What Is A Asterisk VOIP Phone System?

Asterisk Voip

Asterisk is an Internet Protocol PBX software that runs on a number of operating systems - Mac OS X, Open BSD, Linux, Sun Solaris and Free BSD. It supports Voice over Internet Protocol and using relatively cheaper hardware it can operate along with all standard telephony equipment. It also provides many advanced features that are usually related with expensive and high end PBXs. Asterisk is the main influencer of VoIP and the most popular software available with the Asterisk Community.

VoIP is getting recognized by the end users and small shops etc only now. Earlier it was used only by large enterprises and high end users. An agreement recently to standardize open protocols to enable delivery of calls and the combining of high speed bandwidth and inexpensive rates has brought VoIP technology closer to the clasp of Linux/*BSD administrators.

Asterisk Voip

The Session Initiation Protocol (SIP) will be the main protocol that will lead the industry for the deployment of VoIP. For two systems to describe the voice traffic that needs to reach from one point to another, SIP uses interactions that are text based and are similar to HTTP and SMTP. The description that allows a call to qualify between two points includes caller ID information, authentication, voice traffic parameters and other information.

H323 and MGCP are other VoIP protocols supported by Asterisk however SIP has more number of software stacks and phones that use it as a means for call description and for a new user SIP is easier to debug because of its use of headers with simple text.

The SIP channel starts out by managing the call flows. It first checks which channel the call came from and where the call should be sent to in extensions.conf. The calls from two SIP phones are checked and the dialed numbers are matched. The dialed number will be converted into a variable which will be referred to when talking about the number being dialed. Clusters of matching statements of dialed numbers called contexts are defined by the extensions.conf file. Just like a subroutine is used contexts are also used in a similar manner. A match tests are performed against the number that is being processed with the help of the match statements. Then until a match is found the call is passed through a list of comparison.

There are special extensions in a context that are meant for special behavior. One of the most frequently used extensions is "h" which means hang up. To decide what applications should be triggered by a call and to see how the call should be routed there are as number of extension matches in a context.

The applications begin to be executed in the order of preference in which they are listed after a match is found. A number of applications are available and more applications can be integrated easily. One such application is AGI that links to external programs.

To set the SIP Gateway to be the IP address of the Asterisk server the SIP clients will have to be configured. The configuration is only a small part to get VoIP to work but once the process is understood it becomes an easy task to add more phones. Support in registering with the SIP server by the client is a must.

Asterisk has been able to come at the forefront for the development of VoIP because of its many skilled developers who contribute many lines of code and modern features. Asterisk Community has many contributors and ambassadors all around the world and the most eminent of all of them are the Asterisk Developers who has enabled the development of VoIP.