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  Compiling Asterisk Once you've compiled and installed the zaptel and libpri packages (if you need them), you can move on to Asterisk . This section walks you through a standard installation and introduces some of the alternative make arguments that you may find useful. We'll also look at how you can edit the Makefile to optimize the compilation of Asterisk. 3.5.1. Standard Installation Asterisk is compiled with gcc through the use of the GNU make program. Unlike many other programs, there is no need to run a configuration script for Asterisk. To get started compiling Asterisk, simply run the following commands (replace version with your version of Asterisk): # cd /usr/src/asterisk- version # make clean # make # make install # make samples Be aware that compile times will vary between systems. On a current-generation processor, you shouldn't need to wait more than five minutes. At Astricon, someone reported successfully compiling ...

  The IAX protocol was developed by Digium for the purpose of communicating with other Asterisk servers (hence " the Inter-Asterisk eXchange protocol "). IAX is a transport protocol (much like SIP) that uses a single UDP port (4569) for both the channel signaling and Realtime Transport Protocol (RTP) streams. As discussed below, this makes it easier to firewall and more likely to work behind NAT. IAX also has the unique ability to trunk multiple sessions into one dataflow, which can be a tremendous bandwidth advantage when sending a lot of simultaneous channels to a remote box. Trunking allows multiple data streams to be represented with a single datagram header, to lower the overhead associated with individual channels. This helps to lower latency and reduce the processing power and bandwidth required, allowing the protocol to scale much more easily with a large number of active channels between endpoints. 8.2.1.2. Future Since IAX was optimized for voice, it has rece...

  www.chinaroby.com Configuring Inbound IAX Connections The Inter-Asterisk eXchange (IAX) protocol is usually used for server-to-server communication; more hard phones are available that talk SIP. However, there are several soft phones that support the IAX protocol, and work is progressing on several fronts for hard phone support in firmware. The primary difference between the IAX and SIP protocols is the way media (your voice) is passed between endpoints. With SIP, the RTP (media) traffic is passed using different ports than those used by the signaling methods. For example, Asterisk receives the signaling of SIP on port 5060 and the RTP (media) traffic on ports 10,000 through 20,000, by default. The IAX protocol differs in that both the signaling and media traffic are passed via a single port: 4569. An advantage to this approach is that the IAX protocol tends to be better suited to topologies involving NAT. An IAX user is used to authenticate and handle calls comi...

  Asterisk: The Professional's PBX 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 - a Professional 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 Security The many consumer VoIP solutions do not support encryption yet, although having a secure phone is much easier to implement with VoIP than traditional phone lines. As a result, it is relatively easy to eavesdrop on VoIP calls and even change their content. There are several open source solutions that facilitate sniffing of VoIP conversations. A modicum of security is afforded due to patented audio codecs that are not easily available for open source applications, however such security through obscurity has not proven effective in the long run in other fields. Some vendors also use compression to make eavesdropping more difficult. However, real security requires encryption and cryptographic authentication which are not widely available at a co...

  About VoIP Single point of calling With hardware VoIP solutions it is possible to connect the VoIP router into the existing central phone box in the house and have VoIP at every phone already connected. Software based VoIP services require the use of a computer, so they are limited to single point of calling, though telephone sets are now available, allowing them to be used without a PC. Some services provide the ability to connect WiFi SIP phones so that service can be extended throughout the premises, and off-site to any location with an open hotspot.However, note that many hotspots require browser-based authentication, which most SIP phones do not support

  About VoIP Difficulty with sending faxes The support of sending faxes over VoIP is still limited. The existing voice codecs are not designed for fax transmission. An effort is underway to remedy this by defining an alternate IP-based solution for delivering Fax-over-IP, namely the T.38 protocol. Another possible solution to overcome the drawback is to treat the fax system as a message switching system which does not need real time data transmission - such as sending a fax as an email attachment (see Fax) or remote printout (see Internet Printing Protocol). The end system can completely buffer the incoming fax data before displaying or printing the fax image.

  Some broadband connections may have less than desirable quality. Where IP packets are lost or delayed at any point in the network between VoIP users, there will be a momentary drop-out of voice. This is more noticeable in highly congested networks and/or where there is long distances and/or interworking between end points. Technology has improved the reliability and voice quality over time and will continue to improve VoIP performance as time goes on. It has been suggested to rely on the packetized nature of media in VoIP communications and transmit the stream of packets from the source phone to the destination phone simultaneously across different routes (multi-path routing). In such a way, the temporary failures have less impact on the communication quality. In capillary routing it has been suggested to use at the packet level Fountain codes or particularly raptor codes for transmitting extra redundant packets making the communication more reliable. A number of protocols have b...

 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...

  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 th...

  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. TE110PE 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. I...

  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...

  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 ...

  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...