This is a detailed walkthrough of creating a voice service in @tomic. The portal supports several voice service types: SIP Trunk, Microsoft Teams Direct Routing, and Hosted Voice. The form fields change depending on which type you select.
Prerequisites
A Customer record must exist (see Walkthrough: )
You must have the relevant service plan available on your account
Navigation
Sidebar → Services → Add Service → click Voice
"Choose your service" selection screen with the Voice tile highlighted among service options
Step 1: Select Service Type = Voice
The form shows five icon-cards: Voice, NBN, Mobile, SMS and Data Pool. Click Voice. The voice sub-form loads below.
Step 2: Select a Customer
Table (Header Row)
Field | Required | Notes |
|---|---|---|
| Customer | Yes | Dropdown of all sub-customers. The customer must already exist — you cannot create one from this form. |
Step 3: Select Voice Service Type
After choosing a customer, a second dropdown appears:
Table (Header Row)
Option | Use case |
|---|---|
| SIP Trunk | Connect an IP-PBX, phone system, or softswitch to the Atom network for call termination/origination |
| Microsoft Teams Direct Routing | Connect Microsoft Teams Phone System to the PSTN via Atom |
| Hosted Voice | Full white-label cloud PBX (extensions, IVR, voicemail) — see Hosted Voice articles |
Add Voice Service page showing the Service Type dropdown appearing after a customer is selected
Step 4: Core Fields (all voice types)
Once the service type is selected, the following fields appear for all voice types:
Table (Header Row)
Field | Required | Definition |
|---|---|---|
| Service Plan | Yes | The billing/feature plan for this service. Options depend on your account and the service type selected. Check Products → Service Plan for the full catalog with pricing. |
| Service Name | Yes | A descriptive name that identifies the service. |
| Plan Assigned Channels | Read-only | The number of concurrent call channels included in the selected plan. You cannot edit this — it is determined by the plan. |
| Additional Channels | No | Extra channels to add on top of the plan’s included channels. Each additional channel allows one more simultaneous call. Leave at 0 if the plan’s channels are sufficient. |
| Total Channels | Read-only | Plan channels + Additional channels. This is the maximum simultaneous calls the service can handle. |
| Dial Format | Yes | The number format your PBX/device sends. E.164 = international format without + (e.g. 61212345678). Domestic = local format (e.g. 0212345678). Match this to what your equipment sends or calls will fail. |
| Block Service | No | Toggle ON to immediately block all outbound calls on this service. Use for credit holds or suspended accounts. |
| Domestic Only | No | Toggle ON to restrict this service to Australian calls only. International calls will be rejected. Recommended for standard business lines where international calling is not needed. |
| Enable CLI Masking | No | Toggle ON to strip the outbound caller ID so the recipient sees Private Number instead of your number. Used when customers do not want their number displayed. |
| Enable Call Forwarding | No | Toggle ON to allow call forwarding to be configured on this service. When enabled, an additional configuration section appears — see the Call Forwarding KB article for details. |
| Enable Voicemail | No | Toggle ON to enable a voicemail box for this service. Requires two additional fields: Email (where recordings are sent) and PIN (4–6 digits, used when calling *98 from the SIP client to check messages). |
Advanced Settings
Table (Header Row)
Field | Definition |
|---|---|
| CLI Translation | Select the DID number to present as the outbound caller ID. All DIDs assigned to this service appear in the dropdown. If no DIDs are assigned yet, the system uses whatever CLI your equipment sends — configure this after assigning DID numbers. |
| CLD Translation | A regex rule to transform the dialled number before it is sent to the PSTN. Used for prefix stripping or format conversion. Leave blank unless your PBX sends numbers in a non-standard format. |
| Max Session Time (ms) | Maximum call duration in milliseconds before automatic disconnect. Default is 3600ms (60 minutes). Set higher for conference bridges or support lines. |
| Generate Ring Back Tone | When enabled, the Atom platform generates an audible ringing tone to the caller while the called party’s phone is ringing. Enable this if callers hear silence instead of a ring tone. |
| Pass P-Asserted-Id | This is a required header in SIP Signaling messages when sending private or anonymous calling. More information can be found . |
Step 5: Save and connect
Click Save. The service is created immediately and you land on the Service Settings page.
What to do next:
- Go to Connections tab → set up SIP Endpoint or SIP Trunks so calls can be routed (see Walkthrough: Voice Service Configuration)
- Go to Assigned DID tab → assign phone numbers to this service (see Walkthrough: Request a New DID Number or Port a Number)
Notes on specific service types
SIP Trunk
Most common type. Your PBX/phone system connects to sip3.atomcomm.com using the service credentials. See Connection for Services in the KB for the full SIP connection setup.
Microsoft Teams Direct Routing
Connects Teams Phone System. The service plan must be a Teams Direct Routing plan. After creation, configure the Session Border Controller (SBC) connection under the Connections tab.
Hosted Voice
Full cloud PBX. See the Hosted Voice section of the KB for the complete setup guide — this is a multi-step process involving extensions, IVR, and user portal setup.