Whether you're setting up a phone system for the first time or upgrading an existing one, this guide covers everything you need to know before you buy — from choosing the right handset for your team to understanding gateways, DECT systems and expansion modules.
View Top Desk Phones Get Expert AdviceA business phone is only one part of a complete VoIP solution. The right choice depends on your provider, PBX platform, network, user roles, required mobility and the features your staff use every day.
These devices can look similar, but they are built for different calling environments.
A standard VoIP desk phone is a SIP-based device that registers to a phone system (hosted or on-premise) and handles voice calls. It works with virtually any VoIP provider or PBX platform.
If you need a dependable handset for a hosted VoIP service, 3CX or another SIP-based phone system, this is usually the correct category.
A UC phone (Unified Communications phone) goes further — it integrates voice calling with collaboration tools like presence, instant messaging, contact directories and sometimes video. UC phones are typically designed to work with a specific platform (such as Microsoft Teams or Zoom) and may require platform certification to access full features. If you just need reliable voice calls on a business phone system, a standard SIP phone is usually the right choice. If your team lives inside Microsoft Teams or another UC platform, a purpose-built UC phone gives you a better experience.
Standard SIP phones will not register to Microsoft Teams for calling. If your organisation uses Microsoft Teams Phone (formerly Teams Calling), you need a Teams-certified IP phone — one that has been specifically certified by Microsoft for the Teams platform.
Teams phones run a version of Android with the Teams app built in. They authenticate directly to your Microsoft 365 tenant and give staff a familiar Teams interface on their desk. They are not interchangeable with standard SIP phones — buying the wrong type is one of the most common mistakes in VoIP purchasing.
If you're unsure which licence or calling method you have, check with your Microsoft partner or IT administrator before purchasing hardware.
The phone is hardware only. You still need a service, credentials and a platform to manage business calling.
A VoIP phone is hardware only — it does not come with a phone number or a calling plan. To make and receive calls, you need a VoIP service provider (also called a SIP trunk provider or hosted PBX provider) who assigns you SIP credentials and connects your phones to the telephone network.
In Australia, popular business VoIP providers include Aussie Broadband, Vonex, VoIP Studio, Aircall, RingCentral, MyNetFone and many others. Your provider gives you a SIP username, password and server address that you enter into each phone (or your PBX) to register it.
A PBX (Private Branch Exchange) is the phone system that sits between your phones and the public telephone network. It manages internal extensions, call routing, voicemail, hunt groups, call queues, auto-attendants and more.
A traditional hardware PBX is being replaced by IP PBX systems — software-based or hardware appliance systems that run over your IP network. The most popular IP PBX in Australia for small-to-medium businesses is 3CX, which runs on-premise (on your own server or PC) or in the cloud. Other options include FreePBX, Asterisk, and various cloud-hosted platforms.
If you already have a PBX (such as 3CX, FreePBX or a hosted cloud PBX), your phones register to the PBX, which then connects to your SIP trunk. In this case, your provider credentials go into the PBX, not the individual phones.
Do you need a PBX? If you have two or more staff sharing a phone system with internal extensions, yes. A PBX handles the routing between extensions, manages voicemail, and connects your phones to a SIP trunk for outbound calls. Without a PBX, each phone would need its own direct SIP account with your provider — which is inefficient and expensive at scale.
Check how each phone will receive power and whether your network can prioritise voice traffic.
PoE (Power over Ethernet) means the phone draws its power through the same network cable that carries data — so there is no separate power adapter or power point required at the desk. The vast majority of business VoIP desk phones in this range are PoE-powered.
To use a PoE phone, your network switch must support PoE (IEEE 802.3af for standard devices, 802.3at/bt for higher-power devices). Most modern business-grade switches include PoE on some or all ports — check with your IT team or network administrator.
If your switch does not supply PoE: a compatible power adapter (5V DC for most Yealink and Grandstream models) is available as an add-on at checkout. It is important to use the correct adapter for your model — using the wrong voltage will damage the phone and void the warranty.
If you are deploying multiple VoIP phones and your existing switch does not support PoE, you will need a PoE switch before your phones will power up. A managed PoE switch also gives you the ability to prioritise voice traffic (QoS) over other network traffic, which directly affects call quality.
For small deployments (under 10 phones), an unmanaged PoE switch is usually sufficient. For larger or more complex deployments, a managed switch with VLAN and QoS support is recommended. DiGi Phone does not sell switches directly, but our team can advise on what specification to look for based on your phone count and network setup.
For reliable VoIP, one-way latency should generally remain below 150 milliseconds.
Variation in packet delay should generally remain below 30 milliseconds.
Packet loss should normally stay below 1 percent for clear and stable calls.
Managed switches can prioritise phone traffic and separate it from general network activity.
Screen size, programmable buttons, accessories and mounting options directly affect everyday usability.
The display affects how easy the phone is to use day-to-day — navigating menus, reading caller ID, managing multiple calls.
If your staff are on calls frequently or need to monitor multiple lines, a larger colour display reduces errors and speeds up call handling. For back-office or occasional-use desks, an entry-level screen is perfectly adequate.
Almost all business VoIP phones include a row of programmable soft keys or line keys along the side or below the display. These can be configured in your phone system to perform one-touch actions.
BLF (Busy Lamp Field) is the most common configuration — each key monitors a colleague's extension. When a colleague is on a call, their button lights red. When they're available, it's green. This lets receptionists manage incoming calls efficiently without asking "are you free?"
If you need more keys than your phone provides, an expansion module (covered above) is the correct solution.
An expansion module (also called a sidecar or DSS module) attaches to the side of a compatible desk phone and adds rows of programmable physical buttons. These buttons can be configured for:
Expansion modules are most useful for receptionists, attendants and call handlers who need to manage a large number of lines or monitor team availability without navigating menus.
Compatibility is critical: expansion modules are not universal. Each module is designed for a specific phone series. For example, the Yealink EXP40 is compatible with the T46S/T46G/T48S/T48G series, while the Yealink EXP50 suits the T5 series. Always confirm the module matches your phone model before ordering.
Most VoIP desk phones in this range support wall mounting using a bracket. Some phones include a wall mount bracket in the box; others require a separately purchased bracket.
Wall mounting is common in warehouses, factory floors, corridors, kitchens and anywhere a desk phone isn't practical. If you are planning to wall-mount a phone, confirm:
For rugged environments where a standard desk phone wouldn't survive, consider a ruggedised DECT handset (covered below) instead of a wall-mounted desk phone.
A business headset frees your hands during calls and reduces fatigue for staff on the phone all day. The main choice is between wired and wireless.
Wired headsets connect via a 3.5mm jack, USB or a proprietary headset port (RJ9/RJ11) on your desk phone. They are reliable, require no charging, and are generally less expensive. Best for staff who stay at their desk and want a simple, always-ready solution.
Wireless (DECT or Bluetooth) headsets give staff the freedom to move around — away from the desk, to a printer, into a meeting room — while staying connected on a call. DECT wireless headsets connect to your desk phone via an Electronic Hook Switch (EHS) adapter and typically offer 150–180m range. Bluetooth headsets pair directly with phones that have built-in Bluetooth. Most wireless headsets give 10–14 hours of talk time per charge.
An ATA (Analog Telephone Adapter) is a device that bridges the gap between traditional analog telephone equipment and a modern VoIP/SIP network. It converts the analog signal from a standard phone, fax machine, lift/emergency phone, or paging system into SIP, allowing that device to register to your VoIP phone system as if it were an IP phone.
ATAs typically have 1–2 FXS ports (to connect analog devices) and register each port as a separate SIP extension on your phone system. The Grandstream HT series and Cisco ATA series are popular choices.
FXS connects an analog device such as a phone, fax or lift phone to a VoIP network.
FXO connects an incoming analog telephone line to a VoIP system.
For faxing, confirm that both the ATA and your provider support T.38.
DECT (Digital Enhanced Cordless Telecommunications) is the standard technology for cordless business phones. Unlike Wi-Fi phones, DECT uses a dedicated radio frequency (1.88–1.9 GHz) designed specifically for voice — giving low interference, long range, and excellent call quality.
A DECT system consists of a base station (which connects to your network and registers to your VoIP system) and one or more handsets. The base station is the brains of the system — you register SIP accounts to the base, not to individual handsets.
When buying a DECT system, look for kits that include both the base and a handset (for example the Yealink W76P, W77P, W78P or W79P). Adding extra handsets later (for example the W56H or W78H) lets you expand the system without buying another base.
If you already have a Yealink DECT base station and want to add more handsets to the same system, buy a handset-only model — not a complete kit. Important: buying a handset-only model and expecting it to work as a standalone phone is one of the most common purchasing mistakes. A DECT handset requires a compatible base station to operate.
Standard DECT handsets are designed for office environments. For warehouses, factories, outdoor areas or anywhere with dust, moisture or drop risk, choose a ruggedised handset. Yealink's W57R (IP54) and W59R/W59R-Pro (IP67) are rated for water and dust, and some models include an emergency alarm button for lone-worker safety. The W59R is IP67-rated and can be fully submerged for a short period — suitable for wet or cold-storage environments.
A single DECT base station covers a limited area (typically 50m indoors). For larger buildings or multi-floor deployments where staff roam widely, a multi-cell DECT system allows multiple base stations to be linked together. Handsets roam seamlessly between bases — similar to the way a mobile phone roams between towers. Yealink's W80 and W90 systems support multi-cell deployments.
If you already have comprehensive Wi-Fi coverage and want cordless phones without installing DECT infrastructure, Wi-Fi IP phones (such as the Yealink AX83H or AX86R) connect to your existing Wi-Fi network and register as standard SIP extensions. Wi-Fi 6 models with 802.11k/v/r roaming handle handover between access points cleanly during a call. The AX86R is IP67-rated and includes an emergency alarm — a good option for rugged Wi-Fi environments.
The number of extensions your system can support depends on your phone system (PBX), not the phones themselves. The phones are endpoints — the PBX controls how many can be registered and how many simultaneous calls can be made.
Hosted/cloud PBX: your provider sets the maximum extensions based on your plan. Most Australian hosted providers support as many extensions as you pay for — there's no hard technical limit in the phones.
On-premise PBX (e.g. 3CX): the number of simultaneous calls depends on your 3CX licence (which is sold in call-concurrent increments). The number of registered extensions is typically much higher than the simultaneous call limit.
The phones themselves: each SIP phone registers as one extension. A phone with multiple SIP accounts (e.g. 16 SIP accounts) can be logged into up to 16 different extensions — useful for a receptionist who monitors multiple direct numbers.
Confirm these items before placing a multi-phone order or starting a phone-system migration.
Clear answers covering phones, systems, connectivity, codecs, deployment and business continuity.
A regular analog phone connects to the PSTN (public telephone network) via a physical copper line. A VoIP phone converts voice into data packets and sends them over your internet connection or IP network. VoIP phones require a phone system (hosted or on-premise) and a SIP trunk or cloud calling service.
Yes. NBN delivers an internet connection, and VoIP phones run over your internet connection. You need a VoIP service provider to supply SIP credentials — the NBN connection itself does not include a phone service. If you previously had a bundled phone service (PSTN or VoIP) with your ISP, check whether it was discontinued when you migrated to NBN.
No. A VoIP phone requires an active internet or IP network connection to register and make calls. If your internet goes down, VoIP calls will also go down unless you have a failover solution (such as 4G backup).
Yes — this is called number porting. Your VoIP provider can port your existing landline or 13/1300/1800 number to their service. The process typically takes a few business days and is handled by your provider, not by the phone hardware.
The terms are used interchangeably. SIP (Session Initiation Protocol) is the signalling protocol used by virtually all modern VoIP phones. "IP phone" simply refers to a phone that communicates over an IP network. All phones in our range are SIP-based IP phones.
This depends on your internet bandwidth and your codec. A standard G.711 call uses approximately 85–100 Kbps of bandwidth per concurrent call. An HD codec like Opus uses slightly more but delivers better audio. A 50 Mbps connection can comfortably handle 20–30 simultaneous calls — but the quality of your connection (jitter, latency and packet loss) matters more than raw speed.
You can register phones directly to a SIP trunk provider without a PBX — this is called a direct SIP configuration. However, without a PBX you lose features like internal extensions, voicemail-to-email, call queues, auto-attendants, call recording and hunt groups. For most businesses with two or more staff, a cloud PBX or an on-premise system like 3CX is worth the investment.
3CX is a popular software-based IP PBX that runs on a Windows or Linux server, or in the cloud. It is widely used in Australia for small-to-medium businesses. All standard SIP phones in our range work with 3CX. Yealink, Grandstream and Snom phones have deep integration with 3CX including auto-provisioning.
A DECT phone uses a dedicated radio frequency (1.88 GHz) and requires a DECT base station connected to your network. A Wi-Fi phone connects to your existing Wi-Fi access points. DECT is more reliable for voice in most environments and has better range; Wi-Fi is simpler if you already have blanket Wi-Fi coverage and don't want to install additional hardware.
Yes — most business VoIP phones support wall mounting. Some include a bracket; others require a separate bracket accessory. If you are planning a wall-mount installation, confirm the bracket is available for your chosen model before ordering.
Every phone we sell is the genuine Australian version, covered by the official manufacturer warranty. Warranty periods vary by brand and product — check the individual product page or contact us for details on a specific model.
Yes. We offer bulk pricing for larger orders, and dedicated pricing for education institutions, government organisations and businesses. Contact us for a quote.
HD Voice (also called wideband audio) captures a wider frequency range than a standard phone call — 50–7,000 Hz vs 300–3,400 Hz on a traditional line. In practice, voices sound clearer and more natural, fatigue is reduced on long calls, and it's easier to distinguish similar-sounding words. All phones in our range support HD Voice, provided your VoIP provider and phone system also support a wideband codec (G.722 or Opus are the most common).
Noise cancellation on a microphone filters out background sounds — keyboard noise, office chatter, air conditioning — before your voice is transmitted to the other party. This is separate from speaker-side noise cancellation (which affects what you hear). For open-plan offices, call centres or any noisy environment, noise cancellation on both the phone and headset microphone makes a significant practical difference to call quality.
PoE devices are classified by how much power they draw. Class 1 draws under 3.84W (basic desk phones), Class 2 under 6.49W, Class 3 under 12.95W (phones with colour screens, Wi-Fi, Bluetooth), and Class 4/802.3at (PoE+) under 25.5W (video phones, large touchscreens). Your switch must be capable of supplying the required class for each port. If you're unsure, check your switch's PoE budget and the power consumption listed on each phone's datasheet.
A SIP account (also called a line or extension) is a registered identity on your phone system — essentially a phone number or internal extension. A phone with 4 SIP accounts can be logged into 4 different extensions simultaneously. For most office workers, 1–2 SIP accounts is enough. Receptionists and call handlers benefit from 4–16, as each account can represent a different DDI number or department line.
BLF stands for Busy Lamp Field. It's a feature that lets a programmable button on your phone show the real-time status of another extension — green when the colleague is available, red when they're on a call, flashing when their phone is ringing. Pressing a BLF key can speed-dial, transfer a call, or pick up a ringing extension. It's the feature that makes a receptionist's job manageable when handling calls for a team.
Auto-provisioning (also called zero-touch provisioning) lets you configure phones automatically from a central server or cloud management platform, rather than manually entering settings on each device. When a provisioned phone boots up, it connects to a provisioning server, downloads its configuration, and registers itself — without anyone touching it. Yealink, Grandstream and Snom all support auto-provisioning with popular PBX platforms including 3CX. For deployments of 5 or more phones, auto-provisioning saves significant time and reduces configuration errors.
Codecs determine how voice is compressed and transmitted. G.711 (a-law in Australia) is uncompressed, uses around 85 Kbps per call, and delivers the highest quality. G.729 is compressed, uses around 25 Kbps per call, and is useful where bandwidth is limited — but reduces audio quality slightly. Opus is a modern wideband codec that adapts its bitrate dynamically and delivers HD Voice quality. Most Australian VoIP providers support G.711 as default; check with your provider whether G.729 or Opus are available if bandwidth is a concern.
Yes — a SIP phone can register to your business phone system from any internet connection, anywhere in the world. A staff member working from home or a hotel can register their desk phone (or a softphone on their laptop) to the office PBX and make and receive calls on their business number. Some providers use a VPN for security; others support direct SIP registration with TLS encryption. Your IT administrator or VoIP provider can advise on the correct setup for remote workers.
T.38 is a protocol specifically designed for sending faxes over IP networks. Standard fax machines were designed for analog lines and don't work reliably over a regular VoIP connection (G.711 fax pass-through) because VoIP's packet-based nature introduces jitter and packet loss that fax signals can't tolerate. T.38 repackages fax signals in a way that's tolerant of these issues. If you need to send or receive faxes over VoIP, confirm that both your ATA (or phone system) and your VoIP provider support T.38. If fax is critical to your business, discuss this with your provider before migrating.
STUN (Session Traversal Utilities for NAT) helps VoIP devices work correctly when they're behind a router or firewall that uses NAT (Network Address Translation — the way most home and office routers share a single public IP address across multiple devices). Without STUN or an equivalent solution, one-way audio (where you can hear the other party but they can't hear you, or vice versa) is a common problem. Most VoIP providers handle this at their end, and most modern phones handle it automatically. If you experience one-way audio, check your phone's STUN settings or contact your provider.
QoS (Quality of Service) is a network setting that prioritises voice traffic over other types of data. Without QoS, a large file download or video stream on the same network can cause choppy, delayed or dropped calls. On a managed network switch, QoS can tag voice packets so they're processed first. On a home or small office router, enabling QoS settings for your VoIP provider's IP addresses can make a noticeable difference to call quality. For business deployments, placing phones on a separate VLAN with QoS priority is considered best practice.
Concurrent calls refers to how many phone conversations can happen simultaneously across your phone system. This is separate from the number of extensions. For example, a business with 20 staff and 20 extensions might only ever have 5 conversations happening at the same time — so a system licensed for 5 concurrent calls is sufficient. Running out of concurrent call capacity means callers hear an engaged tone even when a staff member is available. Your VoIP provider or PBX licence determines your concurrent call limit — not the phones.
These terms come up when buying ATAs and gateways. An FXS port (Foreign Exchange Station) is the type of port that connects an analog device — like a regular phone, fax machine or lift phone. It's the port that provides the dial tone. An FXO port (Foreign Exchange Office) connects to an analog phone line coming in from the telephone network — used when you want to connect a legacy PSTN line into a VoIP system. ATAs in our range (like the Grandstream HT series) are FXS devices — they let you connect your analog phone to a VoIP network. An FXO gateway does the opposite.
Three things matter more than raw speed: latency (delay — should be under 150ms one-way), jitter (variation in delay — should be under 30ms) and packet loss (lost data packets — should be under 1%). A standard Australian NBN 25 or NBN 50 connection is more than fast enough for multiple VoIP calls. The problem is usually jitter or packet loss, often caused by a congested router, cheap modem or poorly configured network. Run a VoIP-specific quality test (search "VoIP test" — several free tools exist) to assess your connection before deploying phones. If results are poor, QoS settings or a router upgrade usually fixes the problem before any phones are changed.
A softphone is an application that runs on a computer, smartphone or tablet and behaves like a VoIP phone — it registers to your phone system using SIP credentials and lets you make and receive calls through a headset or the device's speaker. Softphones are useful for remote workers, travelling staff or businesses that want to avoid physical hardware altogether. The trade-off is that audio quality depends on the device's microphone and speakers (or headset), and calls can be disrupted by system notifications or background processes. Dedicated desk phones provide more consistent audio quality and are less prone to interruption.
Yes — this is called a shared line appearance or extension sharing. Both phones register to the same SIP account, so when that extension receives a call, both phones ring simultaneously. Either person can answer. This is useful for a manager and their assistant sharing a line, or a front desk with two handsets. Some PBX platforms handle this via BLF and call pickup rather than true shared lines — check with your PBX provider for the best approach on your system.
A ring group (or hunt group) is a phone system feature — not a phone hardware feature — that routes incoming calls to a group of extensions rather than a single number. For example, all calls to your sales number ring the three sales team members' phones simultaneously (ring all) or in sequence (linear hunt). The first person to answer takes the call. This is configured in your PBX or hosted phone system, not on the phones themselves.
If the power goes out, PoE phones lose power along with the switch, and calls drop. Unlike traditional analog phones that were powered by the telephone line, VoIP phones require both network connectivity and power. For business continuity, options include: a UPS (Uninterruptible Power Supply) on your network switches and router to keep the system running during brief outages, a 4G/LTE failover router to maintain internet connectivity, or a mobile softphone as a backup. Discuss your business continuity requirements with your VoIP provider — many hosted providers offer failover routing to a mobile number if your primary phones go offline.
Tell us which provider or phone system you use, how many users you need to equip and which roles require desk phones, headsets or cordless handsets. DigiPhone can help you compare suitable models and prepare a project quote.
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