outbound calls are made through the @tomic platform — the two methods available, when to use each, and how to configure them.

Overview

Atom offers two methods for a customer's PBX to send outbound calls through the platform:

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SIP Endpoint	Username + Password (registration)	Hosted PBX, remote extensions, IP phones
SIP Outgoing Trunk	IP-based (no registration)	On-premise PBX with static IP

Both methods connect to Atom's gateway at sip3.atomcomm.com.

Method 1 — SIP Endpoint (Username/Password)

A SIP Endpoint allows a device or PBX to register with Atom using credentials. Once registered, it can make outbound calls.

When to use

Hosted or cloud PBX systems
Remote workers using softphones or IP phones
Situations where the customer's IP address is dynamic or unknown

How to create a SIP Endpoint in @tomic

Go to Voice → [Your Voice Service] → Manage
Click the Connections tab
Click Add Connection → choose SIP Endpoint
Fill in the fields:

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Connection Name	Friendly label (e.g., Head Office PBX)
Auth Name	The SIP username the PBX will use to register
VoIP Password	The SIP password for registration
Concurrent Calls	Maximum simultaneous calls allowed
Codecs	Preferred audio codecs (G.711, G.729, etc.)

Save — @tomic generates the credentials

How to configure the PBX

Point the PBX SIP trunk to:

Registrar / SIP Server: sip3.atomcomm.com
Auth Name: (from Connection settings)
Password: (VoIP Password from Connection settings)
Port: 5060 (UDP or TCP) or 5061 (TLS)

Once registered, outbound calls from the PBX are authenticated via the SIP credentials.

Call flow

PBX dials number → SIP INVITE sent to sip3.atomcomm.com → Atom authenticates using Auth Name + Password → Atom routes call to PSTN → call connects

Method 2 — SIP Outgoing Trunk (IP Authentication)

A SIP Outgoing Trunk uses IP whitelisting instead of credentials. Atom accepts calls from the customer's known IP address without requiring registration.

When to use

On-premise PBX with a static public IP
Situations where registration is not supported or not desired
High-call-volume environments (no registration overhead)

How to create a SIP Outgoing Trunk in @tomic

Go to Voice → [Your Voice Service] → Manage
Click the Connections tab
Click Add Connection → choose SIP Outgoing Trunk
Fill in the fields:

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Connection Name	Friendly label
Primary IP	The public IP of the customer's PBX
Secondary IP	Failover IP (optional)
Concurrent Calls	Maximum simultaneous calls allowed
Codecs	Preferred audio codecs

Save

How to configure the PBX

Point the PBX outbound SIP trunk to:

SIP Server / Proxy: sip3.atomcomm.com
Port: 5060
No registration required — the PBX sends calls directly; Atom identifies it by source IP

Ensure calls leave from the same IP registered in @tomic. If the PBX is behind NAT, ensure the NAT IP matches.

Call flow

PBX dials number → SIP INVITE sent to sip3.atomcomm.com from whitelisted IP → Atom authenticates by source IP → Atom routes call to PSTN → call connects

Caller ID on Outbound Calls

Atom uses the DID assigned to the Voice Service as the outbound Caller ID by default. To control this:

Ensure a DID is assigned to the Voice Service (Assigned DIDs tab)
The PBX can pass a specific CLI in the SIP From/P-Asserted-Identity header, subject to Atom's CLI validation rules
Sending an unverified CLI may cause the call to be rejected or presented as "Unknown"

Comparing the Two Methods

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Registration required	Yes	No
Credentials	Username + Password	IP address
Dynamic IP support	Yes	No (static IP required)
Setup complexity	Low	Low
Security model	Credential-based	IP whitelist