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:
Method | Authentication | Best For |
|---|---|---|
| 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
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:
Field | Description |
|---|---|
| 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:
sip3.atomcomm.comOnce 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
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:
Field | Description |
|---|---|
| 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:
sip3.atomcomm.comEnsure 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:
Comparing the Two Methods
SIP Endpoint | SIP Outgoing Trunk | |
|---|---|---|
| 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 |
Checklist — Outbound Calling Setup
SIP Endpoint:
SIP Outgoing Trunk: