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What is a 3CX Phone System?

A 3CX phone system is a PBX, which stands for Private Branch Exchange. This is a private telephone network used within a business. The users of the PBX phone system can communicate internally (within their company) and externally (with the outside world), using different communication channels like Voice over IP, ISDN or analogue. A 3CX also allows you to have more phones than physical phone lines (PTSN) and free calls between users. Additionally, it provides features like transfer calls, voicemail, call recording, interactive voice menus (IVRs) and call queues.

Time and technology have changed the consumer telephony landscape in the past years, with the flag-bearer being the Open-Standards-based IP 3CX. Now you can telephone via the Internet Protocol technology. 3CX phone systems are available as hosted or virtual (cloud) solutions and as on-premise solutions on your own hardware.


what is a pbx system


This image gives us an idea of what a 3CX system allows in terms of connectivity and reachability. With a traditional PBX, you are typically constrained to a certain maximum number of outside telephone lines (trunks) and to a certain maximum number of internal telephone devices or extensions. Users of the PBX phone system (phones or extensions) share the outside lines for making external phone calls.

Switching to a 3CX brings many benefits and opens up possibilities, allowing for almost unlimited growth in terms of extensions and trunks, and introducing more complex functions that are more costly and difficult to implement with  a traditional PBX, such as ring groups, queues, digital receptionists, voicemail and reporting. C2 Communications relies on 3CX, as it has established itself as the leading IP-PBX manufacturer, ticking all the checkboxes for any business looking for enterprise-grade features.

What is an Auto-Attendant?

An auto-attendant (or automated attendant) is a voice menu system that allows callers to be transferred to an extension without going through a telephone operator or receptionist. The auto-attendant is also known as a digital receptionist.


For a caller to find a user on a phone system, a dial-by-name directory is usually available. This feature lists users by name, allowing the caller to press a key to automatically ring the extension of a user once his / her extension is announced by the auto-attendant. If a user is not available, the digital receptionist directs callers to the appropriate voicemail of the user to leave a message. Having an auto-attendant in a phone system is a very useful and cost-effective feature for a business, as it replaces / helps the human operator by automating and simplifying the incoming phone call procedure. The 3CX used by C2 Communications includes a free digital receptionist feature.

Benefits of an IP 3CX VoIP Phone System?

What is an IP 3CX?

An IP 3CX is a complete telephony system that provides telephone calls over IP data networks. All conversations are sent as data packets over the network. The technology includes advanced communication features but also provides a significant dose of worry-free scalability and robustness. The IP 3CX is also able to connect to traditional PSTN lines via an optional gateway, so we can upgrade your business communication to the most advanced voice and data network easily.

You don’t need to disrupt your current external communication infrastructure and operations. With an IP 3CX system from C2 Communications, you can even keep your regular telephone numbers. This way, the IP 3CX switches local calls over the data network inside the enterprise and allows all users to share the same external phone lines.

How it works

what is a pbx system


An IP 3CX consists of one or more SIP phones, an IP 3CX server and optionally a VoIP gateway to connect to existing PSTN lines. The IP 3CX server functions in a similar manner to a proxy server. SIP clients, being either softphones or desk phones, register with the IP 3CX server, and when they wish to make a call they ask the IP 3CX to establish the connection. The IP 3CX has a directory of all phones / users and their corresponding SIP address and thus is able to connect an internal call or route an external call via either a VoIP gateway or a VoIP service provider.

Benefit #1: Much easier to install & configure than a proprietary phone system

An IP 3CX runs as software on a computer and can leverage the advanced processing power of the computer and user interface as well as features. C2 Communications takes care of installing and maintaining your IP C3X.

Benefit #2: Easier to manage because of web / GUI based configuration interface

An IP 3CX can be managed via a web-based configuration interface or a GUI, allowing you to easily maintain and fine tune your phone system. Proprietary phone systems have difficult-to-use interfaces which are often designed to be used only by phone technicians.

Benefit #3: Significant cost savings using C2 Communications VoIP

With an IP 3CX, you can easily use the VoIP of C2 Communications for long distance and international calls. The monthly savings are significant. If you have branch offices, you can easily connect phone systems between branches and make free phone calls.

Benefit #4: Eliminate phone wiring

An IP phone system allows you to connect hardware IP phones directly to a standard computer network port (which it can share with the adjacent computer). Software phones can be installed directly on the PC. You can now eliminate the phone wiring and make adding or moving of extensions much easier. In new offices, you can completely eliminate the need for wiring extra ports to be used by the office phone system.

Benefit #5: Eliminate vendor lock-in

IP 3CX is based on the open SIP standard. You can mix and match any SIP hardware or software phone with the SIP-based IP 3CX, PSTN gateway or C2 Communications VoIP. In contrast, a proprietary phone system often requires proprietary phones to use advanced features, and proprietary extension modules to add features.

Benefit #6: Scalable

Proprietary systems are easy to outgrow. Adding more phone lines or extensions often requires expensive hardware modules. In some cases you need an entirely new phone system. Not so with an IP 3CX from C2 Communications. A standard computer can easily handle a large number of phone lines and extensions – just add more phones to your network to expand.

Benefit #7: Better customer service & productivity

With an IP 3CX you can deliver better customer service and better productivity. Since the system is now computer-based, you can integrate phone functions with business applications. For example, bring up the customer record of the caller automatically when you receive his / her call, dramatically improve customer service and cut costs by reducing time spent on each caller. Outbound calls can be placed directly from Outlook, removing the need for the user to type in the phone number.

Benefit #8: Twice the phone system features for half the price

Since an IP 3CX is software-based, it is easy for C2 Communications to add and improve feature sets. The 3CX phone system comes with a rich feature set, including auto-attendant, voice mail, ring groups, and advanced reporting. Unified Communications features are included, to support presence, video and audio conferences and free calls via the data network. These options are often very expensive in proprietary systems.

Benefit #9: Allow hot desking & roaming

Hot desking, the process of being able to easily move offices / desks based on the task at hand, has become very popular. Unfortunately, traditional PBXs require extensions to be re-patched to the new location. With an IP 3CX, the user simply takes his phone to his new desk – no patching required.

Users can roam too – if an employee has to work from home, he / she can simply fire up their SIP software phone and is able to answer calls to their extension, just as they would in the office. Calls can be diverted anywhere in the world because of the SIP protocol characteristics.

Benefit #10: Better phone usability

Employees often struggle using advanced phone features. Setting up a conference, or transferring a call on an old PBX requires detailed instructions. Not so with an IP 3CX – all features are easily performed from a user-friendly GUI. In addition, users get a better overview of the status of other extensions, of inbound calls, call queues, and presence via the apps. Proprietary systems often require expensive “system” phones to get an idea what is going on on your phone system and even then, status information is cryptic at best.


Investing in a software-based IP 3CX from C2 Communications makes a lot of sense, not only for new companies buying a phone system, but also for companies who already have a PBX. An IP 3CX delivers such significant savings in management, maintenance, and call costs, that upgrading to an IP 3CX should be the obvious choice for any business.

How an IP 3CX / VoIP Phone System Works

A VoIP phone system / IP 3CX system from C2 Communications consists of one or more SIP phones / VoIP phones, an IP 3CX server and optionally includes a VoIP Gateway. The IP 3CX server is similar to a proxy server: SIP clients, being either softphones or hardware-based phones, register with the IP 3CX server, and when they wish to make a call, they ask the IP 3CX to establish the connection. The IP 3CX has a directory of all phones / users and their corresponding SIP address and thus is able to connect an internal call or route an external call via either a VoIP gateway or a VoIP service provider to the desired destination.


what is a pbx system


At the center we have the IP 3CX. Starting from the bottom, we see the corporate network. This is the company’s local network. Through that network, Computers running SIP clients such as the 3CX softphones, and IP phones connect directly to the 3CX. On the left, we see the company’s router / firewall connected to the internet. From there it can connect to remote extensions in the form of computers running the softphones, remote IP phones, mobile devices running the 3CX Android and iOS apps, and bridged PBX’s. By using the VoIP network of C2 Communications, we can connect you to the PSTN network. To the right, a VoIP Gateway connects the 3CX directly to the PSTN network.

What is IVR?

Interactive Voice Response or IVR is a telephone technology that allows customers to interact with the company’s host system through configurable voice menus, in real time, using DTMF tones.



How does an IVR system operate?

In an Interactive Voice Response system, callers are given the choice to select options by pressing digits. The press of the digit on the telephone keypad sends a DTMF tone to the company host system which then selects the appropriate action / response according to the digit pressed.

Where are IVR systems used?

IVR systems can normally handle and service high volumes of phone calls. With an Interactive Voice Response system, businesses can reduce costs and improve customers’ experience as Interactive Voice Response systems allow callers to get the information they need 24 hours a day without the need of costly human agents. Some IVR applications include telephone banking, flight-scheduling information and televoting. The 3CX software used by C2 Communications has a built-in IVR that is designed to boost the competence of any business by increasing flexibility, simplifying processes and reducing costs, at the same time as improving customer satisfaction.

What is DID – Direct Inward Dialing?

DID stands for – Direct Inward Dialing (or DDIDirect Dialling Inward in Europe) is a feature offered by telephone companies for use with their customers’ PBX system, whereby the telephone company allocates a range of telephone numbers associated with one or more phone lines. DID allows a company to assign a personal number to each employee, without requiring a separate physical phone line for each to connect to the PBX. This way, telephony traffic can be split up and managed more easily.

For example, if an organisation has 25 employees and each employee has a separate telephone number or extension, within its physical location, the organisation can rent 10 physical trunk lines from the telephone company that will allow 10 phone calls to take place simultaneously. Others would have to wait for an available line and anyone dialling into the system while all 10 lines are in use would get either a busy signal or be directed to a voicemail system. A DID system can be used for fax and voice transmissions.

DID works similarly for VoIP communications. To allow PSTN users to directly reach VoIP users, DID numbers are assigned to a gateway. The gateway connects the PSTN (public switched telephone network) to the VoIP network, routing and translating calls between the two networks for the VoIP user. Calls from the PSTN will be directed to the VoIP user who holds the corresponding DID number. DID requires that you purchase an ISDN or digital line and ask the telephone company to assign a range of numbers. You will then need DID capable equipment at your premises which consists of BRI, E1 or T1 cards or Gateways.

What is a Voicemail System?

A voicemail system is a centralised system used in businesses for sending, storing and retrieving audio messages, just like an answering machine would do at home. Voicemail systems make a phone system more flexible and powerful by allowing information and messages to pass between users even when one of them is not present.

How does a voicemail system work?

Each extension in a phone system is normally linked to a voice mailbox, so when the number is called and the line is not answered or is busy, the caller listens to a message previously recorded by the user. This message can give instructions to the caller to leave a voice message, or provide other available options. Options include paging the user or being transferred to another extension or a receptionist. Voicemail systems also provide notifications to users to inform them of new voicemails. Most modern voicemail systems provide multiple ways for user to check their voicemail including access through PC’s, mobile phones, landlines or even through SIP clients running on smartphones.

A voicemail system in a business is essential to keep external and internal communications flowing seamlessly and efficiently. The 3CX software used by C2 Communications has integrated a free voice mail system in its IP 3CX for Windows. The 3CX phone system for Windows includes a complete voice mail solution that incorporates unified communications by allowing voicemail to be forwarded to the user’s email inbox.

10 Reasons to Switch to an IP 3CX

Why Your Next Phone System Should be Software-Based and Running on a Mainstream Operating System

This article explains the benefits of having a software-based phone system and why it makes sense to move away from proprietary solutions and straight to a real software-based solution running on a mainstream, commercially supported operating system.


The PBX business is clearly fast asleep. In the IT world, in 1970 you paid around 100,000 DM for a computer for basic calculations. In 2008, you pay 1,000 EUR for a laptop and can run your whole business with it. In the PBX business, in 1970 you paid around 10,000 DM for a black box, to place calls and transfer calls. In 2008, you still pay 10,000 EUR for the same thing. The phone system of 1970 is the phone system of today, they look and do almost the same. This is very different from what has been happening in the IT business over the last 40 years, where we have seen lots of progress and a reduction in cost. The PBX business has a lot to catch up on. It desperately needs to innovate, and it is software solutions that can make that happen.

Software-based PBX

The lack of progress and innovation in the PBX industry has been largely caused by the fact that the traditional PBXs run on a proprietary and limited operating system, which has only archaic development tools. A software 3CX will leverage the latest operating system features and modern development environments, allowing developers to reuse features in the operating system and quickly add new functionalities to the 3CX.

The 3CX must run on a mainstream OS

A key point is that the 3CX software runs on a mainstream operating system that is commercially supported and maintained, guaranteeing regular OS updates supporting the latest hardware, and most importantly security updates for vulnerabilities found. Administrators need to get away from black boxes and take control of their phone system.

Windows or Linux?

The choice of Windows or Linux will largely depend on your IT infrastructure and the experience of your administrations. The C2 Communications 3CX does not require additional skills from your team or requires you to buy additional hardware or support agreements with operating system vendors. We’ll go with the computer system you prefer.

10 Reasons to Switch to an IP 3CX

Applying OS patches

Operating systems need to have patches applied on a regular basis. If administrators have chosen a PBX that runs on an operating system they are familiar with then they will be able to easily apply these patches. Windows administrators will struggle to apply Linux patches. This is mostly important if you want to host the 3CX on-premise; if you decide for a hosted or virtual 3CX by C2 Communications, we take care of the whole maintenance process.

Upgrading is easier

Upgrading to a new version of your 3CX is a simple process if you are familiar with the underlying operating system: Upgrades will be done in a matter of 5 to ­10 minutes.

Fault tolerance through easy backup of your 3CX

Software-based also means that you can easily backup your whole 3CX. In the event of a hardware failure, you can restore your phone system on another machine in a matter of minutes using the inbuilt backup function.

Leverage your existing server hardware

Because modern servers have ample processing power, a software-based 3CX can run on an existing server with other applications, saving on hardware cost, energy consumption and administration costs. No need for a dedicated machine or a low performance appliance. Optionally, using Hyper-V, VMware or KVM you can also virtualise your 3CX and separate it from other apps, without requiring a separate server.

Low resource usage

Modern day hardware can easily support 3CX hardware requirements. The table below shows the processor and memory usage of a busy Windows Server for handling 16 continuous calls: Low processor and memory usage means it can be run safely on an existing Windows server that is running other applications.


Machine Specs

Intel Core 2 Duo CPU, E 4500 @ 2.20 GHz, 4 GB of RAM, 50 GB Hard disk SATA and a 100Mbps Network connection

Operating system

Windows server 2012 R2

Other applications installed

IIS, exchange server and active directory

Simulated exchange load (using Exchange Load Simulator)

25 users making heavy use of exchange (sending mail, scheduling meetings, checking inbox etc)

Processor time used by exchange server


3CX phone system


3CX simulated call load

16 simultaneous calls continuously

Call rate

0.5 calls/second equalling 2000 calls per hour

Processor usage of all 3CX services

Less than 15% CPU

Total memory usage of all 3CX services

300 megabytes

Peak processor usage




Alternatively, we can install your 3CX as a virtual instance using Hyper­-V, Vmware or KVM. These are great virtualisation platforms with superior I/O performance which allow you to virtualise your 3CX, even for larger installations.

Easy to scale

Because your 3CX is running on a modern server, it’s easy to scale compared to a traditional PBX or an appliance. Modern server hardware will allow you to add almost unlimited extensions, lines and functions because servers have ample processing power. Appliances on the other hand are limited by the number of telephony ports and by their limited processing power and memory. Your appliance will run out of steam as soon as you start adding more lines, phones, and making use of more processor intensive functions such as conferencing. Before you know it you will need to discard your old appliance and buy a new ‘bigger’ appliance.

On-site support and replacements

An important advantage in working with C2 Communications is that we provide leading server hardware with on-site support and replacements in case you host your 3CX on-premise.

Easy integration with other applications

Another major advantage of a software-based 3CX is that it integrates easily with your other business applications and uses the same API’s. It’s no longer a black box sitting in the corner refusing to talk to the rest of your applications. And with this integration, we can gain features and thus productivity. For example:

Integrate with CRM system or database

Our software 3CX will easily talk to other systems such as the CRM, mail or database server and greatly improve your productivity and customer service. Match a caller ID to a customer and know who is calling. Automatically log calls with customers for reporting and customer service purposes.

Integrate with your user directory

Software-based 3CX will allow you to connect with your user directory of choice, be it LDAP or Active Directory. Ensure that user data is not duplicated and always up to date, saving valuable administration time and ensuring user data is correct and synchronised.


The 3CX system from C2 Communications delivers great advantages to your company. It will offer you:

  • Easy installation & management
  • Ability to leverage your existing server hardware
  • Fault tolerance via easy backup and restore
  • Integration with your existing business apps

What are the Benefits of an IP 3CX?

An IP 3CX phone system has a number of benefits.

Easy Installation and Configuration

A traditional PBX is composed of proprietary hardware and software management tools. These tools are typically managed over a serial or console cable, and each vendor has different tools for this. This means that, once you have committed to a traditional system, you are bound to their professional services, and the vendor can, and will, charge premium prices for the service simply because the customer has no alternative service provider to go to.

Our 3CX, on the other hand, is a software-based solution. This automatically means that it is much easier to install and configure without training.

Ease of Management

Our 3CX solution provides a web-based configuration interface. The obvious benefit to this is that the system administrator has access to the configuration of the system – the configuration tools are no longer hidden away from the system administrator, allowing him to make the changes himself if he so desires.

IP-Based Means IP Network

Every telephone system needs to have wiring to connect phones to the PBX. But here is the point of an IP-3CX – your office already has the wiring, because your phones and IP-3CX run on the same wiring that your corporate network is already using. And again, your system administrator already knows how his LAN network is wired into the network cabinet – the phones are simply additional network devices just like any computer on the LAN.

This also means that a user can easily move his operations from one desk to another within the office. As long as the network wall sockets are connected to the network cabinet, all he needs to do is unplug his phone from the network wall socket, take the phone to his new work location, and plug the phone back into the network wall socket at his new work location. The configuration of the phone does not need to change, and the extension number also does not need to change – it will just work.

Receive and Make calls Anywhere, Everywhere

The SIP protocol is an IP-based protocol, C2 Communications offers SIP softphones for Android and iOS. This transforms your smartphone into an extension on the corporate IP-3CX, so as long as your phone has IP connectivity, it can talk to the IP-3CX – from a coffee shop, from a hotel room, from an airport lounge, from a yacht marina. Be connected – anywhere, everywhere.

Cost Reduction

You can interface a regular land line to an IP-3CX, using a gateway device. But you have an IP-3CX now – you don’t need to do this any more. Or better phrased, you are no longer at the mercy of your regular telephone communications provider. You can use the VoIP services of C2 Communications, that can deliver telephony over the internet. Just with a simple number porting, you can immediately reduce your call costs. Why? Because land-line telcos have been overcharging for telephony since the first “Hello”. This is why VoIP is sometimes considered a “disruptive” technology – it breaks the traditional telco’s model, by removing their position of control on the relationship.

Compliance with SIP Standards Eliminates Vendor Lock-In

Today’s mainstream SIP-based deskphones improves your return on investment. If you need to change from one IP-PBX to another, your phones are still usable – this is because the phones talk a universal language called SIP.

Scalability – No Limits

A traditional PBX was essentially a hardware device sitting in some corner of your office. It would have a number of empty “slots” to add hardware capacity to your system. Each “slot” would allow you to add “x” number of extensions or “y” number of lines. Once the “slots” were full, you would have reached the limit, and the search for a new telephone system would start – but not before you find the money to replace it!

An IP-3CX does not suffer from this limitation, because software does not have a limited number of “slots”. If the computer it runs on has the horsepower, you can scale upwards at will. With C2 Communications, you can simply update your licence and get more “slots” assigned – no need to touch anything on the system.

Reporting and Monitoring

Again, the power of the 3CX software-based solution from C2 Communications really shines through on the reporting and monitoring functions. For C2 Communications, extracting data from the call records is a relatively simple task. If a reporting feature is requested, then we can provide a new report simply by way of a system update. Live monitoring of activity on the system is another great bonus which web-based management offers.

What is Unified Communications?

Currently, there is a large number of communication channels, and of different types, made available to technology users. To put a (indicative but by no means complete) list together:

  • E-mail
  • Telephony (fixed-line, mobile, VoIP-based)
  • Audio/video conferencing
  • Presence (as an example, consider your list of contacts in Skype, and the relevant icons that show individual contacts to be online or away)
  • Social media (think Twitter, Facebook, Vines, Whats App, Instagram, and so on…)



Some of these communication channels are of the “store-and-forward” type, in the sense that the information is delivered in one direction, and remains accessible (almost) indefinitely for the remote parts to view it when he has the time; e-mail is the grand-daddy of this communication style. Others, however, are more immediate, and require rapid response (often interrupting other tasks); telephony is the obvious largest contender in this category.

Each of these different communication channels typically requires its own “app” to access the information being exchanged. As the number of channels we need to give attention to increases, the harder it becomes to manage them all efficiently.

So What is Unified Communications?

Unified Communications, often abbreviated to simply UC, is a generic hold-all term to describe the market’s efforts to integrate all the “apps” (and therefore the communication channels) to allow the user to have all this information easily accessible, irrespective of when or where he needs access (home, work, in a car, on a train…), and how he needs access (laptop, tablet, smartphone, internet cafe…).

UC effectively blurs the demarcation lines between the communication channels. For example, a user can receive a voicemail message and can choose to access it through email or any phone. The sender’s status can be seen through presence information, and if online a response can be sent immediately through chat message or video call. The objective of Unified Communications is to unify and streamline those business procedures that involve human communications.

In summary, the term Unified Communications does not describe a technology, or even a group of technologies, but rather it defines the ongoing process of convergence that is happening in the market, bringing together vendors, technologies, applications, processes, and users – Unified Communications is the integration of all separate communications components into a homogeneous, efficient, productive user experience.

What is Voicemail to Email?

With the voice and fax delivery feature in the 3CX software from C2 Communications, your voicemail messages are delivered to multiple email addresses with 3CX. As part of its extensive unified communications features, our customers leverage the voicemail to email feature to ensure they’re always reachable, even when they’re on the go. 3CX converts voicemails into .wav audio files and sends them directly to you.

What are BLF Function Keys?

Some IP phones have BLF function keys. BLF is an acronym for Busy Lamp Field, which is a light on an IP phone that tells you whether another extension connected to the same 3CX is busy or not. Depending on the type of phone you have, BLF’s will remain green, meaning the extension is free to talk. If the BLF starts to flash red, it normally means someone is calling that extension. If the BLF function key is red, it means the owner of that extension is on a call.

BLF’s are also helpful when answering another colleagues phone. For instance, if Bianca isn’t at her desk and someone rings her extension, Andy can pick up the call simply by clicking the flashing BLF. Also, before Helen calls Chris, she’ll be able to see if Chris is on a call or not.

Configuring BLF Function Keys

BLF function keys can be configured really easily using the 3CX Management Console. Once the BLF function keys have been configured, the IP phone subscribes to a resource list that’s available on the IP 3CX. This resource list gives the phone all the necessary information about all the other extensions on the 3CX.

BLF works through the SIP protocol by making use of the subscribe and notify messages. In a normal scenario, the phone is the subscriber and the IP 3CX is the notifier.

Setting Up BLF Function Keys on snom and Yealink IP Phones

For step-by-step instructions on how to setup BLF keys on phones supported by the 3CX Phone System, consult our detailed configuration guides.


Why is NBN not a business grade service?

NBN is a consumer product and this means the service is contended. A business grade internet connection will provide you with 1:1 contention and a dedicated line to your site. Why is this better?

  • Your internet is not dependant on any other activity in the street or on the exchange
  • You have a guaranteed speed and latency
  • Broadband is the foundation of VoIP, any issues on the line will be first noticed on your phones.

Do I really need Fibre 1000?

Yes! Fibre 1000 is not just about the bandwidth but the quality of the infrastructure. A Fibre 1000 connection can:

  • Provide download and upload speeds of 1000 mbps
  • Latency of less than 1 millisecond
  • Provide the backbone to the best quality VoIP telephony in Australia

More business activity occurs online making the internet a critical service. A Fibre 1000 connection should be considered as important as the foundations to your house.

How long is a contract term?

A Fibre 1000 contract term with C2 Communications is 48 months. Taking a lesser term contract will increase the monthly cost and installation fees will be stacked up.

How do I upgrade to Fibre 1000?

If you currently have a service expiring, we will be in touch to offer the Fibre 1000 product for your business.

If you’re looking to upgrade from another service provider, call us on 03 9819 0066 option 1.

What is there other than Fibre 1000?

We offer a full range of business grade services that includes:

  • Copper EFM services up to 20:20 mbps
  • Copper Hatteras services up to 40:40 mbps
  • Fibre 100 for outer metro zones
  • Fibre 500 for specific metro zones

What do I do if my internet is slow?

If you’re noticing a significant drop in speed on your service, first complete the following:

  • Restart your modem
  • Restart the Network Termination Unit
  • Wait for both units to power on, roughly 10 mins

If the service is still slow call our support on 03 9819 0066 option 2.

What do I do if my internet is down?

If you’re service is completely down, first complete the following:

  • Restart your modem
  • Restart the Network Termination Unit
  • Wait for both units to power on, roughly 10 mins

If the service is still down call our support on 03 9819 0066 option 2.

How To Series

How to use the 3CX Webclient

Angus takes us through the 3CX Webclient, a powerful tool designed to give all the 3CX phone features on a single Chrome tab.

If you have any questions or want to know more get in touch with us.

How to make an Outbound Call on your 3CX Softphone for Windows

Angus takes you through the best ways to make an outbound call using the 3CX Softphone for Windows. 3CX Softphone makes it easy to call without leaving your keyboard.

How to Change your Status and Call Forwarding Using the 3CX Softphone for Windows

Angus takes us through changing you status on the 3CX Softphone for Windows. Changing you status displays your availability within your organisation and gives you the freedom to take calls on the road or away from your desk.

How to Park and Transfer on your 3CX Softphone for Windows

Angus shows us how quick and easy it is to park and or transfer a call using the 3CX Softphone for Windows. There are many ways to move calls around your organisation and these are fastest ways to work.

How to use Call Queues on the 3CX Management Console

Angus shows us best practice when setting up your call queues on your 3CX Phone Systems. Call queues allow you to ‘queue’ multiple callers into your call centre so no call is ever lost.

How to use the Switchboard on your 3CX Softphone for Windows

Angus shows off the powerful Switchboard tool on 3CX Softphone for Windows. The Switchboard allows you to view the activity within your organisation and transfer, park, listen, barge, and end calls.

How to set your Voicemail on 3CX Softphone for Windows

Angus shows us how to setup your voicemail on the 3CX Softphone for Windows. The voicemail feature allows you to configure multiple messages for different status, giving your voicemail more personality and function.

How to configure Automatic Backups on the 3CX Management Console

Angus takes us through the simple steps of configuring automatic backups on the 3CX Management Console. Enabling backups allows you to rest easy should anything happen to your phone system. The 3CX backup can be restored within minutes with no data loss.

How to Configure Office Hours and Holidays in the 3CX Management Console

Angus takes us through the simple steps of setting business hours and holidays in the 3CX Management Console. Setting these hours lets your call routing automatically change in and out of office hours.

How to add a C2 Communications SIP Trunk

Angus takes us through the ease of adding a C2 Communications SIP trunk on the 3CX Management console.

Configuring your C2 Communications SIP trunk is fast and simple, and will have your business making calls in minutes.

How to configure your Digital Receptionist

Angus shows us the powerful Digital Receptionist feature in the 3CX Phone System. The Digital Receptionist can help your callers access departments faster and get your clients serviced better.

How to Make a Video Call on 3CX Softphone for Windows

Angus takes us through one of the most powerful tools on the 3CX Softphone for Windows, the Video Call. A video call between colleagues is a great way to stay in contact in different offices and share the screens you’re working on.

Microsoft Teams Calling

What support can businesses expect with Teams phone calling implementation?

Businesses can expect comprehensive support during Teams phone calling implementation, including planning, configuration, and troubleshooting. C2 Communications offers personalised support and expertise to ensure a successful deployment and ongoing management of the Teams phone calling solution.

How does direct routing with an SBC enhance Teams phone calling?

Direct routing with an SBC enhances Teams phone calling by providing secure and reliable connectivity between the enterprise’s infrastructure and the telephony world. It allows for greater control over call routing, enhanced security, and the ability to leverage existing telephony investments.

What security measures are in place for Teams phone calling?

Teams phone calling employs a range of security measures, including end-to-end encryption, secure access controls, and compliance with international standards. C2 Communications enhances security further with third-party SBC solutions for additional monitoring and protection.

Can Teams phone calling be customised for different business sizes?

Yes, Teams phone calling is highly customisable and scalable, making it suitable for businesses of all sizes. From small start-ups to large enterprises, the platform can be tailored to meet the specific communication needs of any organisation.

What are the benefits of using Microsoft Teams Phone for businesses?

Microsoft Teams Phone offers numerous benefits, including streamlined communication, enhanced collaboration, cost savings, scalability, and robust security features. It consolidates various communication tools into one platform, simplifying the user experience and IT management.

How does Teams phone calling integrate with existing infrastructure?

Teams phone calling is designed to integrate seamlessly with existing infrastructure, leveraging SBCs to connect with SIP Trunks and other telephony systems. C2 Communications assists in the integration process, ensuring compatibility and a smooth transition to the Teams platform.

SIP Trunking

Can you List all Known SIP Responses?

SIP responses are the codes used by Session Initiation Protocol for communication. We have put together a list of all the SIP responses known.

1xx = Informational SIP Responses

  • 100 Trying – Extended search is being perform so a forking proxy must send a 100 trying response.
  • 180 Ringing – The destination user agent has received the INVITE message and is alerting the user of  call.
  • 181 Call Is Being Forwarded – Optional, sent by server to indicate a call is being forwarded.
  • 182 Queued – Destination was temporarily unavailable, the server has queued the call until destination is available.
  • 183 Session Progress – This response may be used to send extra information for a call which is still being set up.
  • 199 Early Dialog Terminated – Send by the user agent server to indicate that an early dialog has been terminated.

2xx = Success Responses

  • 200 OK – Shows that the request was successful.
  • 202 accepted – Indicates that the request has been accepted for processing, mainly used for referrals.
  • 204 No Notification – Indicates that the request was successful but no response will be received.

3xx = Redirection Responses

  • 300 Multiple Choices –  The address resolved to one of several options for the user or client to choose between.
  • 301 Moved Permanently – The original request URI is no longer valid, the new address is given in the contact header.
  • 302 Moved Temporarily – The client should try at the address in the contact field.
  • 305 Use Proxy – The contact field details a proxy that must be used to access the requested destination.
  • 380 Alternative Service – The call failed, but alternatives are detailed in the message body.

4xx = Request Failures

  • 400 Bad Request – The request could not be understood due to malformed syntax.
  • 401 Unauthorized – The request requires user authentication. This response is issued by UASs and registrars.
  • 402 Payment Required – (Reserved for future use).
  • 403 Forbidden – The server understood the request, but is refusing to fulfil it.
  • 404 Not Found – The server has definitive information that the user does not exist at the (user not found).
  • 405 Method Not Allowed – The method specified in the request line is understood, but not allowed.
  • 406 Not Acceptable – The resource is only capable of generating responses with unacceptable content.
  • 407 Proxy Authentication Required – The request requires user authentication.
  • 408 Request Timeout – Couldn’t find the user in time.
  • 409 Conflict – User already registered (deprecated)
  • 410 Gone – The user existed once, but is not available here any more.
  • 411 Length Required – The server will not accept the request without a valid content length (deprecated).
  • 413 Request Entity Too Large – Request body too large.
  • 414 Request URI Too Long – Server refuses to service the request, the requested URI is longer than the server can interpret.
  • 415 Unsupported Media Type – Request body is in a non supported  format.
  • 416 Unsupported URI Scheme – Request URI is unknown to the server.
  • 417 Uknown Resource-Priority – There was a resource-priority option tag, but no resource priority header.
  • 420 Bad Extension – Bad SIP protocol extension used, not understood by the server.
  • 421 Extension Required – The server needs a specific extension not listed in the supported header.
  • 422 Session Interval Too Small – The request contains a session-expires header field with duration below the minimum.
  • 423 Interval Too Brief – Expiration time of the resource is too short.
  • 424 Bad Location Information – The request’s location content was malformed or otherwise unsatisfactory.
  • 428 Use Identity Header – The server policy requires an identity header, and one has not been provided.
  • 429 Provide Referrer Identity – The server did not receive a valid referred-by-token on the request.
  • 430 Flow Failed – A specific flow to a user agent has failed, although other flows may succeed.
  • 433 Anonymity Disallowed – The request has been rejected because it was anonymous.
  • 436 Bad Identity Info – The request has an identity info header and the URI scheme contained cannot be de-referenced.
  • 437 Unsupported Certificate – The server was unable to validate a certificate for the domain that signed the request.
  • 438 Invalid Identity Header – Server obtained a valid certificate used to sign a request, was unable to verify the signature.
  • 439 First Hop Lacks Outbound Support – The first outbound proxy doesn’t support “outbound” feature.
  • 470 Consent Needed – The source of the request did not have the permission of the recipient to make such a request.
  • 480 Temporarily Unavailable – Callee currently unavailable.
  • 481 Call/Transaction Does Not Exist – Server received a request that does not match any dialog or transaction.
  • 482 Loop Detected – Server has detected a loop.
  • 483 Too Many Hops – Max-forwards header has reached the value ‘0’.
  • 484 Address Incomplete – Request-URI incomplete.
  • 485 Ambiguous – Request-URI is ambiguous.
  • 486 Busy Here – Callee is busy.
  • 487 Request Terminated – Request has terminated by bye or cancel.
  • 488 Not Acceptable Here – Some aspects of the session description of the request-URI are not acceptable.
  • 489 Bad Event – The server did not understand an event package specified in an event header field.
  • 491 Request Pending – Server has some pending request from the same dialog.
  • 493 Undecipherable – Undecipherable request contains an encrypted MIME body, which recipient can not decrypt.
  • 494 Security Agreement Required – The server has received a request that requires a negotiated security mechanism.

5xx = Server Errors

  • 500 Server Internal Error – The server could not fulfil the request due to some unexpected condition.
  • 501 Not Implemented – The SIP request method is not implemented here.
  • 502 Bad Gateway – The server, received an invalid response from a downstream server while trying to fulfil a request.
  • 503 Service Unavailable – The server is in maintenance or is temporarily overloaded and cannot process the request.
  • 504 Server Time-out – The server tried to access another server while trying  to process a request, no timely response.
  • 505 Version Not Supported – The SIP protocol version in the request is not supported by the server.
  • 513 Message Too Large – The request message length is longer than the server can process.
  • 580 Precondition Failure – The server is unable or unwilling to meet some constraints specified in the offer.

6xx = Global Failures

  • 600 Busy Everywhere – All possible destinations are busy.
  • 603 Decline – Destination cannot / doesn’t wish to participate in the call, no alternative destinations.
  • 604 Does Not Exist Anywhere – The server has authoritative information that the requested user does not exist anywhere.
  • 606 Not Acceptable – The user’s agent was contacted successfully but some aspects of the session description were not acceptable.

What are SIP Phones?

Simply put, a SIP phone is a phone that uses the Open Standard SIP to set up and manage phone calls. The actual voice is carried over an IP-based network using another Open Standard called RTP. Since these protocols are generically termed VoIP (voice over internet protocol), these phones are also sometimes called VoIP phones or VoIP clients.

SIP phones can be classified in 2 main categories:

  • Hardphones or desk phones or hardware SIP phones
  • Softphones or software SIP phones


A hardphone looks like a regular telephone, and indeed behaves as one. However, the hardware is built using network-aware, or more specifically, IP-aware components. The hardphone will connect to an IP-Network using regular Ethernet cables or using WiFi. Cordless hardphones are also available, and these devices take another industry standard cordless technology called DECT, so that the phones communicate with a base station using the DECT protocol, while the base station communicates with an IP-PBX using SIP and RTP as their transport protocols.


A softphone is quite simply what their name implies – a software program that provides telephone functionality. Again, a softphone will, just like a hardphone, use the Open Standards protocols SIP and RTP for call setup and voice delivery. Any computing device such as:

  • Desktop computers (Windows, Mac, Linux)
  • Tablets (Android, iOS)
  • Smartphones (Android, iOS)

…can run softphone programs, providing a very wide array of options from which to choose. Any computer or smart device that has a microphone and speakers (or a headset) can double up as a softphone. The only pre-requisite is an IP-based connection a 3CX system, like the one of C2 Communications.

Softphone Benefits

Using a softphone allows us to make better use of computing resources, but a more important benefit is actually the fact that it is software-based. The functionality that can be added to a softphone is limited to the software developer’s imagination, allowing him to create powerful visual tools for the user, integrate into other systems using the softphone itself as the intermediary, and so on…

The 3CX software used by C2 Communications can be used with most popular hardware SIP phones, but also comes with a completely free software-based SIP phone that showcases the benefits of a softphone, with extended functions which are not possible to achieve with a hardphone.

SIP Call Session Between 2 Phones


A SIP call session between two phones is established as follows:

  1. The calling phone sends out an INVITE.
  2. The called phone sends an information response 100 – Trying – back.
  3. When the called phone starts ringing a response 180 – Ringing – is sent back.
  4. When the caller picks up the phone, the called phone sends a response 200 – OK.
  5. The calling phone responds with ACK – acknowledgement.
  6. Now the actual conversation is transmitted as data via RTP.
  7. When the person calling hangs up, a BYE request is sent to the calling phone.
  8. The calling phone responds with a 200 – OK.

It’s as simple as that! The SIP protocol is logical and very easy to understand.

What SIP-Based IP PBX’s are Available?

An IP PBX or VoIP phone system replaces traditional PBX or phone systems, giving employees an extension number, the ability to conference, transfer and dial other colleagues. All calls are sent via data packets over a data network instead of the traditional phone network. With the use of a VoIP gateway, you can connect existing phone lines to the IP PBX and make and receive phone calls via a regular PSTN line. The IP PBX FAQ helps answer common questions about VoIP, SIP, IP PBX / VoIP phone system hardware and software, implementation and more.

This list shows some of the currently available SIP-based IP PBXs in the telco sector:

  • 3CX Phone System – a cross-platform IP PBX that runs on Windows and Linux, trusted by C2 Communications. 3CX offers benefits over other vendors, such as:
    • easily deployed both in-house and in-cloud;
    • automation for in-cloud deployment to top-tier hosting platforms such as Google, Azure and Amazon;
    • best-of-breed phone provisioning automation;
    • robust feature-set, including:
      • IVR,
      • voicemail,
      • queues,
      • groups,
      • user-based and group-based rights management,
      • and security and anti-hacking.
  • Asterisk®* – a Linux-based IP PBX
  • sipX – a Linux-based IP PBX
  • Elastix – a Windows and Linux-based PBX available On-premise or in the cloud.
  • and others.

A number of open source alternatives are available, but these all suffer from:

  • hard-to-use interfaces,
  • poor or non-existent provisioning tools,
  • limited or non-existent commercial support,
  • feature sets not driven by market needs, but by the wants of the individuals that contribute the code.

What is a SIP Server?


A SIP server is the main component of our IP 3CX, and mainly deals with the management of all SIP calls in the network. A SIP server is  also referred to as a SIP proxy or a registrar. Although the SIP server can be considered the most important part of a SIP-based IP-3CX phone system, it only handles or manages sessions; more specifically, a SIP server can:

  • Set up a session between two (or more) endpoints (an audio conference would have more than two endpoints)
  • Negotiate the media parameters and specifications for the session for each endpoint using the SDP protocol
  • Adjust the media parameters and specifications of a session during the session (putting a call on hold, for example)
  • Substituting one endpoint with a new endpoint (call transfer)
  • Terminate a session

The SIP server does not actually transmit or receive any media – this is done by the media server in using the RTP protocol. Within the context of an IP-3CX environment, it is almost always true that the SIP server and its Media server companion reside on the same machine. Do keep in mind, however, that very-high-volume SIP servers (such as a large VoIP provider, for example), may separate their media server to a different machine to better handle the workload, and could also possibly distribute the load to multiple media servers.

SIP Phones / VoIP Phones Types

A VoIP phone system requires the use of SIP phones / VoIP phones. SIP phones come in several versions and types:

SIP / VoIP Soft Phones – Software-Based SIP Phone


A software-based SIP phone is an application which makes use of your computer’s microphone and speakers or an attached headset to allow you to make or receive calls. An example of a SIP phone is 3CX’s own SIP clients,  which are free to use for all 3CX 12 and above users.

Hardware SIP Phone

A hardware-based SIP phone looks and behaves just like a normal phone. However, it is connected directly to the data network, rather than to standard PSTN line(s). These phones have an integrated mini hub, so that they can share the network connection with the computer, which means you don’t need an additional network point for the phone. Examples of hardware SIP phones are snom and Yealink IP phones which work seamlessly with 3CX software used by C2 Communications.

Use an Analogue Phone via an ATA Adapter

If you want to use your current phone with the 3CX VoIP phone system from C2 Communications, you can use an ATA adapter. An ATA adapter allows you to plug in the Ethernet network jack into the adapter and then plug the phone into the adapter. Your old phone will appear to the 3CX phone system software as a regular SIP phone.

VoIP phones are very inexpensive to buy and can bought online via one of the many VoIP product online shops. 3CX  supports all popular VoIP phones as it’s based on the Open SIP Standard. You can even automatically provision most IP phone models too.

What is SIP?

SIP (Session Initiation Protocol) is a signalling protocol used to establish a session between 2 or more participants, modify that session, and eventually terminate that session. It has found its major use in the world of IP telephony. The fact that SIP is an open standard has sparked enormous interest in the telephony market, and manufacturers shipping SIP-based phones have seen tremendous growth in this sector.

The SIP protocol is text-based, and bears significant resemblance to the HTTP protocol. The messages are text-based, and the request-response mechanism makes for easier troubleshooting. The actual data transmission is done by the Transmission Control Protocol (TCP) or the User Datagram Protocol (UDP) on layer 5 of the OSI model. The Session Description Protocol (or SDP) controls which of the protocols is used.

The SIP messages describe the identity of the participants in a call, and how the participants can be reached over an IP network. Encapsulated inside the SIP messages we can sometimes also see an SDP declaration. SDP (Session Description Protocol) will define the type of media channels that will be established for the session – typically this will declare which codecs are available, and how the media engines can reach each other over an IP network.

Once this exchange of setup messages is completed, the media is exchanged using yet another protocol, typically RTP (Real-Time Transmission Protocol). SIP was developed by the IETF and published as RFC 3261, and its flexibility has allowed it to replace almost completely the H.323 protocol in the VoIP world.

What is SIP Forking?

SIP forking refers to the process of “forking” a single SIP call to multiple SIP endpoints. This is a very powerful feature of SIP. A single call can ring many endpoints at the same time.



With SIP forking you can have your desk phone ring at the same time as your softphone or a SIP phone on your mobile. For example, you would use SIP forking to ring your desk phone and your Android SIP phone at the same time, allowing you to take the call from either device easily. No forwarding rules would be necessary as both devices would ring. In the same manner SIP forking can be used in an office and allow the secretary to answer calls to the extension of his/her boss when he is away or unable to take the call. 3CX software from C2 Communications fully supports SIP forking.

SIP Methods / Requests and Responses?

SIP Methods / Requests and Responses

SIP uses methods / requests and corresponding responses to communicate and establish a call session.



SIP Requests

There are fourteen SIP request methods of which the first six are the most basic request / method types:

  • INVITE = Establishes a session.
  • ACK = Confirms an INVITE request.
  • BYE = Ends a session.
  • CANCEL = Cancels establishing of a session.
  • REGISTER = Communicates user location (host name, IP).
  • OPTIONS = Communicates information about the capabilities of the calling and receiving SIP phones.
  • PRACK = Provisional acknowledgement.
  • SUBSCRIBE = Subscribes for notification from the notifier.
  • NOTIFY = Notifies the subscriber of a new event.
  • PUBLISH = Publishes an event to the server.
  • INFO = Sends mid session information.
  • REFER = Asks the recipient to issue call transfer.
  • MESSAGE = Transports instant messages.
  • UPDATE = Modifies the state of a session.

SIP Responses

SIP requests are answered with SIP responses, of which there are six classes:

1xx = Informational responses, such as 180 (ringing).
2xx = Success responses.
3xx = Redirection responses.
4XX = Request failures.
5xx = Server errors.
6xx = Global failures.

What are SIP Trunks?

The trusted old Public Switched Telephone Network (PSTN), with its analogue lines, ISDN BRI, E1 or t1 lines, is to disappear. Telephony is moving from PSTN to much more modern and flexible SIP trunks. The big telecom providers are fast phasing out the old PSTN functionality, and are moving customers to IP. And so a SIP trunk and a phone system upgrade in the near future is going to be inevitable. Phone companies like Verizon will phase out ISDN in the U.S. by 2018. In the UK, ISDN lines are down to less than 3 million lines, from 4.7 million lines in 2007 and the trend is accelerating. In 2017 major telcos such as BT, KPN, France Télécom, Deutsche Telekom and Telecom Italia began to phase out ISDN lines.

As a result Session Initiation Protocol (SIP) trunking has increased by 62 percent in 2015 from the prior year, driven primarily by North America. The SIP trunking service is usually provided by an internet service provider (ISP). Unlike a PTSN provider, the lines provided are not physical lines, but a service provided over your internet connection. The SIP trunk provider provides phone numbers and lines, usually at better rates than the traditional providers and with more flexibility and shorter contract durations. This guide explains what SIP trunks are, their advantages and how you can make the move.

What are SIP Trunks?

SIP trunks are phone line trunks delivered over IP using the SIP protocol. Using this standard protocol, telecom service (VoIP) providers connect one or more channels to the customer’s PBX. Phone numbers and DIDs are linked to the SIP trunk. In many cases numbers can be ported to the SIP trunk.

Benefits of SIP Trunking

Our farewell to the PSTN brings many benefits. SIP trunks deliver:

  • Lower monthly line & DID rental – The monthly fee to have a number of lines installed at your office drops significantly with SIP trunks. And DIDs cost a lot less.
  • Lower call charges – With C2 Communication, SIP trunking results in very low call charges. We also offer unlimited call packages.
  • Better customer service – Provide better customer service by adding more geographical and international numbers. Quickly and easily add numbers to your SIP trunk and terminate them on your IP 3CX – you can give customers more options to dial in at a significantly lower cost, even long distance. Customers can contact you more easily and sales will increase.
  • Move offices and keep the same number – SIP trunks are not bound to a location, so it’s easy to move offices without having to change your stationary or inform your customers. There is no longer any need to pay to forward phone calls to the new offices.
  • Eliminate VoIP Gateways – SIP trunks will eliminate the need to buy and manage VoIP Gateways. All phone calls come in via IP. No extra conversion often means better quality too.
  • Leverage a modern IP PBX – Modern IP 3CX / unified communications solutions give customers increased productivity, mobility and boost sales. Connecting an IP 3CX to SIP trunks is much easier than via the PSTN. You can go on-premise or hosted with C2 Communications, the choice is yours.
  • Flexibility – It is easy to add channels to your SIP trunk to cope with increased calls. A simple phone call will allow you to add channels, and often this can be done immediately.
  • Correct number of channels – With SIP trunks, you can easily choose the correct number of channels that you need. Using ISDN / T1, you often have to choose to add either 15 or 30 lines. This usually means you end up with expensive extra capacity.

Selecting the Right SIP Trunk Provider

The next step is to choose a SIP trunk provider who will supply the necessary SIP Trunks. A few factors come into play when making this decision:

  • Security – As SIP trunks are exposed to the Internet, it is very important that the SIP trunk has a well secured network and an anti-fraud system in place. The anti-fraud system must monitor the system and provide protection against call fraud.
  • Own network – Does the SIP trunk provider run its own network or is it a rebranded service? There are quite a few providers out there reselling SIP trunks from other providers. Select a provider who has control over their service and network.
  • Competitive cost – Costs vary widely between services. Some vendors will overcharge for SIP trunks. Look for competitive rates, but ensure that you are getting business quality SIP trunk service. For example, telecom providers will provide a cheaper quality to Internet call shops. Be cost conscious, but expect to pay a bit more for business class service.
  • Number porting – Can the provider port your phone numbers? Ensure that you choose a provider who can port all the existing numbers – not all providers are able to do this for all regions.

With C2 Communications, you select an experienced and highly secure SIP trunk provider, who offers you the highest flexibility for lowest costs, as well as outstanding customer service and lots of useful features.

Upgrading Internet Connectivity

Once you have selected your SIP trunk provider, consider a dedicated Internet line for the SIP trunk. Most firewalls are able to handle multiple WAN connections, and, considering the low cost of an Internet line in most places, a separate VoIP connection will be the most reliable way to ensure the quality of your VoIP calls.

However, some SIP trunk providers bundle their service with a dedicated Internet line. This keeps your voice traffic separate from your data traffic. Much will depend on the cost and your network infrastructure. Check that your firewall is up-to-date and will be capable of handling VoIP traffic correctly.

Upgrading the PBX to an IP 3CX

Chances are that the trusted old PSTN lines are connected into another old device, the hardware-based PBX system. This device is inflexible, difficult to manage and often expensive to maintain. Technically it is possible to buy a gateway that allows the old PBX to talk to the SIP trunks. But why not upgrade to a modern IP 3CX with C2 Communications and leverage the flexibility and modern features IP telephony can bring to your business phone system. This allows you to take advantage of the cost savings, easy management, and productivity increases with full-scale unified communications that the 3CX offers. You can choose from a hosted 3CX, an appliance 3CX, or a software-based PBX.

What is a SIP-URI?

A SIP-URI is the SIP addressing scheme that communicates who to call via SIP. In other words, a SIP URI is a user’s SIP phone number. The SIP URI resembles an e-mail address and is written in the following format:


SIP-URI = sip:x@y:Port        where x=username and y=host (domain or IP)


Note: If you do not specify a port, the default SIP port will be assumed (5060). This is shown in the first two examples below. If you have changed the default SIP port to something else, then you need to specify it in the SIP-URI (third example below).


  1. sip:[email protected]
  2. sip:[email protected]
  3. sip:[email protected]:6000

The SIP URI scheme has been defined in the RFC 3261 standard. 3CX from C2 Communications uses SIP URIs.


What is a 3CX Phone System?

A 3CX phone system is a PBX, which stands for Private Branch Exchange. This is a private telephone network used within a business. The users of the PBX phone system can communicate internally (within their company) and externally (with the outside world), using different communication channels like Voice over IP, ISDN or analogue. A 3CX also allows you to have more phones than physical phone lines (PTSN) and free calls between users. Additionally, it provides features like transfer calls, voicemail, call recording, interactive voice menus (IVRs) and call queues.

Time and technology have changed the consumer telephony landscape in the past years, with the flag-bearer being the Open-Standards-based IP 3CX. Now you can telephone via the Internet Protocol technology. 3CX phone systems are available as hosted or virtual (cloud) solutions and as on-premise solutions on your own hardware.


what is a pbx system


This image gives us an idea of what a 3CX system allows in terms of connectivity and reachability. With a traditional PBX, you are typically constrained to a certain maximum number of outside telephone lines (trunks) and to a certain maximum number of internal telephone devices or extensions. Users of the PBX phone system (phones or extensions) share the outside lines for making external phone calls.

Switching to a 3CX brings many benefits and opens up possibilities, allowing for almost unlimited growth in terms of extensions and trunks, and introducing more complex functions that are more costly and difficult to implement with  a traditional PBX, such as ring groups, queues, digital receptionists, voicemail and reporting. C2 Communications relies on 3CX, as it has established itself as the leading IP-PBX manufacturer, ticking all the checkboxes for any business looking for enterprise-grade features.

What are IP Phones / IP Telephones?

An IP telephone, very broadly speaking, is a telephone designed to work with an IP PBX. Almost all IP phones nowadays are SIP-based phones.

This is good news for businesses and end users because it means that PBX vendors can’t force you into locking in with their proprietary software or hardware. Phone manufacturers can produce SIP phones, and as long as the phones can support the SIP standard, then your investment is protected, and the phones can be used with virtually any SIP-based IP PBX. The 3CX software is tested against all major brands to ensure compatibility, such as:

  • Aastra
  • Cisco
  • Fanvil
  • Grandstream
  • Htek
  • Polycom
  • Snom
  • Yealink

C2 Communications also offers a free 3CX softphone for Windows that can be used to make and receive VoIP phone calls from your PC. With the 3CX softphone for Windows, your costs are strongly reduced and you can enjoy free VoIP calls.

Today’s modern SIP Phones are also able to work without being constrained to a particular office or location, which makes them more mobile. We now can have on-premise phones and remote ones working off of the same systems. Additionally, you can use the 3CX mobile apps in order to receive and make calls over the 3CX phone system, no matter where you are. Thanks to the great connectivity of 3CX, the collaboration between team members is enhanced.

IP phones are sometimes called VoIP telephones, SIP phones or softphones. Although they have different name, it’s all the same device or software client that is designed to transmit voice over the internet (VoIP), mostly using the SIP protocol. IP telephones come in different types for different user roles. 3CX software supports the most popular IP phones.

What is a STUN Server?

A STUN (Session Traversal of User Datagram Protocol [UDP] Through Network Address Translators [NATs]) server allows NAT clients (i.e. IP phones behind a firewall) to setup phone calls to a VoIP provider hosted outside of the local network.

The STUN server allows clients to find out their public address, the type of NAT they are behind and the internet side port associated by the NAT with a particular local port. This information is used to set up UDP communication between the client and the VoIP provider to establish a call. The STUN protocol is defined in RFC 3489.

The STUN server is contacted on UDP port 3478, however the server will hint clients to perform tests on alternate IP and port number too (STUN servers have two IP addresses). The RFC states that this port and IP are arbitrary. STUN functionality is seamlessly handled by the 3CX software that C2 Communications relies on.

What is a VoIP Phone?

VoIP phones are connected to an IP phone system using LAN (Local Area Network) or the Internet. VoIP converts the standard telephone audio into a digital format which can be transmitted over the internet and also converts incoming digital phone signals coming from the internet to standard telephone audio. A VoIP telephone allows users to make phone calls using VoIP, to any softphone, mobile or landline. A VoIP telephone can be a simple, software-based softphone or a hardware device.

Hardware SIP phones look very similar to traditional analogue phones, however are developed to be used in IP networks. Softphones or smartphone apps, on the other hand, are software applications that are used on your computer, tablet or smartphone and behave exactly like a normal IP telephone; allowing you to carry out regular functions such as making and receiving calls, as well as some additional functions such as video conferences, presentation giving and more.

Most IP phones use SIP (Session Initiation Protocol), whether they are hardware IP phones or softphones. Some well-known manufacturers of IP phones include Fanvil, Htek, Snom and Yealink. The 3CX software used by C2 Communications includes high-performing and feature-rich softphones for Windows and Mac as well as smartphone apps for iOS and Android for free.

The 3CX used by C2 Communications supports auto-provisioning of a wide range of popular IP phones sold worldwide. Furthermore, we also provide legacy support to numerous older or proprietary devices including Cisco.

VoIP Phone Features

Some of the common features of a VoIP telephone are: caller ID, call park, call transfer, call hold, phonebook access and multiple account configuration. Some VoIP phones also allow the transmission of video along with audio during calls. C2 Communications offers a completely free VoIP telephone for use with the 3CX phone system.

What is IP Telephony?

IP telephony (Internet Protocol telephony) is describing technologies that use the IP protocol to exchange voice, fax, and other forms of information, traditionally carried over the Public Switched Telephone Network (PSTN). The call travels in the form of packets, over a Local Area Network (LAN), or the Internet, avoiding PSTN tolls.

Starting in the mid to late 1990s, the Internet and the TCP/IP protocol began to change the telephone and communications industry. The Internet Protocol becomes the transport for almost all data communications. Today, all communication carriers are using an IP infrastructure for a part or for all of its voice services. With C2 Communications, you can use VoIP for your internal voice communications or you can implement it as part of your unified communication solution.

What is Voice over IP?

Voice over IP is short for Voice over Internet Protocol, and is better known as VoIP.

Voice over IP refers to the transmission of voice traffic over internet-based networks instead of the traditional PSTN (Public Switched Telephone Network) telephone networks. The Internet Protocol (IP) was originally designed for data networking and following its success, the protocol has been adapted to voice networking by packetizing the information and transmitting it as IP data packets. VoIP is now available on many smartphones, personal computers and on internet access devices such as tablets.

VoIP can facilitate tasks and deliver services that might be cumbersome or costly to implement when using traditional PSTN:

  • More than one phone call can be transmitted on the same broadband phone line. This way, voice over IP can facilitate the addition of telephone lines to businesses without the need for additional physical lines.
  • Features that are usually charged extra by telecommunication companies, such as call forwarding, caller ID or automatic redialing, are simple with voice over IP technology.
  • Unified Communications are secured with voice over IP technology, as it allows integration with other services available on the internet such as video conversation, messaging, etc.

These and many other advantages of VoIP are making businesses adopt our C2 Communications VoIP phone systems at a staggering pace.

VoIP Defined

VoIP is an acronym for Voice Over Internet Protocol, which by itself means voice over the internet. It’s a technology that delivers voice communication and multimedia sessions (such as video) over Internet Protocol (IP) networks.

Initial VoIP service providers offer solutions that mirror the architecture of the legacy telephone network whereas second- and third-generation providers have either built closed networks for private user bases, offering free calls or have completely departed from the legacy telephone network architecture. VoIP solutions allow a dynamic interaction between users on any two domains on the Internet when a user wishes to place a call. To place calls via VoIP, a user will need a software-based SIP phone program or a hardware-based VoIP phone. Phone calls can be made to anywhere and to anyone: Both to VoIP numbers as well as PSTN phone numbers.

Businesses that choose to use our VoIP systems instead of traditional copper-wire telephone systems experience many benefits such as reducing their monthly phone costs, increased mobility and productivity among others. In 2008, 80% of all new PBX lines installed internationally were VoIP.

VoIP solutions aimed at businesses also include Unified Communications features which include web conferencing, presence, fax and voicemail to email, instant chat and more as well as smartphone clients so that employees can take their office extension with them wherever they go. The smartphone clients also use VoIP in order to make and receive calls from a user’s cell phone as if they were using their office extension number.

Learn About VoIP Gateways

A VoIP gateway (or PSTN Gateway) is a device which converts telephony traffic into IP for transmission over a data network. They are used in two ways:

1. To Convert Incoming PSTN / Telephone Lines to VoIP / SIP:

The VoIP gateway allows calls to be received and placed on the regular telephony network. In many business cases, it is preferable to continue to use traditional phone lines because one can guarantee a higher call quality and availability.

2. To Connect a Traditional PBX / Phone System to the IP Network:

The VoIP gateway allows calls to be made via VoIP. Calls can then be placed via a VoIP service provider, or in the case of a company with multiple offices, interoffice calling costs can be reduced by routing the calls via the Internet. VoIP gateways are available as external units or as PCI cards. The vast majority of devices are external units. A VoIP gateway has a connector for the IP network and one or more ports to connect the phone lines to it.


Types of VoIP Gateways

1. Analogue units: Analogue units are used to connect regular analogue phone lines to the gateway. Analogue units are available for between 2-24 lines.

2. Digital units: Digital units allow you to connect digital lines, either one or more BRI ISDN lines (Europe), one or more PRI / E1 lines (Europe) or one or more T1 lines (USA).

VoIP Gateway Manufacturers

There are many VoIP gateways available today, and as demand has increased drastically, prices have decreased considerably. Our 3CX software automatically configures VoIP Gateways to allow you to easily continue using your existing PSTN lines.

VoIP Definitions

This FAQ lists all the popular VoIP definitions.

  • VoIP – Voice over Internet Protocol (also called IP telephony, internet telephony, and digital phone) is the routing of voice conversations over the Internet or any other IP-based network.
  • SIP – Session Initiation Protocol is a protocol developed by the IETF MMUSIC Working Group and proposed standard for initiating, modifying, and terminating an interactive user session that involves multimedia elements such as video, voice, instant messaging, online games, and virtual reality.
  • PSTN – Public Switched Telephone Network is the concentration of the world’s public circuit-switched telephone networks, in much the same way that the Internet is the concentration of the world’s public IP-based packet-switched networks.
  • ISDN – Integrated Services Digital Network is a type of circuit switched telephone network system, designed to allow digital (as opposed to analogue) transmission of voice and data over ordinary telephone copper wires, resulting in better quality and higher speeds, than available with analogue systems.
  • PBX – Private Branch eXchange (also called Private Business eXchange) is a telephone exchange that is owned by a private business, as opposed to one owned by a common carrier or by a telephone company.
  • IVR – In telephony, Interactive Voice Response is a computerised system that allows a person, typically a telephone caller, to select an option from a voice menu and otherwise interface with a computer system.
  • DID – Direct Inward Dialing (also called DDI in Europe) is a feature offered by telephone companies for use with their customers’ PBX system, whereby the telephone company (telco) allocates a range of numbers all connected to their customer’s PBX.
  • RFC – Request for Comments (plural Requests for Comments – RFCs) is one of a series of numbered Internet informational documents and standards very widely followed by both commercial software and freeware in the Internet and Unix communities.

What is ALG?

ALG or Application Layer Gateway is a software component that manages specific application protocols such as SIP (Session Initiation Protocol) and FTP (File Transfer Protocol). An ALG acts as an intermediary between the Internet and an application server that can understand the application protocol. The ALG appears as the end point server and controls whether to allow or deny traffic to the application server. It does this by intercepting and analysing the specified traffic, allocating resources, and defining dynamic policies to allow traffic to pass through the gateway.

An ALG has the following functions:

  • It allows client applications to use dynamic TCP / UDP ports to communicate with known ports used by server applications, even if the firewall configuration allows traffic through only a limited number of ports. Without an ALG, the ports would either get blocked, or the network administrator would need to open up a large number of ports in the firewall, weakening the network and allowing potential attacks on those ports.
  • It recognises application specific commands and offers security controls over them.
  • It can convert the network layer address information that is found in an application payload.
  • Synchronises multiple streams or sessions between hosts.

What is a Good Source of VoIP Information?

3CX has a lot of useful information about VoIP on its website. Check out these VoIP information sources:

Another good starting point is at Wikipedia: Voice Over IP

Your Account

How do I port my numbers to C2comm?

Porting is easy, simply email [email protected] with:

  • A copy of your latest telephony bill from your current carrier
  • And list the numbers you wish to port to us

Once received we will lodge the port on your behalf and notify you when to expert the port completion.

How long does it take to port numbers?

This is dependant on your current carrier. Typically, when porting from the traditional carriers the time frame is 4 to 6 weeks. When porting from a VoIP carrier, the time frame is 1 to 2 weeks.

Porting takes so long because of the administration involved to change the registration and routing within the telco network.

Is my VoIP service under contract?

We provide month-to-month PAYG service on all numbers and lines. Depending on the service you have signed up to with us the typical answer is no, you are not under contract. Almost all of C2 Communications customers are on monthly pay as you go services. This is especially relevant for our SIP services however it may vary depending on your situation. If you would like to find out about your services, get in contact with us and we will be happy to help you out!

How do I order more numbers and lines?

To order more numbers and lines send us an email to [email protected] with:

  • Site address for the new numbers
  • Required lines
  • And, whether or not this is for a new or existing trunk.

A new number will take 24 hrs to provision. A temporary number will take 1 hr to provision. Extra lines will take 1 hr to provision.

Can I host my 1300 / 1800 numbers with C2comm?

Yes, you can! Hosting your 1300 / 1800 numbers with C2 Communications is easy and allows you to received one bill for all of your phone bills. If you have a toll-free number already or wish to purchase one, get in contact with us at [email protected] and we will be delighted to help you out!

How do I cancel my account with C2comm?

Your SIP services can be cancelled at any time. As discussed earlier, almost all of our customers are on monthly pay as you go services. Therefore, if you no longer require your SIP, simply email [email protected] with service numbers and lines you wish to cancel.

Cancellations require 30 days’ notice.