What is PBXware?

Incorporating a range of traditional telephony and modern VoIP technologies, PBXware is a scalable solution that enables enhanced management of business telecommunications. From routing and voicemail, to auto attendants and conferencing, PBXware can offer advanced features in a single package, saving you money on multiple systems.

What is VoIP?

Voice over Internet Protocol is a convenient, affordable means for businesses and individuals to use the internet to conduct telephone calls, and represents a more cost-effective alternative to traditional PSTN.

How secure is VoIP?

Security for VoIP uses industry standard encryption technology such as SSL and VPN.

Can I use VoIP with a regular (analog) telephone?

To connect VoIP phones to traditional telephony networks, you need to install an Analaog Telephone Adapter (ATA) which converts the analog signal into digital data.

Does PBXware work if the power fails?

In the event of power failure, PBXware will continue to operate should it have an Uninterruptible Power Supply device installed in the system. This has the ability to maintain operations for minutes/hours until power is restored.

Does VoIP work on dial-up?

A dial-up connection can support VoIP, but it is recommended to use broadband since certain codecs require higher bandwidths for quality purposes.

Can VoIP receive calls from PSTN?

VoIP is fully compatible with calls to/from PSTN lines.

How good is a VoIP sound quality?

Depending on bandwidth quality and availability, VoIP quality is excellent.

What about sound quality on LAN?

Sound quality on LAN is excellent and a standard feature of PBXware.

Why should I consider purchasing PBXware?

The benefits of PBXware are associated with cost, simplicity, efficiency and reliability. From saving money on telecommunications to keeping employees connected remotely, quality control to scalability, PBXware represents a new level of telephony efficiency.

How can PBXware help me improve my business results?

PBXWare offers a number of benefits to your business, aiding cost savings, productivity, and efficiency. Among those features available: – Conference calling – Call recording – Call forwarding – Call waiting – Direct Inward Dialling – Interactive Voice Response – Music On Hold – Destinations Permissions – Backup – Automatic updates

How can PBXware save time and money?

PBXware enables business locations to communicate through VoIP, reducing costs. In addition, remote workers can also access these same networks, eliminating travel costs and time, while the ability to divert to mobile phones enhances flexibility. There is also the benefit of working with a browser-based administration system, reducing expenses on system maintenance, technical support, training, etc.

Will my telephone bills reduce if I switch to VoIP?

In comparison to Public Switched Telephone Network (PSTN) services, switching to VoIP with significantly reduce your business telephone costs.

What is the OS platform for PBXware?

PBXware operates on a Linux OS platform.

Does PBXware support emergency call services?

Emergency calls can be placed by direct dialling, or with a prefix number for an outgoing phone followed by the emergency number. – Select your primary, secondary, and tertiary trunks for each destination – On placing a call to any of the configured destinations, PBXware will attempt to connect using first the primary trunk, then the secondary, and finally the third, depending on performance issues

Do employees need training to use PBXware?

The PBXware system administrator offers easy navigation through configuration, with only a few clicks of a mouse to activate the business features you need.

What are SIP Phones?

SIP phones are used for VoIP calls and are available in two types: hardphone (resembling a common telephone) and softphone (a computer software phone).

What is SIP?

Session Initiation Protocol (SIP) is a telephony signalling protocol used to establish, modify and terminate VoIP telephone calls.

What is SDP?

Session Description Protocol (SDP) is a format for describing streaming media content initialisation parameters.

What is echo cancellation?

Echo cancellation removes echo from a voice communication in order to improve voice call quality and reduce bandwidth consumption. Echo cancellation is required since speech compression techniques and packet processing delays generate two types of echo: acoustic and hybrid.

What is RTP?

Real Time Transport Protocol (RTP) defines a standard packet format for delivering audio and video data over the internet.

What is RTCP?

RTP Control Protocol (RTCP) works with RTP to send control packets to call participants. The primary function is to provide feedback on the quality of service provided by RTP.

What is a SIP URI?

Put simply, a SIP URI is a user’s SIP phone number, but resembles an email address in appearance. For example, the structure is: sip:[email protected]:Port (where x=username and y=host)

What are SIP Methods and Requests?

SIP Methods and Requests are the means with which a call session is established.

SIP Requests:

INVITE = Establishes session ACK = Confirms an INVITE BYE = Ends session CANCEL = Cancels establishing a session REGISTER = Communicates user location (host name, IP) OPTIONS = Communicates information about the capabilities of the calling and receiving SIP phones

26 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. Proxies should use proxy authorisation -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 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 two phones

A SIP call session between two phones is established as follows: -The calling phone sends out an invite. -The called phone sends an information response: 100 – Trying. -When the called phone starts ringing a response is sent: 180 – Ringing. -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: 200 – OK.

How does FAX work in VoIP environments?

To deal with fax, set PBXware options thus: -Connect phone/fax line to PBXware box -Create a trunk for this line -Create a new DID and point it to the fax destination Once the 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 for?

Codecs convert analog signals to digital. This is needed for voice transmission over a network. The following codecs are supported by PBXware: – GSM – 13Kbps (full rate) – iLBC – 15Kbps size – ITU G.711 – 64Kbps (ulaw|alaw) – ITU G.722 – 48/56/64Kbps – ITU G.723.1 – 5.3/6.3Kbps – ITU G.726 – 16/24/32/40Kbps – ITU G.728 – 16Kbps – ITU G.729 – 8Kbps – Speex – 2.15 to 44.2Kbps – LPC10 – 2.5Kbps – DoD CELP – 4.8Kbps

What is FoIP?

FoIP stands for Fax over Internet Protocol and refers to the process of transmitting faxes over a VoIP network. FoIP works via T38 (compatible with PBXware) and requires a T38 capable VoIP gateway as well as a T38 capable fax machine, fax card or fax software.

What is DID?

Direct Inward Dialling is a feature used with PBX systems and sees a telephone company allocate a range of numbers associated with one or more phone lines. DID enables businesses 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 an optional VoIP Gateway. Those with soft or hardware based phones register with the PBXware server to establish connections to make calls. A PBXware system features a directory of all users and is able to connect an internal call or route an external call via VoIP gateway or a VoIP service provider.

SIP/VoIP phone types

VoIP phones are available in a number of forms including VoIP softphones, USB phones, hardware SIP phones, and analog phones with an ATA adapter.

What do FXS and FXO mean?

FXS and FXO are the name of ports used by analog phone lines. Foreign eXchange Subscriber (FXS) is a port that delivers the analog line to the subscriber; and Foreign eXchange Office (FXO) is a port that receives the analog line. FXO and FXS are always paired, similar to a male/female plug.

What is VoIP gateway?

A VoIP gateway is the means of converting telephony traffic into IP for transmission over a network.

What is a STUN server?

Simple Traversal of User Datagram Protocol Through Network Address Translators (STUN). The STUN server enables a client to find out its public address, the NAT they are behind, and the internet side port associated by the NAT with a particular local port. This information is used to establish communications between the client and the VoIP provider.

What is a SIP server?

The main component of an IP PBX, the SIP server handles the setup of all calls in the network. PBXware is an example of a SIP server.

What does ENUM mean?

ENUM stands for Telephone Number Mapping, linking a phone number to an internet address published in the DNS system. The owner of an ENUM number can publish where a call should be routed to via DNS entry. Different routes can be defined for different types of calls.

Which ports are required for PBXware?

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