Voice

Understand when Telstra may screen calls with an IVR and what callers need to do to connect.

Identify what a SIP response means, from call setup and redirection to request, server, and global failures.

Check the exact per-minute rate for a number before you call, including international destinations.

Connect your switch or device to Atomic SIP and configure inbound DID delivery with the correct server details.

Use mobile caller location codes to route inbound calls to the right regional queue or IVR.

Present the original caller’s number on forwarded calls while meeting Atom’s SIP trunk requirements.

Format caller and called numbers, choose which CLI to send, or block specific numbers on your SIP trunk.

Enable voicemail and check messages by calling 98 or receiving them as email attachments.

Adjust the number of channels assigned to a Voice service account in the Atomic portal.

Create a voice service and set its plan, call channels, dial format, and calling options.

Route incoming SIP trunk calls to backup numbers when the service is unavailable, busy, or unregistered.

Choose when SIP trunk media is relayed through the Atomic platform for NAT handling or network topology masking.

Configure the required SIP headers so anonymous and private calls connect successfully.

Connect SIP phones and ATAs to your account using username-and-password authentication.

Keep callers hearing ringback while a SIP call is being set up, even when their phone cannot generate the tone.

Connect SIP devices and trunks so your service can make and receive calls through Atomic.

Match a failed call’s CDR result code to its cause and find the next troubleshooting step.

Connect Yeastar Cloud PBX to your SIP trunk and route inbound and outbound calls with the right DID and caller ID settings.

Understand how SIP Trunks handle PRACK acknowledgements and find the P-Asserted-Identity reference.

Configure SIP connections, route inbound DID calls, and set the voice service options your PBX needs.

Offer customers a branded cloud PBX with flexible call handling, voicemail and reporting from the Atomic platform.

Give customers a branded portal to manage Hosted Voice users and extensions, with the option to use your own domain.

Create a Hosted Voice tenant, configure calling and licenses, and give users a branded portal.

Give customers a branded SIP domain for connecting their SIP phones and softphones.

See how Hosted Voice licenses determine channel capacity and how many simultaneous calls each user can handle.

Set the correct timezone for each Hosted Voice account so time-based rules run at the right local time.

Enable 2-way SMS on Hosted Voice and assign a number to the right user or extension.

Choose your Hosted Voice country to get the right timezone and dial plan during setup.

See what’s needed to connect your Teams users and Microsoft 365 tenant with Atomic.

Check the subscriptions, licenses and administrator access you need to connect Microsoft Teams with Atomic.

Create a Teams Direct Routing account and find the domain needed to connect your Microsoft 365 tenant.

Verify your Teams Direct Routing domain with a TXT record and complete Microsoft 365 setup without enabling extra services.

Skip domain activation when setting up Teams Direct Routing; Microsoft no longer requires this step.

Connect your Microsoft Teams tenant to Atomic by configuring its SBC and resolving domain activation errors.

Assign an Atomic phone number to a licensed Teams user and enable calling through Direct Routing.

Route Microsoft Teams calls through Atomic by setting up voice routes and assigning a routing policy.

Convert dialled numbers to +E.164 and assign the right dial plan so Teams calls connect.

Set Teams calling options that help calls connect through the @voice platform.

Choose whether your Microsoft Teams calls show your DID number or appear as private.

Connect a phone number to your Teams Auto Attendant or Call Queue using a resource account.

Find an available phone number, assign it to your service, and track activation results.

Prepare your PBX and routing before number cutover to help minimise downtime for incoming calls.

Choose the number format your switch, PBX, or device receives for incoming calls and sends for outgoing calls.

Forward calls from your hosted number directly to a designated phone number.

Move your number to a new provider with the right account details and port-out steps.

Keep number ports authorised with a customer-signed PAF and download the generated copy from pending requests.

Route your Atomic DID to one or more Asterisk extensions with a simple dialplan update.

Move your US or Canadian DID numbers to Atom by submitting the required customer and carrier details.

Port your numbers into the right service and connection, with clear requirements and expected timelines.

Prepare your simple number for a CAT-A port and know what to expect from submission through cutover.

Prepare CAT-C number ports, track verification and cutover, and check your numbers after transfer.

Manage DID connections, caller ID rules, and number formats individually or in eligible bulk updates.

Move a DID to another Service or Connection and see when the change takes effect.

Connect Genesys Cloud to your Atomic SIP Trunk and enable secure two-way SMS.

Connect your PBX, voice AI, or contact-centre platform to Atomic for inbound and outbound calls.

Connect your Atom number to an ElevenLabs Agent for inbound and outbound calls.

Connect your Atom number to a LiveKit AI agent and test inbound and outbound calling.

Connect your Atom number to a Vapi assistant and test inbound and outbound calls.

Configure your SIP interconnect with the correct gateway details, codecs, dialing plans, and security settings.

Set your SIP Trunk to Domestic dial format to make calling local numbers easier.

Choose the caller ID shown on outbound SIP Trunk calls, or set a unique CLI on each user device.

Configure local calling translations so 8-digit and 13 numbers route correctly for your area.

Get notified of incoming SMS on your iPhone and reply using your Virtual Mobile Number.

Configure your Voice connection correctly so you can assign or port a number.

Check how channels affect concurrent calls and manage your account’s channel count in the portal.

Correct the business name recipients see when you call from your DID by updating caller ID provider records.

Know why calls to iPhones may be screened or sent to voicemail, and what callers can do to reach the recipient.

Disable SIP ALG on your router to help prevent one-way audio, failed calls, and other VoIP connection issues.

Set up caller identity on outbound calls and improve compatibility with carrier networks.

Choose the right SIP connection and configure your PBX to send outbound calls through Atom.