Asterisk PBX Voip

Asterisk supports many different communications protocols from both the modern world of VoIP and from the legacy PSTN. This makes it a powerful tool for building gateways and protocol converters.

Below is a recipe for building a VoIP-to-PSTN gateway using Asterisk, an analog or digital telephony interface card and a standard PC server. The steps are as follows:

1. Select your telephony interface hardware.

Asterisk applications that connect with legacy telephony systems (PBXs or the PSTN) require telephony interface hardware. Small system generally use analog or ISDN BRI connections. Larger systems (more than 12 lines) frequently use T1, E1 or J1 digital connections.

2. Select your computer hardware.

Asterisk can run on virtually any modern computer, but when building a production telephony application server you should follow a few basic best-practice guidelines.

3. Install Asterisk

Once you have your Asterisk hardware the next step is software. You will either need to install Linux or use a ready-to-run distribution to install Linux, Asterisk and various related software packages. Since these application tutorials are intended to help you create custom telephony applications we will start with a generic installation of CentOS 5.3 and then install Asterisk from the Yum repository. This make it relatively easy to keep Asterisk up to date and avoids the complexities of hand compiling the Asterisk source code.

4. Configure your connections

Now that Asterisk is installed and running you need to edit the system configuration files to implement connections to VoIP and PSTN services. Since this step is common to all applications (Asterisk doesn't do much good if it is not connected to anything) it contains information on creating both service connections (connections to VoIP or PSTN services) and endpoint connections (connections to phones or terminal adapters). Some applications require both service and endpoint connections (PBX, ACD) while others may require only service connections.

Asterisk Call Center

Call Center
Install Asterisk and start building custom telephony applications with AsteriskNOW. AsteriskNOW installs a complete Linux distribution, Asterisk, the DAHDi driver framework, an administrative GUI, the MySQL database, the Apache web server and a wide variety of development tools and components.
What Is AsteriskNOW?

AsteriskNOW makes it easy to create custom telephony solutions by automatically installing the "plumbing". Much of the complexity of Asterisk and Linux is handled by the installer and the administrative GUI. Application developers and intergrators can concentrate on building their solution.
What Can I Build With AsteriskNOW?

Among the many applications you can create with AsteriskNOW are:

* VoIP Gateway
* Skype Gateway
* IP PBX
* Call Center ACD
* Conference Bridge

* IVR Server
* Voicemail System
* Call Recorder
* Fax Server
* Speech Server

Who Is It For?

AsteriskNOW was built for application developers, systems integrators, students, hackers and others who want to create custom solutions with Asterisk.
Who Supports AsteriskNOW?

AsteriskNOW users have two options. There is an active community of AsteriskNOW users, integrators and developers who provide community support on the AsteriskNOW forums and mailing list.

Digium offers commercial support subscriptions for AsteriskNOW. If your application requires direct manufacturer support, check out the subscriptions page.
Features And Functions?

AsteriskNOW has a long list of features that make it the preferred Asterisk ready-to-run distribution:

* Quick and easy installation. Typically installs in 15 minutes or less.

* Requires no in-depth knowledge of Linux. Safe for Mac and Windows users.

* Web-based configuration interface makes most management tasks point-and-click.

* Build data-driven applications with integrated support for ODBC and HTTPS

* Wizard-driven configuration of VoIP connections makes it simple to connect.

* Install pre-built, packaged applications using the Application Manager.

* Automatic detection and configuration of Digium analog and digitial telephony hardware.

* Sound Manager makes it easy to create, install and manage system prompts and recordings

* Dialplan and AEL script editor with syntax highlighting and validation simplifies development.

* Log and Call Detail Record (CDR) viewer gives instant access to system activity.

* Integrated help text for applications, functions, CLI, AGI and AMI commands.

* Realtime monitoring and debugging console helps simplify the development process.

* Step-by-step application development tutorials teach the fundamentals quickly.

* Support for clustering and high availability make it easy to scale your applications.

* Automatic updates keep your system in tune and secure.

* Save money with a special offer on Digium support and training services.

Asterisk Digium Card

Digium® digital cards are high-performance, cost effective, digital telephony interfaces. With the combination of Digium hardware and Asterisk software, numerous telephony configurations are possible. From the traditional PBX to VoIP Gateways, Digium solutions are paving the way for a new generation of worldwide communications. All Digium E1/T1 interface cards have a built-in CSU/DSU and may be connected directly to the E1/T1 physical interface.

Combined with Asterisk, the Open Source PBX, these cards provide an interface to traditional telephony equipment, such as channel banks and provider T1 /E1 lines. Asterisk can extend the capabilities of the phone network, by acting as a bridge between traditional telephony systems and Voice over IP equipment.

Asterisk Telephony System

Asterisk is a powerful and flexible open source framework for building feature-rich telephony systems. As a Private Branch Exchange (PBX) which connects one or more telephones, and usually connects to one or more telephone lines, Asterisk offers very advanced features, including station-to-station calls, line trunking, call distribution, call detail rerecords, and call recording.

Asterisk can be used to provide Interactive Voice Response (IVR). The power and flexibility of a programmable phone system gives us the ability to respond to our customers in meaningful ways.

Asterisk has also a fully-functional voicemail system included. It supports voicemail contexts so that multiple organizations can be hosted from the same server. It supports different time zones so that users can track when their phone calls come in. It even provides the option to notify the recipient of new messages via email. In fact, we can even attach the message in audio!

Finally, Asterisk is a Voice Over IP (VoIP) system. The benefits of VoIP are numerous. For instance, we can have multiple users using the same Asterisk service from a variety of locations; we can have users in the local office using PSTN phones or IP phones; we can have remote VoIP users; we can even have entire Asterisk systems operated and run completely separately but with integrated routing; We can have an extension anywhere we have a reasonably fast Internet connection, which means employees can have an extension on the phone system at home if they have a broadband connection. The benefits are too many to list.

Flexibility often means complexity, and this is true in the case of Asterisk. This book is all you need to understand and use Asterisk to build the telephony system that meets your need. You will learn how to use the many features that Asterisk provides you with.

Asterisk PBX Elastix 2.0

Elastix 2.0 Alpha 3 has been released and now you can downloaded it for testing in its' 32 bits and 64 bits versions. Early next year our first beta version will be released. This new version will have better improvements such as better usability, a new image, software packages updated, and much more. Here we share with you some screen shots of what Elastix 2.0 will be like in the future.
Renovated Dashboard

The new Dashboard in Elastix 2.0 will allow users to be aware of their system, its performance, lost calls, mails, news from Elastix, and much more. The user will be able to choose the applets he wants to see and how he would see them. Everything will work by dragging and dropping the applets without the need of a browser plugin.

Elastix Operator Panel

Elastix Operator Panel is similar to Flash Operator Panel, but there is no need to install any plugin in your web browser. Another advantage is that you can modify the structure of your panel with out the need of touching any text file. Everything is drag and drop!

Elastix Conference Room

The Elastix Conference Room is a great tool to have virtual meetings. The Elastix Conference module can be extended with the Conference Room. This will give you nice features, like viewing presentations in several file formats, a document repository to share, and chat support.

Mail Improvements

Now it is easier to integrate Elastix with an External mail server. All you need is to fill in a form and you are set, no need to edit config files. Other useful improvement is the integration with the mailing list engine, Mailman. This will allow administrators to create mailing lists from the Elastix web interface.

Now you will be able to send faxes directly from the web interface. There will be no need for a fax client to be installed in the users desktops. You may send faxes by writing directly in the text area or you can upload pdfs, tiff, or plain text files.

DHCP Module

The new DHCP server module includes two important improvements. The first one is the screen that will allow administrators to see the IPs and MAC adresses of all their DHCP clients. The second one is that you can assign static ips to MAC addresses. This will allow for easier identification and configuration of the phones.

These are just some of the new functionalities that will came with Elastix 2.0.. Beta-testers and other enthusiastics can download this Alfa version (32 bits and 64 bits) to be tested. Bugs should be reported in the bug tracker.

Asterisk PBX Hardware

The Components

Your will have to set up three main components: the IP PBX itself, the phones (or softphones) to be used with it, and the gateway service that lets you call other people on the PSTN. I will describe how to set up each of these. I assume you already have a home network and broadband access. If you are behind a NAT firewall, it does not matter – you don’t have to do anything special like running the IP PBX outside the NAT. All that matters is that you have enough bandwidth (upstream and downstream) to carry voice traffic.
The IP PBX

You will need a computer to run the IP PBX. While you may already have a Linux server at home, I do not recommend using it to also run your phones. If you want to do that, you will have to figure out how to load and configure the PBX software on your own, although the rest of the instructions here will be helpful.

If you have an old PIII class machine lying around, then use that. If not, you can buy one on EBay for about $40.

I used a Dell OptiPlex GX1, a PIII 450MHz system with 128M RAM. It has a built-in sound and Ethernet, so no additional hardware is required.

You will load Asterisk@Home on this computer. It will take it over – it starts by formatting the hard disk, so make sure there is nothing on the machine that you want to keep. Detailed instructions are given below.
Phones

You can buy SIP phones from Grandstream, SNOM, Cisco, even Avaya. Or you can buy an adapter from SNOM or Cisco and use an analog phone. To get started, though, it is easiest to get a softphone and run it on another computer. We will describe how to obtain and install the XLite softphone on a Windows machine to use for a phone. You will also use this Windows machine to administer the IP PBX, through a browser. If you want to use a Linux machine instead, you will need to get an appropriate softphone for it, but you can still use its browser to administer the IP PBX.
SIP Gateway

I presume you want to communicate with others on the PSTN network, so you need to obtain gateway service. Since part of the call is being carried on the circuit switched network, it costs real money. This means you will have to pay for this part of the system.

I will describe how to set up access to two services, one for outgoing calls and another for incoming calls.

VoipJet is used for outgoing calls. It is priced at 1.3 cents per minute (USA), with no monthly minimum or service charges.

BroadVoice is used for incoming calls. You get a phone number (you can choose the area code and exchange). They have various plans, but I signed up for one that is $5.95 per month, with unlimited incoming minutes. You can also make outgoing calls using this account (it includes 100 minutes per month, and 3.9 cents per minute additional). There is a $9.95 one time activation fee to sign up for BroadVoice.

So if you have the hardware, you can set up your IP PBX for a total of $9.95 and run it for $5.95 per month.
Home Network

I assume you have a home network, and that you are setting up behind a Gateway Router (otherwise known as a NAT firewall). You will need to pick a static IP address for your IP PBX that is on your home network.