View source for GENERAL QUESTIONS FAQ

=What is PBXware?=
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.
=What is VoIP?=
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.
=How secure is VoIP?=
VoIP can be deployed to use industry standard high encryption technologies (SSL and VPN).
=Can I use VoIP with a regular (analog) telephone?=
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.
=Does PBXware work if the power fails?=
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.
=I only have dialup, can VoIP work?=
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.
=Can VoIP receive calls from PSTN?=
YES! It is possible to place or receive any type of calls (local, long distance, international, etc) to/from PSTN lines.
=How good is a VoIP sound quality?=
The quality achieved is usually excellent although the voice quality depends on bandwidth quality and its availability.
=What about sound quality on LAN?=
Excellent audio quality on LAN is a standard feature of PBXware.
=Why should I consider purchasing 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
=How can PBXware help me improve my business results?=
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’s while waiting to speak to someone
The following features will help business get most out of their networks:
*Destinations Permissions will control dialling of specific destinations, which will prevent unauthorized use of services and resources.
*Backup will automatically store your system settings, recordings, and other important data for easy retrival/restore.
*Auto Updates will update your system with latest bug fixes and enhancements of existing features automatically. 
=How can PBXware save my time and money?= 
*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.
=Will my telephone bills reduce if I use VoIP feature comparing to PSTN (Public Switched Telephone Network)?=
Yes. VoIP features will rapidly reduce your business telephone costs. 
=What is the OS platform for PBXware?= 
PBXware uses Linux as its OS platform.
=What features and functionalities are included with the PBXware?=
Please refer to:
www.bicomsystems.com/products/
=Does PBXware support Emergency call services?=
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.
=Do my employees need special education to use PBXware?=
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.
=What are SIP Phones?=
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.
=What is VoIP?=
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
=What is PBXware?=

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...
=What is SIP?=
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.
=What is SDP?=
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.
=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 RTP?=
RTP stands for Real Time Transport Protocol. It defines a standard packet format for delivering audio and video over the internet.
=What is RTCP?=
RTCP stands for Real Time Transport Control 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.

=What is a SIP URI?=
A SIP URI is basically 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)
Examples:

'''sip:joe.bloggs@212.123.1.213'''

'''sip:support@phonesystem.com'''

'''sip:22444032@phonesystem.com'''

=What are SIP Methods?=
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
 
=SIP responses=
'''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
 
=Example of SIP Call session between 2 phones=
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 – Trying – back.
*When the called phone starts ringing a response 180 – Ringing – is sent back
*When the caller picks up the phone, the called phone sends a response 200 – OK
*The calling phone responds with ACK – acknowledgement
*Now the actual conversation is transmitted as data via RTP
*When the person calling hangs up, a BYE request is sent to the calling phone
*The calling phone responds with a 200 – OK.
=How does FAX work in VOIP environments?=
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.
=What are Codecs Used For?=
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
=What is FOIP?=
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.
=What is DID?=
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.
=How does a PBXware system work?=
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.
=SIP/VoIP Phone Types=
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
=What do FXS and FXO mean?=
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 ‘FXO device’.

FXO and FXS are always paired, similar to a male/female plug.
=What is VoIP gateway?=
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:
=What is a STUN Server?=
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.
=What is a SIP server?=
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
=What does ENUM mean?=

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.
=Which Ports are Required for PBXware?=
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 and 5223

=How Does Call Rating Work in PBXware?=
From the call rating (billing) perspective, an Extension can be set as ''Slave'' or ''Master''. 
:*A Master Extension has its own credit and users are able to set the reminders, for example, when balance reaches certain amount and certain limitations.
:*Slave Extensions use funds that are available under a Master Extension. 

<span style="color: red;">'''NOTE:'''</span> When a Master Extension runs out of funds, none of the Extensions are able to make calls until the Tenant administrator adds more funds to the Master Extension. 

''PBXware does not have real-time billing.'' Instead, once call is finished, the system will charge for that call based on the rating duration.

{| class="wikitable" style="border: none; background: none;"
! colspan="1" rowspan="1" style="border: none; background: none;"|
! colspan="2"| Call Rating Info Section
|-
! Account Balance:
| Displays the available account balance.  
|-
! Available Funds:
| Displays available account funds (account balance + credit limit). 
|}

==Call Rating Procedure Explained==
[[image:DiagramFAQnew.png|right|450px]]
Available funds on the Master Extension will be divided by the number of Slave Extensions’ outgoing (or incoming) channels. Each Slave Extension gets the amount in advanced. The amount defines how long the call can last. 

:'''For example''': There is one Master Extension and two Slave Extensions. The account balance on the Master Extension is set to 100. The Slave Extension (101) has '''two''' outgoing channels and the Extension (102) has '''three''' outgoing channels. 

:This is what happens when making a call:
:*Extension (101) ''will have'' available funds in the amount of '''50''' per call
:*Extension (102) ''will have'' available funds in the amount of '''33,333333333''' per call.

''For visual representation of the example, please refer to the diagram.''