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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 the le

 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 symmetric

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 lines,

  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 continually call back. Fo

  VoIP: Bridging the Gap Between Traditional Telephony and Network Telephony While VoIP is often thought of as little more than a method of obtaining free long-distance calling, the real value (andlet's be honestchallenge as well) of VoIP is that it allows voice to become nothing more than another application in the data network. It sometimes seems that we've forgotten that the purpose of the telephone is to allow people to communicate.  It is a simple goal, really, and it should be possible for us to make it happen in far more flexible and creative ways than are currently available to us.  Since the industry has demonstrated an unwillingness to pursue this goal, a large community of passionate people have taken on the task. The challenge comes from the fact that an industry that has changed very little in the last century shows little interest in starting now.   TE110P is an ISDN PRI E1 T1 card,users can build an Asterisk PBX voip phone system based on the card.

  What Are Some disadvantages of VoIP? If you're considering replacing your traditional telephone service with Internet Voice, there are some possible differences: Some Internet Voice services don't work during power outages and the service provider may not offer backup power; It may be difficult for some Internet Voice services to seamlessly connect with the 911 dispatch center or identify the location of Internet Voice 911 callers.See also IP PBX Phone System ,Asterisk PABX,Gateway phone appliance, VoIP Phone Systems for Small Business. Asterisk is an open source IP PABX that runs on the Linux operating system. It is an extremely powerful product capable of the most advanced PABX functions including voice-mail, conference calls, trunking, hunt groups, and much more.  

  What Are Some Advantages of VoIP? Because Internet Voice is digital, it may offer features and services that are not available with a traditional phone. If you have a broadband internet connection, you need not maintain and pay the additional cost for a line just to make telephone calls. With many Internet Voice plans you can talk for as long as you want with any person in the world (the requirement is that the other person has an Internet connection).  You can also talk with many people at the same time without any additional cost. IP PBX Asterisk , 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.

           ISDN BRI and ISDN PRI Services   There are two main services associated with ISDN Basic Rate Interface (BRI) and Primary Rate Interface (PRI). Both services consist of multiple channels over which data can be sent (known as B channels) and also include a signaling channel (the D channel). The D channel is used for control and signaling purposes, such as setting and tearing down ISDN call. Referred to as out-of-band signaling, this method ensures that other ISDN calls do not interfere with existing connections, that bandwidth on the B channels is reserved for data only, and ultimately results in quicker call setup and teardown. Basic Rate Interface (BRI) ISDN Basic Rate Interface ISDN is made up of two 64 Kbps B channels that are used for sending and receiving data in full duplex, and one 16K D channel for signaling. In total, an ISDN BRI interface provides 144K of bandwidth (64

  What is IVR ? IVR (Interactive Voice Response) The Digital Receptionist menu within the FreePBX interface provides for Interactive Voice Response (IVR) menus. A well designed IVR system is one of the key features that can give a company a very professional appearance. An example of an IVR menu would go like this: "Thank you for calling American Widgets, for sales press 1, for support press 2, for a company directory press the pound key, or you may dial an extension at any time" FreePBX interface allows us to easily build complex, multi-branching voice menus to help route callers to appropriate departments. For example, once a user presses "2" for support, another menu will ask the user to "press 1 for the Presidio model line, press 2 for the Pendleton model line". Using the IVR menu system, we can route any valid key sequence to another me

        Difference Between Asterisk and Issabel,VitalPBX The simplest way of looking at it is that Asterisk is simply one of the many components of Issabel,VitalPBX. While Asterisk is the actual PBX software, Issabel,VitalPBX is a self-installing package that installs a complete operating system, Asterisk PBX, and all the supporting components as listed in the previous section. The core strength of Issabel,VitalPBX lies in its simple setup and FreePBX (the web interface). This book will cover FreePBX in detail for the configuration of our PBX system as well as the other components. Voip ： voice over internet protocol To get an Asterisk phone system up and running, we would have to pick a supported Linux distribution, install the distribution, configure it securely, and then install Asterisk and configure that. With Issabel,VitalPBX we have one installation routine, which not only gives us a f

  The PRI (PRI : Primary Rate Interface) is a telecommunications interface standard used on an ISDN (ISDN: Integrated Services Digital Network) for carrying multiple voice and data transmissions between the network and end users. PRI is the standard for providing telecommunication services to businesses,offices,hotels,schools and home.There are 3 types of PRI.It is based on the E1 is common in Europe,Asia,South America and Australia,while T1 transmission in the US, Canada, and J1 in Japan. The T1 line consists of 23 bearer (B) channels and one data (D) channel for control purposes. The E1 carrier provides 30 B- and one D-channel for a bandwidth of 2.048 Mbit/s. ISDN PRI cards have 1/2/4 E1/T1 ports, with PCI/PCIE interface.They support Asterisk,FreePBX,Issabel,VitalPBX,Dahdi ...  Users can build a VoIP phone PBX ,Asterisk phone system,call center for enterprises,offices,hotels,schools,business,home...

  Issabel Server : Asterisk PBX Issabel is Powered By Asterisk. Issabel is an Open Source Software that brings together IP Communications Services in one place It's a platform that simplifies the management of your business interaction channels, incorporating a Telephone exchange (VoIP) with email, CRM, fax, videoconference, recording, reports and more. And it is free,download it from www.issabel.org All of ChinaRoby's Asterisk cards can wok with Issabel. With Issabel and ChinaRoby's Asterisk cards,users can build an Asterisk IP PBX ,VoIP Phone System,call centre,Issabel server for home / businesses / hotels / schools ...

  Freepbx Server : Asterisk PBX Freepbx is based on Asterisk which is an Open Source Software of phone system. Freepbx is an Open Source Software that brings together IP Communications Services in one place. It's a platform that simplifies the management of your business interaction channels, incorporating a Telephone exchange (VoIP) with email, CRM, fax, videoconference, recording, reports and more. And it is free,download it from www.Freepbx.org All of Asterisk cards (FXS FXO cards and ISDN PRI E1 cards) can wok with Freepbx,that is also an Open Source Software,so,users can edit the code to meet the need. And the features and function is powerful.  With Freepbx and ChinaRoby's Asterisk cards,users can build an Asterisk VoIP PBX ,VoIP Phone System,call centre,Freepbx server for home / businesses / hotels / schools ...

  What is Asterisk  IP PBX ? Asterisk is one of the world's leading open source softwares, telephone PBX platforms. With Asterisk cards ( FXS cards,FXO cards,PRI T1 cards,PRI E1 cards),Asterisk can convert a general purpose computer into a VoIP communications solution ,free phone PBX server,phone system for home / small business / school,hotel,office etc . Asterisk IP PBX , VoIP Phone system used by all kinds of home / Soho / companies / business / schools / hotels to improve their communication solution. Asterisk is the market leader in open source VoIP PBX (Asterisk PBX).There are many voip phone appliances softwares based on Asterisk to meet all kinds of requirement .such as FreePBX,Issabel ,VitalPBX,ViciBOX ... They are great communication servers in functionality, scalable and sophisticated than those available today. Asterisk software is free to download,it is a