PBXware is a scalable PBX solution featuring a range of traditional telephony and emerging VOIP technologies. The creation of national/global voice networks and a full range of PSTN and VOIP technologies are supported. PBXware features least cost routing, voicemail, ACD queues, IVR auto attendants, conferences, music on hold and much more - a real cost saving solution in a fully featured PBX package! Some of the advanced features included are self-install on any Linux distribution, auto updates, system backup, provider templates, web interface, call recordings, and real time call monitoring.

VoIP (Voice over Internet Protocol) is a category of hardware and software that enables people to use the Internet as the transmission medium for telephone calls by sending voice data in packets using IP rather than by traditional circuit transmissions of the PSTN. VoIP is usually much less expensive than PSTN.

VoIP can be deployed to use industry standard high encryption technologies (SSL and VPN).

YES. To use VoIP with your regular analog phone, you will need to install an ATA (Analog Telephone Adapter) or channel bank devices. These devices convert the analog signal to digital data in order to work with VoIP.

Yes, but only with UPS (Uninterruptible Power Supply) device installed in the system. A UPS can maintain operation of critical equipment for a few minutes or hours until utility power is restored.

Dial-up can be used for VoIP when necessary or if it is the only type of connection available. However, we recommend using broadband since certain VoIP codecs (e.g. G.711) require higher bandwidth in order to achieve excellent audio quality.

YES! It is possible to place or receive any type of calls (local, long distance, international, etc) to/from PSTN lines.

The quality achieved is usually excellent although the voice quality depends on bandwidth quality and its availability.

Excellent audio quality on LAN is a standard feature of PBXware.

• PBXware will make your business run faster and easier then ever before and it will save you money on your total communications spending.
• PBXWARE is reliable
• PBXware is EASY TO USE with multiple role based administration 
• Your bottom line should grow due to better communication with customers/clients and reduced telephones bills. 
• No more unavailable employees. PBXware can keep your employees in contact even while travelling or at home
• Never be worried again about how your front line employees talk and behave to your important clients/customers or business partners, because PBXware will help you to record or monitor the calls
• Scalable proven solution
• TurnKey Solution
• Easy to use
• Wide range of supported handsets
• Superior Support
• Flexible platforms and delivery

The following PBXware features will help businesses improve employee productivity by helping them reduce the amount of time they spend on tasks, reach each other more easily, and work together more efficiently:

• Conference calls will help employees to conduct meetings regardless of time, location, saving traveling time, expenses, inconveniences, etc.
• Call recordings will improve calls, cost, and staff management.
• Call forwarding will help employees be more mobile, be on different locations and still receive calls
• Call waiting will help employees to take multiple calls at the same time
  

The following features will help business maintain high quality customer service:

• DID (Direct Inward Dialing) will help clients to reach a line directly without going through an operator or dialing several numbers
• IVR (Interactive Voice Response) answers all incoming calls and prompt callers to dial an extension, other destinations or leave a voicemail message-all without the help of an operator. Therefore, clients get information quickly and easily.
  
• Music On Hold will entertain your client
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• PBXware allows business branch offices to communicate easily using VoIP features which rapidly reduces telephone costs and time and allows users to contact remote coworkers via extension numbers.
• PBXware easily creates conference bridges between employees on local or remote (overseas) levels, which directly saves significant amount of time, and possible travel costs.
• PBXware allows call divert to employee mobile phones that gives more flexibility, mobility and time for achieving better business results. This directly saves time and reduces expenses.
•  PBXware browser based system administrattion allows users to easily navigate through the configuration. Therefore, it saves significant amounts of time and expenses on system maintenance, technical support, training, etc.
• VoIP usage will dramatically lower telephone communications costs.

Yes. VoIP features will rapidly reduce your business telephone costs.

PBXware uses Linux as its OS platform.

Please refer to:
www.bicomsystems.com/products/

Yes. Emergency calls can be placed by direct dialing, or by using a prefix number for an outgoing phone and then dialing the emergency number.

NO. PBXware browser based system administrator allows users to easily navigate through the configuration. It takes only couple of mouse clicks to add/ change relevant features for your business.

for accessing: - web GUI - TCP 80, 443, 81 - ssh - TCP 2020 for SIP phones: - TCP 10001, 5060-5069 - UDP 5060-5069, 10000-20000 for IAX phones: - TCP 5038, 5037 - UDP 4569 for jabber: - TCP 5222 or 5223

SIP phones are the same as VoIP or soft phones and are used for VoIP (Voice over Interver Protocol) calls. There are two types of SIP phones: Hardphone (resembles common phone) Softphone (computer - software phone) PBXware can be used with virtually all SIP phones on the market.

VoIP stands for Voice over Internet Protocol and it refers to the diffusion of voice traffic over internet-based networks.
VoIP is widely used to cut the cost of traditional PSTN and some of its major benefits are:

• More than one phone call on same line.
• Advance features, such as call forwarding, caller ID or automatic redialing, are simple and usually free.
• Communications can be secured with encryption

PBX stands for Private Branch Exchange. PBXware is our implementation of PBX - taken to a next level.
PBXware users share a number of outside lines for making external phone calls. Local calls, between the users, are available as well. On top of that, PBXware offers many aditional features, such as:

• Ring groups
• Conferences
• Inveractive Voice Reponses
• Queues
• Voicemail boxes
• Realtime monitorring
• Call recordings
• and much more...

SIP stands for Session Initiation Protocol. It is an IP telephony signaling protocol used to establish, modify and terminate VOIP telephone calls.
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.

SDP stands for Session Description Protocol. It is a format for describing streaming media initialization parameters. Streaming media is content that is viewed or heard while it is being delivered.

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.

RTP stands for Real Time Transport Protocol. It defines a standard packet format for delivering audio and video over the internet.

RTCP stands for Real Time Transport Protocol. It works hand in hand with RTP.
RTP does the delivery of the actual data, where as RTCP is used to send control packets to participants in a call. The primary function is to provide feedback on the quality of service being provided by RTP.

A SIP URI is basically a user

SIP uses Methods and Requests to establish a call session.

SIP Requests:
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

SIP responses:
1xx = informational responses, such as 180, which means ringing
2xx = success responses
3xx = redirection responses
4xx = request failures
5xx = server errors
6xx = global failures

1xx = informational responses

• 100 Trying
• 180 Ringing
• 181 Call Is Being Forwarded
• 182 Queued
• 183 Session Progress

2xx = success responses

• 200 OK
• 202 accepted: Used for referrals

3xx = redirection responses

• 300 Multiple Choices
• 301 Moved Permanently
• 302 Moved Temporarily
• 305 Use Proxy
• 380 Alternative Service

4xx = request failures

• 400 Bad Request
• 401 Unauthorized: Used only by registrars. Proxys should use proxy authorization 407
• 402 Payment Required (Reserved for future use)
• 403 Forbidden
• 404 Not Found: User not found
• 405 Method Not Allowed
• 406 Not Acceptable
• 407 Proxy Authentication Required
• 408 Request Timeout: Couldn't find the user in time
• 410 Gone: The user existed once, but is not available here any more.
• 413 Request Entity Too Large
• 414 Request-URI Too Long
• 415 Unsupported Media Type
• 416 Unsupported URI Scheme
• 420 Bad Extension: Bad SIP Protocol Extension used, not understood by the server
• 421 Extension Required
• 423 Interval Too Brief
• 480 Temporarily Unavailable
• 481 Call/Transaction Does Not Exist
• 482 Loop Detected
• 483 Too Many Hops
• 484 Address Incomplete
• 485 Ambiguous
• 486 Busy Here
• 487 Request Terminated
• 488 Not Acceptable Here
• 491 Request Pending
• 493 Undecipherable: Could not decrypt S/MIME body part

5xx = server errors

• 500 Server Internal Error
• 501 Not Implemented: The SIP request method is not implemented here
• 502 Bad Gateway
• 503 Service Unavailable
• 504 Server Time-out
• 505 Version Not Supported: The server does not support this version of the SIP protocol
• 513 Message Too Large

6xx = global failures

• 600 Busy Everywhere
• 603 Decline
• 604 Does Not Exist Anywhere
• 606 Not Acceptable

A sip call session between 2 phones is established as follows:

• The calling phone sends out an invite
• The called phone sends an information response 100
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To deal with fax, set PBXware options like this:

• Connect phone/fax line to PBXware box
• Create a trunk for this line
• Create a new 'DID' and point it to fax destination

Once fax enters the DID, PBXware will accept its signal and receive the fax. The same will be converted to a pdf and emailed to administrator.

Codecs convert analog signal to a digital one. This is needed for voice transmission over a network.
PBXware works with following codecs:

• GSM - 13 Kbps (full rate)
• iLBC - 15Kbpssize
• ITU G.711 - 64 Kbps (ulaw|alaw)
• ITU G.722 - 48/56/64 Kbps
• ITU G.723.1 - 5.3/6.3 Kbps
• ITU G.726 - 16/24/32/40 Kbps
• ITU G.728 - 16 Kbps
• ITU G.729 - 8 Kbps
• Speex - 2.15 to 44.2 Kbps
• LPC10 - 2.5 Kbps
• DoD CELP - 4.8 Kbps

FOIP stands for Fax over IP. It refers to the process of sending and receiving faxes over a VOIP network.
Fax over IP works via T38 and requires a T38 capable VOIP gateway as well as a T38 capable fax machine, fax card or fax software.
PBXware includes compatible T38 fax service.

DID stands for Direct Inward Dialing (also called DDI in Europe).
It is a feature used with PBX systems, whereby the telephone company allocates a range of numbers associated with one or more phone lines.
The purpose of DID is to allow a company to assign a personal number to each employee, without requiring a separate phone line for each.

PBXware consists of one or more SIP/VoIP phones and optionally includes a VoIP Gateway.
Clients and software or hardware based phones register with the PBXware server and may then establish connections to make calls.
PBXware has a directory of all phones/users and their corresponding address and is able to connect an internal call or route an external call via either a VoIP gateway or a VoIP service provider.

VoIP system requires the use of VoIP phones. These come in several versions/types:

• VoIP Soft phones
• USB VOIP phones
• Hardware SIP Phone
• Analog phone via an ATA adapter
NOTE: ATA adapter allows an analog phone to be connected to a VoIP system
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FXS and FXO are the name of ports used by Analog phone lines.

FXS - Foreign eXchange Subscriber is a port that delivers the analog line to the subscriber.

FXO - Foreign eXchange Office is a port that receives the analog line. Since the FXO port is attached to a device, such as a fax or phone, the device is often called the

VoIP gateway is a device which converts telephony traffic into IP for transmission over a network. Usage:

• Convert incoming PSTN/telephone lines to VoIP
• Connects a traditional PBX/Phone system to the IP network:

STUN stands for Simple Traversal of User Datagram Protocol Through Network Address Translators.
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 communication between the client and the VoIP provider and to establish a call.

A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network. A SIP server is also referred to as a SIP Proxy or a Registrar. An example of a SIP server is our PBXware

ENUM stands for Telephone Number Mapping.
The idea behind ENUM is 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. Different routes can be defined for different types of calls.

Licensing is based prinicpally on extentions with a variety of features included that correspond to a typical usage for that number of extentions.

Duration of call recording is unlimited.

Yes. PBXware includes enhanced servicea called

Yes, by connecting a special device to your PBXware and inserting a suitable mobile phone SIM card, PBXware can place and receive calls to your mobile phone network. Depending upon the mobile phone tax, these calls can be free of charge!

Yes. Standard plain old telephone service (POTS) and ISDN are all supported.

Yes, PBXware can connect with your existing analog or digital telephones. Depending on the number of existing extensions, PBXware can use channel banks or ATAs to offer the network all PBXware features and services.

YES. PBXware hardware and software are being continuously upgraded in accordance with latest software and technical achievements. Therefore, PBXware can replace any PBX system and might even provide you with more simplicity in use, features, effectiveness and reliability.

PBXware can provide an unlimited number of extensions using appropriate hardware.

Yes. We offer customers the possibility to upgrade the number of extensions and purchase other optional features. The functionality and effectiveness will remain same.

YES. PBXware can connect all your offices (local and remote) together into a network. Using VoIP, your company will have a cost savings in making phone calls versus traditional long distance phone calls over the PSTN.

YES. Your VoIP data will be encrypted between each office over the Internet

YES. Conference bridges are as easy to create as any other feature that PBXware supports.

YES. BICOM SYSTEMS provides all necessary equipment including IP or analog telephones and other equipment necessary to setup the network to your requirements.

No. PBXware will be previously configured according to your business needs and technical requirements.

YES. We can easily configure and keep all your existing extensions within PBXware.

BICOM SYSTEMS accepts Credit Cards (Visa, Mastercard, American Express , Debit Cards (Switch), Cash, Bank Money Transfers, PayPal, Moneybookers. If you don

No you are not obligated to do this. You may simply buy extra extentions as you need them.

No, you are not obligated to do this. You may simply buy the extra features as you need them.

Simply the price difference between the two packages.

Updates and Minor Updates are included within the 12 month support contract that is part of the sale price. Renewing your support contract will maintain this.

Minor Upgrades will be labeled by the numerals following the first decimal version number e.g. x.1, x.0.2, x.5 etc. These are included in your support contract that is part of the purchase price. Renewing your support contract will maintain this. Major Upgrades are offered as an Upsell and will be labeled by the numerals preceding the first decimal place in the Version Number e.g. 1.x, 2.x, 3.x etc.

PBXware is priced according to the number of extensions/lines and exact business needs. Please contact sales for further details.

The warranty period for PBXware hardware is one (1) year.

Yes. BICOM SYSTEMS informs customers and clients about all necessary upgrades, news, release notes, etc.

Yes. We offer standard, enhanced, and emergency support and maintenance contracts. Please contact sales for further details.