FAQs

Below is a list of Frequently Asked Questions (FAQ’s). Click on the Question to reveal the answer.

Does VoIP Business Solutions provide Internet service?

No, you’ll need to get a high-speed Internet connection through a third-party internet provider such as Verizon or Comcast.

Does my computer need to be on to make or receive calls?

No, your VoIP phone system from VoIP business Solutions is a complete turnkey system that is independent from your computer. You do need a high-speed Internet connection, which you can still use if your computer is off.

Can I keep my existing number?

In virtually all cases, yes, you can keep your existing telephone number.

What is an auto-attendant?

An auto-attendant (or automated attendant) is a term commonly used in telephony to describe 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 or simply a “phone tree”. 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 auto-attendant directs callers to the appropriate voice mailbox of the user to leave a voicemail 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 calls procedure.

 

What different types of CODECS are there?

A Codec converts an analog signal to a digital one for transmission over a data network. The following Codecs are in use today

  • GSM – 13 Kbps (full rate), 20ms frame size
  • iLBC – 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size
  • ITU G.711 – 64 Kbps, sample-based. Also known as alaw/ulaw
  • ITU G.722 – 48/56/64 Kbps
  • ITU G.723.1 – 5.3/6.3 Kbps, 30ms frame size
  • ITU G.726 – 16/24/32/40 Kbps
  • ITU G.728 – 16 Kbps
  • ITU G.729 – 8 Kbps, 10ms frame size
  • Speex – 2.15 to 44.2 Kbps
  • LPC10 – 2.5 Kbps
  • DoD CELP – 4.8 Kbps

What is DID – Direct Inward Dialing?

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

What does ENUM mean?

ENUM stands for Telephone Number Mapping. Behind this ‘abbreviation’ hides a great idea: To be reachable anywhere in the world with the same number – and via the best and cheapest route. ENUM takes a phone number and links it to an internet address which is published in the DNS system.  The owner of an ENUM number can thus publish where a call should be routed to via a DNS entry. What’s more, different routes can be defined for different types of calls – for example you can define a different route if the caller is a fax machine. ENUM does require the phone of the caller to support it. You register an ENUM number rather like you register a domain. At present many registrars and VOIP providers are providing this as a free service. ENUM is a new standard, and is not that widespread yet, though it looks set to become another revolution in communications and personal mobility.

What is Echo cancellation?

Echo cancellation is the process of removing echo from a voice communication in order to improve the voice call quality. Echo cancellation is often needed because speech compression techniques and packet processing delays generate echo. There are 2 types of echo: acoustic echo and hybrid echo. Echo cancellation not only improves quality but it also reduces bandwidth consumption because of its silence suppression technique.

What is T38?

T38

T38 is a protocol that describes how to send a fax over a computer data network. T38 is needed because fax data cannot be sent over a computer data network in the same way as voice communication. In essence, with T38 a fax is converted to an image, sent to the other T38 fax device and then converted back to an analog fax signal. Most VoIP Gateways and ATA’s now support T38 reliably. T38 is described in RFC 3362, and defines how a device should communicate the fax data. In the picture above both the gateway and the fax machine behind the gateway would have to be capable of T38. For the G3 fax machine on an analog line, this process will be transparent. The analog fax machine does not need to know T38. FAX was designed for analog networks, and does not travel well over a VOIP network. The reason for this is that FAX communication uses the signal in a different way to regular voice communication. When VOIP technologies digitize and compress analog voice communication it is optimized for VOICE and not for FAX. Subsequently, there are a number of things you need to take note of when you move to a VoIP Phone System. If you want to continue using your old fax machine, and you want to connect to your VoIP phone system, its best to use a VoIP Gateway and an ATA that supports T38. T38 is a protocol designed to allow fax to ‘travel’ over a VoIP network. An example configuration of such a setup can be found here. It is also possible to convert to computer based fax and choose a VoIP phone system that supports fax. Another way to deal with fax when you switch to voip are to connect the fax machine directly to the existing analog phone line and bypass your VOIP system.

How does FAX work in VoIP environments?

FAX was designed for analog networks, and does not travel well over a VOIP network. The reason for this is that FAX communication uses the signal in a different way to regular voice communication. When VOIP technologies digitize and compress analog voice communication it is optimized for VOICE and not for FAX. Subsequently, there are a number of things you need to take note of when you move to a VoIP Phone System. If you want to continue using your old fax machine, and you want to connect to your VoIP phone system, its best to use a VoIP Gateway and an ATA that supports T38. T38 is a protocol designed to allow fax to ‘travel’ over a VoIP network. It is also possible to convert to computer based fax and choose a VoIP phone system that supports fax. Another way to deal with fax when you switch to voip are to connect the fax machine directly to the existing analog phone line and bypass your VOIP system.

What are IP Phones?

IP Phones are sometimes called VoIP telephones, SIP phones or soft phones. These are all the exact same thing and are based on the principle of transmission of voice over the internet, or what is better known as VoIP (or voice over internet protocol) technology. IP Telephones come in several types. Learn more about the different kinds of IP phones. Most popular IP Phones are supported by VoIP Business Solutions including those from Cisco, Linksys, Snom, Aastra, and Polycom to name a few.

What are SIP Phones?

SIP Phones are the same thing as VoIP Phones or soft phones. These are telephones that allow phone calls to be made using VoIP (voice over internet protocol) technology. There are two types of SIP Phones. The first type is the hardware SIP phone, which resembles the common telephone but can receive and make calls using the internet instead of the traditional PSTN system. SIP Phones can also be software-based. These allow any computer to be used as a telephone by means of a headset with a microphone and/or a sound card. A broadband connection and connection to a VOIP provider or a SIP server are also required. The system from VoIP Business Solutions can be used with most popular hardware SIP Phones. It also comes with a completely FREE software-based SIP Phone that functions as a VoIP Client.

What is SIP?

SIP, short for Session Initiation Protocol is an IP telephony signaling protocol used to establish, modify and terminate VOIP telephone calls. SIP was developed by the IETF and published as RFC 3261. SIP describes the communication needed to establish a phone call. The details are then further described in the SDP protocol. SIP has taken the VOIP world by storm. The protocol resembles the HTTP protocol, is text based, and very open and flexible. It has therefore largely replaced the H323 standard.

What is IVR?

Interactive Voice Response or IVR is a telephone technology that communicates with a caller through configurable voice menus and data in real time. In an IVR system, callers are given the choice to select options by pressing digits. 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 information they need 24 hours a day without the need of costly human agents. Some IVR applications include telephone banking, flight-scheduling information and tele-voting. VoIP Business Solutions 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. VBS helps businesses enhance their communications with an interactive voice response system, built into its PBX. IVR is a telephone technology that is normally a pricey addition to a phone system. Despite this, VBS delivers it as a standard built-in feature. With its IVR system it is possible for pre-recorded audio files or dynamically generated TTS (text to speech) audio to explain options available to callers. The caller responds by pressing a digit or a combination of digits. Interactive Voice Response technology allows businesses to supply callers with around the clock information without the need of a human receptionist or agent. This allows for cost savings and improvements in customer satisfaction. The VBS Phone System IVR offers a complete IP Telephony solution that is scalable, feature-rich and at no extra cost.

What is the difference between Microsoft Response Point & VoIP Business Solutions?

Microsoft Response Point and the VBS phone system are both IP PBXs that replace the traditional PBX and can improve business communications and help cut costs. But the two phone systems differ in the delivery of some important advantages. The main differences between Microsoft Response Point and system from VoIP Business Solutions are: Microsoft Response Point ties your business to specific hardware. On the contrary, our system works with most popular SIP and IP phones, allowing you to choose the hardware that best suits your business’ needs and budget. Microsoft Response Point is focused on a small number of core capabilities. In the case of our phone system businesses can enjoy a full set of enterprise level features at a very low cost. Microsoft Response Point is limited to 50 users. The system from VoIP Business Solutions can be used with an unlimited number of extensions.

Why should I switch to an IP PBX?

An IP PBX 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 that all enterprises seek. The IP PBX is also able to connect to traditional PSTN lines via an optional gateway – so upgrading day-to-day business communication to this most advanced voice and data network is a breeze! Enterprises don’t need to disrupt their current external communication infrastructure and operations. With IP PBX deployed, an enterprise can even keep its regular telephone numbers. This way, the IP PBX switches local calls over the data network inside the enterprise and allows all users to share the same external phone lines.

How it works:

SIP Phone

An IP PBX or IP Telephone System consists of one or more SIP phones, an IP PBX server and optionally a VOIP Gateway to connect to existing PSTN lines. The IP PBX server functions in a similar manner to a proxy server: SIP clients, being either soft phones or hardware-based phones, register with the IP PBX server, and when they wish to make a call they ask the IP PBX to establish the connection. The IP PBX 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. More information and commonly asked questioned about IP PBXs can be found on IP PBX, SIP & VOIP FAQ

 

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

An IP PBX runs as software on a computer and can leverage the advanced processing power of the computer and user interface as well as Windows’ features. Anyone proficient in networking and computers can install and maintain an IP PBX. By contrast a proprietary phone system often requires an installer trained on that particular proprietary system!

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

An IP PBX 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 the phone technicians.

Benefit #3: Significant cost savings using VOIP providers:

With an IP PBX you can easily use a VOIP service provider 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 Telephone system allows you to connect hardware phones directly to a standard computer network port (which it can share with the adjacent computer). Software phones can be installed directly onto 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 extra ports to be used by the office phone system!

Benefit #5: Eliminate vendor lock in!

IP PBXs are based on the open SIP standard. You can now mix and match any SIP hardware or software phone with any SIP-based IP PBX, PSTN Gateway or VOIP provider. 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 PBX: 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 PBX you can deliver better customer service and better productivity: Since the IP telephone 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 improving customer service and cutting cost 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 PABX is software-based, it is easier for developers to add and improve feature sets. Most VOIP phone systems come with a rich feature set, including auto attendant, voice mail, ring groups, advanced reporting and more. 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 PBX 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 are 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: SIP phones are easier to use

Employees often struggle using advanced phone features: Setting up a conference, transferring a call – On an old PBX it all requires instruction.

Not so with an IP PBX – all features are easily performed from a user friendly Windows GUI. In addition, users get a better overview of the status of other extensions and of inbound lines and call queues via the IP PBX Windows client. Proprietary systems often require expensive ‘system’ phones to get an idea what is going on on your phone system. Even then, status information is cryptic at best.

Conclusion

Investing in a software-based IP PBX 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 PBX delivers such significant savings in management, maintenance, and ongoing call costs, that upgrading to an IP PBX, should be the obvious choice for any company.

 

 

 

 

What is a PBX Phone System?

PBX stands for Private Branch Exchange, which is a private telephone network used within a company. The users of the PBX phone system share a number of outside lines for making external phone calls. A PBX connects the internal telephones within a business and also connects them to the public switched telephone network (PSTN) One of the latest tendencies in PBX phone system development is the VoIP PBX, also known as IP PBX, which uses the Internet Protocol to transmit calls. Nowadays, there are two different PBX phone system options:

  • Traditional PBX
  • IP PBX

An IP PBX is a software-based PBX phone system solution which helps accomplish certain tasks and delivers services that can be difficult and costly to implement when using a traditional proprietary PBX.

What is a STUN Server?

A STUN (Simple Traversal of User Datagram Protocol [UDP] Through Network Address Translators [NATs]) server allows NAT clients (i.e. computers 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 and so 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.

What is Unified Communications?

Unified Communications is defined as the process in which all means of communication, communication devices and media are integrated, allowing users to be in touch with anyone, wherever they are, and in real time. The objective of Unified Communications is to optimize business procedures and boost human communications by simplifying processes. Until very recently, voice and data communications in the business environment were not unified. Telephones were simple black boxes sitting on the users’ desks, connected via their own network and completely isolated from the company’s computer data network. With the evolution of telephony via IP, the integration of both communication worlds has become possible, allowing for companies to transform their business processes. This development of breaking down communication barriers and allowing people using different forms of communication, different devices, and different media to communicate to anyone, anywhere and at any time is known as Unified Communications. The system from VoIP Business Solutions is an IP PBX that delivers unified communications technology by merging voice and data networks, allowing businesses to simplify real time delivery of information and ensure ease of use. Collaboration between co-workers is made easier and cooperation with business partners and clients becomes more direct; even when the parties are not located in the same place.

Key Unified Communications features:

  • Unified messaging – receive voice mail in your e-mail Inbox
  • Presence management (BLF) – eliminates expensive telephone tag
  • Integrated fax server
  • Connect branch offices
  • TAPI – initiate calls from Microsoft Outlook or CRM application
  • VOIP Client – can be used as a softphone and in/outside the office
Initiate phone calls from Microsoft Outlook by simply right-clicking on an Outlook contact

Initiate phone calls from Microsoft Outlook by simply right-clicking on an Outlook contact

The add-on for MS Outlook delivers a full set of enterprise level features that make it easy to install, configure, manage and use; allowing employees to increase mobility and productivity. All these benefits come at a very competitive price.

What is a voice mail system?

A voice mail system is a centralized system used in businesses for sending, storing and retrieving audio messages, just like an answering machine would do at home. 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 gives other available options, such as paging the user or being transferred to another extension or a receptionist. A voicemail system in a business is essential to keep external and internal communications flowing seamlessly and efficiently. VoIP Business Solutions has integrated an advanced voicemail system in its IP PBX. The VBS Phone System delivers a complete voice mail solution that incorporates Unified Communications by allowing voice mail to be forwarded to the user’s email inbox.

What is Voice over IP?

Voice over IP is the same as Voice over Internet Protocol, and is better known as VoIP. Voice over IP refers to the diffusion of voice traffic over internet-based networks. Internet Protocol (IP) was originally designed for data networking and following its success, the protocol has been adapted to voice networking. Voice over IP (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. 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 voice over IP, are making businesses adopt VoIP Phone Systems at a staggering pace. Voice over IP (also called VoIP, IP Telephony, and Internet telephony) refers to technology that enables routing of voice conversations over the Internet or a computer network. 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 / anyone: Both to VOIP numbers as well as people with normal phone numbers.

 

 

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 – the 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 analog) transmission of voice and data over ordinary telephone copper wires, resulting in better quality and higher speeds, than available with analog 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 (plurals Requests for Comments but 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 a virtual phone number?

A virtual phone number a feature of VoIP that allows you to attach additional phone numbers with different area codes to your basic VoIP service. This feature allows people to phone you without incurring long-distance charges from the same or adjacent non-toll area codes. All outgoing calls, however, are billed as if coming from your main phone number. Virtual phone numbers typically each cost a few extra dollars per month.

What is a converged network?

A converged network is a single network capable of carrying a mixture of voice (telephone), video (production and training), and application data.

 

How do IP PBX / VOIP phone systems work?

VOIP Phone System / IP PBX system consists of one or more SIP phones / VOIP phones, an IP PBX server and optionally includes a VOIP Gateway. The IP PBX server is similar to a proxy server: SIP clients, being either soft phones or hardware based phones, register with the IP PBX server, and when they wish to make a call they ask the IP PBX to establish the connection. The IP PBX 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.

How an IP PBX integrates on the network and how it uses the PSTN or Internet to connect calls

How an IP PBX integrates on the network and how it uses the PSTN or Internet to connect calls

What are the benefits of an IP PBX?

  • Much easier to install & configure than a proprietary phone system

A software program running on a computer can take advantage of the advanced processing power of the computer and user interface & features of Windows. Anyone with an understanding of computers and Windows can install and configure the PBX. A proprietary phone system often requires an installer trained on that particular proprietary system!

 

  • Easier to manage because of web based configuration interface:

A VOIP phone system has a web based configuration interface, allowing you to easily maintain and fine tune your phone system. Proprietary phone systems often have difficult to use interfaces which are often designed so that only the phone system installers can effectively use it.

 

  • Call cost reduction:

You can save substantially by using a VOIP service provider for long distance or international calls. Easily connect phone systems between offices/branches and make free phone calls.

 

  • No need for separate phone wiring – use computer network:

A VOIP phone system allows you to connect hardware phones directly to a standard computer network port (which it can share with the adjacent computer). Software phones can be installed directly onto the PC. This means that you do not need to install & maintain a separate wiring network for the phone system, giving you much greater flexibility to add users/extensions. If you are moving into an office and have not yet installed phone wiring, you can save significantly by just installing a computer network.

 

  • No vendor lock in:

Use standard phones: VOIP phone systems are open standard – all modern IP phone systems use SIP as a protocol. This means that you can use almost any SIP VOIP phone or VOIP gateway hardware. In contrast, a proprietary phone system often requires proprietary phones to use advanced features, and proprietary extension modules to add features.

 

  • Scalable:

Proprietary systems are easy to outgrow: Adding more phone lines or extensions often requires expensive hardware upgrades. In some cases you need an entirely new phone system. Not so with a VOIP phone system: a standard computer can easily handle a large number of phone lines and extensions – just add more phones to your network to expand!

  • Better customer service & productivity:

Since calls are computer based, it’s much easier for developers to integrate with business applications. For example: an incoming call can automatically bring up the customer record of the caller, dramatically improving customer service and cutting cost 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.

  • Software based Phones are easier to use:

It’s often difficult to use advanced phone system features such as conferencing on proprietary phones. Not so with software based SIP phones – all features are easily performed from a user friendly windows GUI.

  • More features included as standard:

Since a VOIP phone system is software based, it’s more easy for the developers to develop, add and improve feature sets. Therefore most VOIP phone systems come with a rich feature set, including auto attendant, voice mail, call queueing and more. These options are often very expensive in proprietary systems.

  • Better control via better reporting:

VOIP settings store inbound and outbound call information in a database on your server, allowing for much more powerful reporting of call costs and call traffic.

  • Better overview of current system status and calls:

Proprietary systems often require expensive ‘system’ phones to get an idea what is going on on your phone system. Even then, status information is cryptic at best. With VOIP systems you can define which users can see phone system status graphically via a web browser.

  • Allow users to hot plug their phone anywhere in the office:

Users simply take their phone, plug it into the nearest Ethernet port and they keep their existing number!

  • Allows easy roaming of users:

Calls can be diverted anywhere in the world because of the SIP protocol characteristics.

What is H323?

H323 is a set of standards from the ITU-T, which defines a set of protocols to provide audio and visual communication over a computer network. H323 is a relatively old protocol and is currently being superceded by SIP – Session Initiation Protocol. One of the advantages of SIP is that it’s much less complex and resembles the HTTP / SMTP protocols. Therefore most VOIP equipment available today follows the SIP standard. Older VOIP equipment though would follow H 323.

What do the terms FXS and FXO mean?

FXS and FXO are the name of ports used by Analog phone lines (also known as POTS – Plain Old Telephone Service).

FXS – Foreign eXchange Subscriber interface is the port that actually delivers the analog line to the subscriber. In other words it is the ‘plug on the wall’ that delivers a dialtone, battery current and ring voltage. FXO – Foreign eXchange Office interface is the port that receives the analog line. It is the plug on the phone or fax machine, or the plug(s) on your analog phone system. It delivers an on-hook/off-hook indication (loop closure). Since the FXO port is attached to a device, such as a fax or phone, the device is often called the ‘FXO device’. FXO and FXS are always paired. Without a PBX, a phone is connected directly to the FXS port provided by a telephone company.

FXS / FXO without a PBX

FXS / FXO without a PBX

If you have a PBX, then you connect the lines provided by the telephone company to the PBX and then the phones to the PBX. Therefore, the PBX must have both FXO ports (to connect to the FXS ports provided by the telephone company) and FXS ports (to connect the phone or fax devices to).

FXS / FXO with a PBX

FXS / FXO with a PBX

FXS & FXO & VOIP

You will come across the terms FXS and FXO when deciding to buy equipment that allows you to connect analog phones to a VOIP Phone System or traditional PBXs to a VOIP service provider or to each other via the Internet.

An FXO gateway

To connect analog phone lines to an IP phone system you need an FXO gateway. This allows you to connect the FXS port to the FXO port of the gateway, which then translates the analog phone line to a VOIP call.

FXO Gateway

FXO Gateway

An FXS gateway

An FXS gateway is used to connect one or more lines of a traditional PBX to a VOIP phone system or provider. You need an FXS gateway because you want to connect the FXO ports (which normally are connected to the telephone company) to the Internet or a VOIP system.

FXS Gateway

FXS Gateway

An FXS adapter a.k.a. ATA adapter

An FXS adapter is used to connect an analog phone or fax machine to a VOIP phone system or to a VOIP provider. You need this because you need to connect the FXO port of the phone/fax machine to the adapter.

FXS Adapter

FXS Adapter

FXS/ FXO gateways are widely available.

Connecting

FXS/ FXO procedures – how it technically works

If you are interested to know in more technical detail how an FXS/ FXO port interoperate, here is the exact sequence:

When you wish to place a call:

  1. You pick up the phone (the FXO device). The FXS port detects that you have gone off hook.
  2. You dial the phone number, which is passed as Dual-Tone Multi-Frequency (DTMF) digits to the FXS port.

Inbound call:

  1. The FXS port receives a call, and then sends a ring voltage to the attached FXO device.
  2. The phone rings
  3. As soon as you pick up the phone you can answer the call.

Ending the call – normally the FXS port relies on either of the connected FXO devices to end the call.