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    FXO and FXS are the ports used by analog phone lines and analog phones and faxes. These two interfaces are always paired (FXO is always connected to FXS and vice versa). In telecommunications, FXS and FXO are used to indicate whether VoIP equipment is designed to support analog phones (stations, FXS) and analog lines (office, FXO). FXS is the port used by analog lines, example: telephone jack on the wall / FXS ports of an analog telephone exchange used to connect analog devices. To connect analog devices to VoIP PBX, use FXS media gateway (media gateway with FXS port). FXO is the port used by analog devices, example: FXO port of an analog phone, modem, fax (these devices are often called FXO devices ), FXO port of an analog telephone exchange used to connect lines.

 About VoIP VoIP challenges: Available bandwidth Delay/Network Latency Packet loss Jitter Echo Security Reliability Pulse dialing to DTMF translation Many VoIP providers do not translate pulse dialing from older phones to DTMF. The VoIP user may use a VoIP Pulse to Tone Converter, if needed. Fixed delays cannot be controlled but some delays can be minimized by marking voice packets as being delay-sensitive (see, for example, Diffserv). The principal cause of packet loss is congestion, which can be controlled by congestion management and avoidance. Carrier VoIP networks avoid congestion by means of teletraffic engineering. Variation in delay is called jitter. The effects of jitter can be mitigated by storing voice packets in a buffer (called a play-out buffer) upon arrival, before playing them out. This avoids a condition known as buffer underrun, in which the playout process runs out of voice data to play because the next voice packet has not yet arrived, but increases delay by th...

 About VoIP Implementation Because UDP does not provide a mechanism to ensure that data packets are delivered in sequential order, or provide Quality of Service guarantees, business VoIP implementations face problems dealing with latency and jitter. This is especially true when satellite circuits are involved, due to long round trip propagation delay (400 milliseconds to 600 milliseconds for geostationary satellite). The receiving node must restructure IP packets that may be out of order, delayed or missing, while ensuring that the audio stream maintains a proper time consistency. This functionality is usually accomplished by means of a jitter buffer. Another challenge is routing VoIP traffic through firewalls and address translators. Private Session Border Controllers are used along with firewalls to enable VoIP calls to and from a protected enterprise network. Skype uses a proprietary protocol to route calls through other Skype peers on the network, allowing it to traverse symme...

Never in the history of telecommunications has a system so suited to the needs of business been available, at any price. Asterisk is an enabling technology, and, as with Linux, it will become increasingly rare to find an enterprise that is not running some version of Asterisk, in some capacity, somewhere in the network, solving a problem as only Asterisk can. Asterisk: The Professional's PBX This acceptance is likely to happen much faster than it did with Linux, though, for several reasons:   Linux has already blazed the trail that led to open source acceptance, so Asterisk can follow that lead.   The telecom industry is crippled, with no leadership being provided by the giant industry players. Asterisk has a compelling, realistic, and exciting vision.   End users are fed up with incompatible, limited functionality, and horrible support. Asterisk solves the first two problems; the community has shown a passion for the latter.

 About VoIP Functionality VoIP can facilitate tasks that may be more difficult to achieve using traditional networks: Ability to transmit more than one telephone call down the same broadband-connected telephone line. This can make VoIP a simple way to add an extra telephone line to a home or office.( TE110P Asterisk card is an ISDN PRI E1 card.) Incoming phone calls can be automatically routed to your VoIP phone, regardless of where you are connected to the network. Take your VoIP phone with you on a trip, and wherever you connect to the Internet, you can receive incoming calls. Many VoIP packages include PSTN features that most telcos (telecommunication companies) normally charge extra for, or may be unavailable from your local telco,such as 3-way calling, call forwarding, automatic redial, and caller ID. VoIP can be secure by using existing off the shelf protocols as Secure Real-time Transport Protocol. Most of the difficulties of creating a secure phone over traditional phone li...

  Voice over Internet Protocol, also called VoIP, IP Telephony , Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband is the routing of voice conversations over the Internet or through any other IP-based network. Companies providing VoIP FXS FXO service are commonly referred to as providers, and protocols which are used to carry voice signals over the IP network are commonly referred to as Voice over IP or VoIP protocols. They may be viewed as commercial realizations of the experimental Network Voice Protocol (1973) invented for the ARPANET providers. Some cost savings are due to utilizing a single network - see attached image - to carry voice and data, especially where users have existing underutilized network capacity that can carry VoIP at no additional cost. VoIP to VoIP phone calls are sometimes free, while VoIP to public switched telephone networks, PSTN, may have a cost that's borne by the VoIP user. There are two types of PSTN to VoIP services:...

  What is Call Queues? The   call   queue   is one of the most advanced features of Asterisk and yet is still fairly simple to implement, thanks to the FreePBX interface. Previously available only in high?end phone systems, the call queue is a standard feature of Asterisk PBX chinaroby . A call queue is different from a ring group in that the caller is not sent immediately to all the available agents. When a caller is sent to a call queue, they are sent to a virtual holding area to wait for the next available agent. During the wait, they can be listening to music on-hold and be told their position in the queue and the estimated hold time. Call queues are extremely valuable in sales and support organizations where inbound call volume can sometimes exceed the number of available agents. This provides a level of additional call capacity to the company while assuring the caller that they will be taken care of in the order that they called in, rather than having to contin...