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  Asterisk card VoIP Protocols The mechanism for carrying a VoIP connection generally involves a series of signaling transactions between the endpoints (and gateways in between), culminating in two persistent media streams (one for each direction) that carry the actual conversation. There are several protocols in existence to handle this. In this section, we will discuss some of those that are important to VoIP in general and to Asterisk specifically. Asterisk FXS FXO Card : TDM410P FXS card  8.2.1. IAX (The "Inter-Asterisk eXchange" Protocol) The test of your Asterisk-ness comes when you have to pronounce the name of this protocol. Newbies say "eye-ay-ex"; those in the know say "eeks." IAX is an open protocol, meaning that anyone can download and develop for it, but it is not yet a standard of any kind. [*] Officially, the current version is IAX2, but all support for IAX1 has been dropped, so whether you say "IAX" or "IAX2,...

  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...

  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. ...

  www.chinaroby.com   About VoIP Technical details The two major competing standards for VoIP are the IETF standard SIP and the ITU standard H.323. Initially H.323 was the most popular protocol, though in the "local loop" it has since been surpassed by SIP. This was primarily due to the latter's better traversal of NAT and firewalls, although recent changes introduced for H.323 have removed this advantage. However, in backbone voice networks where everything is under the control of the network operator or telco, H.323 is the protocol of choice. Many of the largest carriers use H.323 in their core backbones, and the vast majority of callers have little or no idea that their POTS calls are being carried over VoIP. Where VoIP travels through multiple providers' softswitches the concepts of Full Media Proxy and Signalling Proxy are important. In H.323, the data is made up of 3 streams of data: 1) H.225.0 Call Signaling; 2) H.245; 3) Media. So if you are in London, your pr...

 About VoIP Legal issues in different countries As the popularity of VoIP grows, and PSTN users switch to VoIP in increasing numbers, governments are becoming more interested in regulating VoIP VoIP in a manner similar to legacy PSTN services, especially with the encouragement of the state-mandated telephone monopolies/oligopolies in a given country, who see this as a way to stifle the new competition. In the European Union, the treatment of VoIP service providers is a decision for each Member State's national telecoms regulator, which must use competition law to define relevant national markets and then determine whether any service provider on those national markets has "significant market power" (and so should be subject to certain obligations). A general distinction is usually made between VoIP services that function over managed networks (via broadband connections) and VoIP services that function over unmanaged networks (essentially, the Internet). VoIP services tha...

  About VoIP Mass-market telephony ( https://www.chinaroby.com ) A major development starting in 2004 has been the introduction of mass-market VoIP services over broadband Internet access services, in which subscribers make and receive calls as they would over the PSTN. Full phone service VoIP phone companies provide inbound and outbound calling with Direct Inbound Dialing. Many offer unlimited calling to the U.S., and some to Canada or selected countries in Europe or Asia as well, for a flat monthly fee. These services take a wide variety of forms which can be more or less similar to traditional POTS. At one extreme, an analog telephone adapter (ATA) may be connected to the broadband Internet connection and an existing telephone jack in order to provide service n...

 About VoIP Pre-Paid Phone Cards VoIP has become an important technology for phone services to travelers, migrant workers and ex-pats, who either, due to not having a fixed or mobile phone or high overseas roaming charges, choose instead to use placa VoIP services to make their phone calls. Pre-paid phone cards can be used either from a normal phone or from Internet cafes that have phone services. Developing countries and areas with high tourist or immigrant communities generally have a higher uptake. VoIM Voice over Instant Messenger, like popular Skype, Voice over MSN, Yahoo, QQ in China and Google Talk. VoIM is one kind of general VoIP that was based on an IM. VoIP , specifically, usually is referred as traditional SIP or H.323 IP phone, as opposed to VoIM as newly emerged Skype-like services/phones.

  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 Integration into global telephone number system While the traditional Plain Old Telephone System (POTS) and mobile phone networks share a common global standard (E.164) which allocates and identifies any specific telephone line, there is no widely adopted similar standard for VoIP networks. Some allocate an E.164 number which can be used for VoIP as well as incoming/external calls. However, there are often different, incompatible schemes when calling between VoIP providers which use provider specific short codes.

  About VoIP Emergency calls The nature of IP makes it difficult to locate network users geographically. Emergency calls, therefore, cannot easily be routed to a nearby call center, and are impossible on some VoIP systems . Sometimes, VoIP systems may route emergency calls to a non-emergency phone line at the intended department. In the US, at least one major police department has strongly objected to this practice as potentially endangering the public. Moreover, in the event that the caller is unable to give an address, emergency services may be unable to locate them in any other way. Following the lead of mobile phone operators, several VoIP carriers are already implementing a technical work-around.For instance, one large VoIP carrier requires the re...

  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...

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...