Asterisk Geekz » SK

How To Put Click to Call On Your Website-Step 1

SK — Wed, 09 Dec 2009 10:23:03 +0000

Upto this i assume you have sound knowledge of a linux system and have an Asterisk PBX running.If not, lets first check how this WIFI technology work from here

Create a new directory, under your main www directory, called "clickcall". Create a new file in there, called "clickcalltest.html". Copy the following code into that file. You will need to change "example.com" to your own domain.

Code:
 <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd"> <html> <head> <meta http-equiv="Content-Type" content="text/html; charset=ISO-8859-1"> </head> <script language="javascript"> // VoIP User Click To Call System 1.0 BETA // AJAX Main Functions // From http://www.voipuser.org/forum_topic_9696.html // May 2007 // // Set up our XMLHttpRequest Object var XMLHttpRequestObject = false; var currentStatus = 'offline'; var newStatus = 'offline'; if (window.XMLHttpRequest) { // Mozilla, Safari... XMLHttpRequestObject = new XMLHttpRequest(); } else if (window.ActiveXObject) { // Internet Explorer try { XMLHttpRequestObject = new ActiveXObject("Msxml2.XMLHTTP"); } catch (e) { try { XMLHttpRequestObject = new ActiveXObject("Microsoft.XMLHTTP"); } catch (e) {} } } if (!XMLHttpRequestObject) { // Remove this once you're done debugging and let failures go. Worst case, the visitor just won't see the button // Some users disable Java in their browsers. Not a great deal we can do about that. alert('Failed to create XMLHTTP instance.'); } // This function checks the presence flag on the webserver and outputs the button, or not function getData(dataSource, divID) { if (XMLHttpRequestObject) { var obj = document.getElementById(divID); XMLHttpRequestObject.open("GET", dataSource); XMLHttpRequestObject.onreadystatechange = function () { if ( (XMLHttpRequestObject.readyState == 4) && (XMLHttpRequestObject.status == 200) ) { if (XMLHttpRequestObject.responseText == 1) { newStatus = 'online'; } if (XMLHttpRequestObject.responseText == 0) { newStatus = 'offline'; } if (currentStatus != newStatus) { if (newStatus == "online") { obj.innerHTML = '<a href="http://www.example.com/clickcall/callme.php" target="_blank"> <input type="image" src="http://www.example.com/clickcall/click.jpg" alt="Click To Call" name="clickTocall"></a>'; } else { obj.innerHTML = ''; } currentStatus = newStatus; } } } } XMLHttpRequestObject.send(null); } var timerID = null function StartTimer() { getData('http://www.example.com/clickcall/getstatus.php', 'targetDiv'); timerID = self.setTimeout("StartTimer()", 500); } </script> <body onLoad="StartTimer()"> <div id="targetDiv"> </div> </body> </html>

How To Put Click to Call On Your Website-Introduction

SK — Wed, 09 Dec 2009 10:13:25 +0000

This question has cropped up a number of times in the forum and I felt it was time to address it more fully and present an open-source solution.We’ll also be using AJAX to control an active “presence” system for the on screen button.

Let’s have a look at some click-to-call background first and for the moment I’m going to assume that you’re some kind of ecommerce webmaster using Asterisk or TrixBox for your VoIP PBX. You may be an online store or site providing an online service. Whether product or service, you may be thinking that if you could get your visitor on the telephone, you’ll convert the sale or convince them they need your service. This is really the only consideration when analysing click to call as a commercial tool – will you generate more sales.

Here are some numbers* from a report which are worth considering:-
* a 22%-25% reduction in Website abandonment from pages with click-to-call services
* as much as a 100% increase in transaction conversions from click-to-call users compared with toll-free callers
* 88% of click-to-call users say they are more likely to contact a company that offers a click-to-call service than one that does not
I’m not talking here about telephone lead generation, which is something that works much like AdWords. As opposed to clicking through to a website, the user engages on the telephone. Google do have a program for this which they’re rolling out extremely slowly. It has taken two years to date. The Google program may interest you and be available in your country. You can check if it is here if you’re interested.

The real benefit of telephone lead generation is that it doesn’t require a website. But that’s out of scope here and not what this article is about.

This is also not what I call a “true” click to call system, this is a “click, give us your number and we’ll call you” type system. “Click to Call” has become a generic expression and the system described above probably should be called “Click to Speak”.

The click to callback type setup I’m describing here is similar to the commercial offerings from the likes of Estara. But I’m going to be talking about doing this in a completely open-source (free) environment, using a combination of Asterisk, AJAX and PHP.

If you want true click to call, using an embeded VoIP softclient within your page, you’ll need a widget.
So, you want to go ahead with this and put click to call buttons on your site and start seeing how it performs for you? There are a couple of immediate things to consider and we should look at some of the theories first.

Presence and Availability

Nothing would be more annoying to a site visitor than to see a click to call button, click it, and the phone just rings and rings. What’s critical therefore is some kind of presence management system.

There are three basic presence states with click to call – (1) you’re available, (2) you’re away, and (3) you’re there, but on the phone and already engaged. We can combine states (2) and (3) into one “away” state.

Visitors will sometimes land on a page and spend some time reading it – the “presence” indicator needs to be dynamically generated. Hence I’m using AJAX to allow a “real-time” presence update when a call is routed and answered (you’re “engaged”) or you’re “away” and unable to answer. We’ll do this in a very simple fashion to start with – if you’re away or engaged the button is going to disappear from the page, on the fly. This could be easily adapted to display a different button or alert the viewer in any way you choose that there’s been a change. For the usability reasons stated, it is important to do indicate presence somehow.

Security

You can’t just let any Tom, Dick or Harry onto your Asterisk box via the webserver. Nor can you allow any other webmaster put your button on someone elses site. So you need to ensure that the button will only display and operate on your pages, and you’ll ignore any calls to it from any other referrer.

The referrer field, part of the HTTP Header Specification, is not 100% reliable. Some browsers will not forward it to the server. In my experience, it’s about 95% reliable. You will need to assume 5% of calls or clicks on the click-to-call button are going to break. You can, however, detect that a header is invalid and send back a nice message.

You also need to restrict the call back number, assuming you do not want to pick up the cost of international calls to mobiles. Ian has done this within the Asterisk portion of code, allowing, in this example case, calls to the UK only. You’ll need to adapt this call plan as you see fit, and we can have some further discussion on doing that later in this thread.

The theory and code presented here should scale easily enough to a multiple person/call centre type arrangement. The theory and considerations are the important part. This warrants much further discussion and I invite you to that in this thread.
OK let’s get stuck in with some code. As this is likely to be adapated and improved, we’ll call this starting point the VoIP User Click To Call System 1.0 BETA.

If you’re not familiar with AJAX, you might want to read the starter guide here. There are plenty of additional AJAX resources on the web if you want to search around. We’re a VoIP forum and specific advice on AJAX should be sought elsewhere. But I will give a basic overview by way of comments within the code.

We need some way to store the 2 basic states of presence that we’ll be using. We’ll do that on the webserver, assuming, as should be the case, your webserver and Asterisk box are running on two different servers. Asterisk is going to talk to the webserver, to update the presence status flag, as and when calls are routed or something changes. We’ll drive the actual page the viewer sees with a 500ms Java timer, checking the presence status flag by calling a GET once every half a second. If you put something like this on a high traffic site, bear in mind the number of HTTP requests that will get generated by this timer and adjust the interval accordingly. Anything down to 2 or 3 seconds should be fine.

Now Lets dive into the topic more deeply and right some codes Step 1

Top VOIP Security Threats Continued

SK — Tue, 08 Dec 2009 11:30:12 +0000

So in the last post, you saw how DoS attacks and eavesdropping would be big concerns this year. Here are the more 3 points

Microsoft Office Communications Server: Hackers love attacking Microsoft, and Microsoft loves being unprepared. VIPER Lab predicts that hackers will find vulnerabilities in Microsoft Office Communications Server’s VoIP client and use it to access networks that had previously been secure, and the organization is not alone in reaching this conclusion. Network World blogger Mitchell Ashley suggests that Microsoft could learn from Vonage’s vulnerability to spoofing attacks.

I guess those of us who are using Windows are out of luck in this point. Then again, this is why business are leaning towards alternatives.

Vishing by VoIP: The FBI has been aware of vishing for nearly a year now, and the IC3 (Internet Crime Complain Center) recently released a report stating that vishing attacks are on the rise. With caller ID spoofing, the criminals can be very difficult to track, “due to rapidly evolving criminal methodologies,” according to the IC3.

Yup, first it was phishing, now it’s vishing.

VoIP Attacks Against Service Providers: These sorts of attacks will escalate, VIPER Lab predicts, because of readily available, anonymous $20 SIM cards. As UMA (Unlicensed Mobile Access) technology becomes more widely deployed to allow calls to switch from cell networks to VoIP networks, VIPER Labs warns that “service providers are, for the first time, allowing subscribers to have direct access to mobile core networks over IP, making it easier to spoof identities and use illegal accounts to launch a variety of attacks.” Such attacks include scripting “various flood, fuzzing and spoofing attacks,” according to VoIP blogger Rich Tehrani. “The hacker could set up multiple IPSec tunnels to various PDGs in the network or across multiple GPRS sessions [generating] up to 10,000 messages per second … equal [to] the traffic of 10 million users,”

Knowledge is power. I hope that exposing these threats will help you make your VOIP system more secure.

A Guide to Understanding the VoIP Security Threat

SK — Tue, 08 Dec 2009 11:18:30 +0000

“There’s no such thing as a bulletproof VoIP system, but there are things you can do to make your setup more secure”

At its heart, a VoIP system is a data network. This means VoIP deployments are vulnerable to the same internal and external threats that plague any enterprise data local area network (LAN) or wide area network (WAN).

Enterprises pondering voice over Internet protocol (VoIP) primarily focus on the technology’s cost benefits. Yet, in their zeal to converge voice and data networks and shave telephony costs, many organizations are failing to adequately consider VoIP’s single drawback: security.

Like Seinfeld’s George Costanza and the cashmere sweater with the little red dot, most VoIP supporters would prefer to ignore the ugly defect that mars their otherwise stainless technology. Unfortunately, VoIP’s little red dot has the potential to cripple enterprise VoIP systems. Worse yet, VoIP’s security gaps threaten to wreck havoc in several different, often insidious ways.

In-Stat, a US technology research firm, predicts that the number of business IP phones sold will grow from 9.9 million in 2006 to 45.8 million in 2010. Yet, the company ominously notes that over 40 percent of the enterprises it surveyed don’t have any specific plans for securing their VoIP deployments. Additionally, when asked to rate their VoIP security knowledge, most enterprise managers In-Stat contacted characterized themselves as being “somewhat knowledgeable,” the lowest rating the survey offered.

Locking Down Your System

There’s no such thing as a bulletproof VoIP implementation, but there are a handful of fundamental steps you can take today to ensure that your system, or the systems that you’re planning, will be highly secure.

According to network vendor Cisco, preventing unauthorized access to the network is a smart first step in a voice security program. For an additional layer of protection, in case somebody does gain unauthorized access, organizations can also encrypt voice traffic. Voice and video-enabled VPN (V3PN) technology, available in many routers and security appliances, encrypts voice as well as data traffic using IP Security (IPsec) or Advanced Encryption Standard (AES). Encryption is performed in hardware so that firewall performance is not affected.

Many security experts also recommend limiting VoIP data to a single virtual local area network (VLAN). A VLAN will keep voice network traffic hidden from data network users, providing an additional layer of security. The technique can also limit the scope of damage to the VLAN in the event of an attack. An additional side benefit is that a VLAN help prioritize VoIP data over other types of network traffic.

When creating the VLAN, be sure to place its equipment behind separate firewalls. This practice will restrict traffic crossing VLAN boundaries to applicable protocols and prevent viruses and other kinds of malware from spreading from clients to servers. When looking for firewall technology, be sure to examine products that support both leading standards: Session Initiation Protocol (SIP) and the International Telecommunication Union’s H.323 protocol.

Data and Physical Security

By now, just about everybody is aware of the need for packet data encryption to safeguard VoIP transmissions. Yet call signaling encryption is important as well to prevent hackers from misdirecting or otherwise interfering with call traffic.

To install multiple encryption layers, turn to Transport Level Security (TLS), which encrypts the entire call process. The Secure Real Time Protocol (SRTP) is useful as well for encrypting communication between endpoints.

A secure gateway, properly configured, is a VoIP system’s cornerstone. The gateway will limit system access to authenticated and approved users while keeping hackers safely on the outside. Gateways themselves, as well as the networks that lie behind them, can be protected through the use of a stateful package inspection (SPI) firewall and network address translation (NAT) tools.

Eternal Vigilence

VoIP security requires constant vigilance. This means monitoring the network for suspicious activities, as well as maintaining the operating system and VoIP applications. Be sure to install updates, particularly security patches, as soon as they become available. Consider using an operating system that has been “hardened” to deflect hacker attacks. It’s also important to disable non-essential operating and application services, since hacker can exploit these pathways to enter your system.

Ethernet ports are also prime hacker entry points. You can help keep the bad guys out of your network by using management tools that limit access to authenticated and pre-approved users and devices. You may also want to bar softphones from your system, since these products are vulnerable to malware and can also be imitate IP and MAC addresses when linked into the network via an RJ44 port.

Building redundancy into a VoIP system can help it better withstand hacker attacks as well as equipment failure. Multiple gateways, nodes, routers, servers and power supplies make a system more resilient and reliable.

Final Point

The good news is that VoIP threats are still a largely theoretical issue. So far, few enterprise VoIP networks have experienced anything close to a serious hacker attack. But complacency shouldn’t lull enterprise VoIP adopters into a false sense of security. Enterprises should strive to follow security best practices and demand that VoIP technology vendors build adequate safeguards into their products. Doing anything less is to court disaster.

VOIP: The Good and The Bad

SK — Tue, 08 Dec 2009 11:11:17 +0000

In the recent months, VOIP services have gotten more and more attention – both for individual and business purposes. Sometimes, though, we get all caught up in the hype that we may not really see the details. Is VOIP really all that it is made out to be? Does it really bring about advantages? If so, what are they? On the other hand, does VOIP have disadvantages? And if so, what are they?

Let’s look at the positive side first. What are the benefits that VOIP offers? Of course, the benefits vary depending on the specific VOIP service provider. More so, these will depend on the specific features of the VOIP provider.

Unlimited calls for a very low price.
This benefit is actually quite dependent on the plan that you take on. For certain plans, VOIP users can make unlimited phone calls for a very low fee. This could mean that a flat fee is charged for a certain period of time, with no limits as to how many calls can be made. For other plans, it could be that the charges are made per call – for a very low fee still. In any case, the bottom line is that more calls can be made – long distance especially – for much cheaper.

Charging is done per second.
Again, this depends on the VOIP provider and the specific plan. For some VOIP plans that charge per call, however, the calls are charged per second and not by the minute. This makes for a lot of savings as you only really pay what you use.

Mobility and flexibility.
There are many VOIP platforms today which can be used anywhere in the world – as long as the user has a computer and a fast Internet connection. This makes for a lot of convenience as we all know that almost anywhere you go, you can connect to the Internet. This is particulary efficient and cost effective for business people who travel all the time.

Now how about the downside? VOIP surely has some disadvantages as well.

Security.
Like it or not, there are some security threats to VOIP. Generally, though, service providers are handling these issues the best way they can. This is not to say that security threats will not come up in the future, though.

Dependency on Internet speed.
The main thing that makes VOIP great can also be its downfall. Call quality is highly dependent on the speed of the Internet connection. If the Internet speed is not enough, then VOIP calls just might not be possible.

Your Telco Might Be Ripping You Off

SK — Tue, 08 Dec 2009 11:08:22 +0000

VoIP is traditionally thought of as an alternative to the plain-old-telephone system (a.k.a. “POTS” or public switched telephone network). True, it’s usually cheap and would require none of the usual expensive switching equipment that telephone companies usually use. However, users of traditional telephone systems may not be aware that their telcos are ripping them off by routing calls via the Internet but still charging access fees.

Long distance and overseas calls are usually expensive because of these reasons: For international calls, international gateway facilities charge your local telephone company access fees whenever you dial someone from outside the counrty. For long distance, the same system applies, but this time, the telephone company of the person you’re calling charges your telco for access–or they both have access charges against each other.

The reason they feel they can justify charging expensive fees is because the international gateway facilities (IGFs) are expensive to set up and maintain, and they usually have to comply with regulatory requirements (such as franchises, permits, and the like) before they can run their business. In contrast, Internet service providers are usually viewed as value-added service providers instead of utilities–hence lower barriers and costs to entry.

However, with the onset of VoIP as a popular alternative, telcos have started to route their calls through the Internet, too. I have personally seen several prepaid services (that use calling cards) marketing themselves as good alternatives to the regular telephone system, but not exactly as VoIP solutions. You still have to contact your correspondents by dialing through your local phone system.

What I do notice when I try these services for calling overseas is that the voice quality is different. I mean, you will definitely be able to distinguish between a phone call that’s routed through an analog circuit, and one that’s packet-based. So what does this mean? My telephone company is actually routing my calls through VoIP.

What’s worse is that some telcos I know are charging the same rates for calls, but they’re routing the voice calls via the Internet. This means they’re no longer paying the IGF charges, but still charging us end-users the same rates!

Asterisk Vulnerability Discovered

SK — Tue, 08 Dec 2009 11:03:01 +0000

Here is something for all Asterisk users out there. Though we may all be very enthusiastic about Asterisk and the service it provides, we have to be practical and keep our eyes open for vulnerabilities. Even the people over at Digium do not act like ostriches and keep their head buried in the sand – I guess most other service providers act the same way. They are always on the look out for weaknesses that other unscrupulous individuals may take advantage of.

Recently, Joel R. Voss aka. Javantea reported a vulnerability in Asterisk systems that may result in denial of service. Many other sites and blogs have subsequently spread the word about the possible problems that may arise from the vulnerability. People over at Digium themselves have released an advisory about the issue. They have also released work arounds that could help solve the issue and avoid potential problems that may arise from it.

Below is the description of the vulnerability as well as other important details that you may need to resolve the issue. This was taken from Secunia:

Description:
A vulnerability has been reported in Asterisk, which can be exploited by malicious people to cause a DoS (Denial of Service).

The vulnerability is caused due to improper verification of ACK responses during IAX2 handshakes, which can be exploited to spoof an IAX2 handshake and cause a DoS via high bandwidth usage.

The vulnerability is reported in the following versions:
* Asterisk Open Source 1.0.x (all versions)
* Asterisk Open Source 1.2.x (all versions prior to 1.2.28)
* Asterisk Open Source 1.4.x (all versions prior to 1.4.19.1)
* Asterisk Business Edition A.x.x (all versions)
* Asterisk Business Edition B.x.x (all versions prior to B.2.5.2)
* Asterisk Business Edition C.x.x (all versions prior to C.1.8.1)
* AsteriskNOW 1.0.x (all versions prior to 1.0.3)
* Asterisk Appliance Developer Kit 0.x.x (all versions)
* s800i (Asterisk Appliance) 1.0.x (all versions prior to 1.1.0.3)

Here’s to hoping that you will be able to take care of the vulnerability or Just Leave it to AsteriskGeekz

Voip Predictive Dialer – Build your Call Center

SK — Tue, 08 Dec 2009 10:37:12 +0000

Today, I will be discussing about Call centers and how small businesses can run a call center with voip predictive dialer. I have always been fascinated by how VOIP can empower some of the industries, especially the upcoming call center and BPO industry which relies heavily on “Making and receiving calls”. A fraction of delay or missed call can cost some $$ to the industry. The Industry is no more in the seed stage and it has grown faster than industry analyst expected.

Although call centers are still costly to maintain, operate or even outsource. Big companies can afford to pay the price but the small companies are often stuck with in-house calling facilities. The in-house call facilities are however not up-to the mark. However with Voip Predictive dialer the picture might be different in near future. People can build their own call center with a easy to use Voip predictive dialer.

Now you must be wondering What is a Predictive dialing? Predictive dialing is simply automating the outbound dialing process to effectively manage your calls with little or no manual intervention. Voip predictive dialer basically does the same thing with VOIP technology as backbone.

some of the common Features in Voip predictive dialer:

* Multiple campaigns can be run simultenously.
* Auto-Transfer calls between agents (based on availability)
* Auto-dialing to customers
* Auto-Scanning Voice Mail system and auto-dialing the missed call numbers
* Call Logging, recording and management

On top of this, the predictive system can be easily integrated with your exisiting CRM system making it easier to control and store data related to your customers. There are many providers who also offer predictive dialing bundled with their CRM products . Although if you buy a standardVoip Predictive dialer , it will cost you anything from $300 ++, which isn’t really a great cost considering you will be able to manage your customers more effectively and data mining could be more relevant.

Most predictive dialers provide most robust, feature-rich, and innovative predictive dialer on the market today. Built on the latest technologies, this cutting-edgecall center solution combines sophisticated VoIP predictive dialing capabilities with a comprehensive lead management system. So, call centers of all sizes can achieve immediate value – operating more efficiently and cost-effectively, while boosting sales, market share, and profits.

Imagine running your business with your own call center like only Cisco or Nortel can think of. You can do it yourself.

In my next series of this article, I will be comparing voip predictive dialer available in the market based on their cost and features.

VOIP with Asterisk

SK — Mon, 07 Dec 2009 21:35:27 +0000

VoIP (Voice over Internet Protocol) and IP PBXs and Call Centers have become increasingly popular to enable larger businesses to gain many of the long-promised advantages of Internet Telephony. Until now, however, there has been no cost-effective way to virtualize VoIP services so they can be hosted, and delivered by managed service providers in the increasingly prevalent Communications-as-a-Service (CaaS) and emerging Cloud Computing models. What’s more, VoIP’s questionable quality and reliability combined with “techno speak” from the vendor community have slowed adoption by mainstream businesses and their solutions providers. Now, Globussoft has emerged as “Your NEW Asterisk Support Company” to lead the VoIP movement by bringing to businesses of every size and type Hosted Virtual VoIP with the Quality of Telephony™ formerly associated with conventional telephony, TDM PBXes and the legacy PSTN.

What is Asterisk PBX

Asterisk is the world’s number one open source PBX system. It offers flexibility unheard of in the world of proprietary telephone systems. Asterisk facilitates creation of advanced communication solutions.

Asterisk is licensed as open source under the GNU General Public License
(GPL) and at the time of writing is the most popular open source software available today.

Asterisk as a telephony switch

Asterisk has many uses, but primarily as the core of an IP PBX, switching calls, managing routes, enabling features, and connecting callers with the outside world via IP, analogue or digital
(ISDN2/ISDN30/E1) trunks.

Asterisk runs on a wide variety of operating systems but was designed on Linux. It provides all of the features you would expect from a PBX including many advanced features that are often associated with high end, expensive, proprietary telephone systems. Asterisk’s architecture is very flexible and supports voice over IP in many protocols.

Need an IVR? Asterisk has it covered. Need a conference bridge? Asterisk has that covered too. What about an automated attendant? No problem. How about a replacement for your aging voicemail system? That too. Unified messaging? Yep. Telephony interface for click2call via your web site?
Consider it done. Asterisk does it all.

Asterisk for your Call Center

Asterisk PBX has been used by call centers worldwide. Call center solution providers have built complete Automatic Call Distribution (ACD) systems based on Asterisk PBX. Asterisk has also breathed new life to existing call center systems by adding support for home based agent capabilities, predictive dialling, and much more.

Asterisk has become the basis for thousands of communications solutions the world over. If you need to communicate, Asterisk is your answer.

Asterisk Support and Installation Services

We offer various services for Asterisk, Asterisk installation, Asterisk configuration, Dialplan programming, IVR Designing, AGI scripting, manager api, Fax Support , etc.

We have team of experts having wide range of skills and extensive experience in Asterisk. We have successfully completed various projects based on Asterisk.

We have expertise in various opensource solutions based on Asterisk such as A2Billing , FreePBX , TrixBox etc. and many more.

Currently we offer following services for Asterisk.

Asterisk installation
Asterisk configuration
Dialplan programming
AGI scripting
IVR Designing
Fax Support
Trixbox installation
FreePBX installation
A2Billing Installation
AsteriskGUI Installation

Asterisk Installation

Asterisk PBX is open-source, meaning there is no charge for the software itself, yet it offers a huge array of telephony features normally associated only with high end telephone switches costing tens of thousands of pounds.

Furthermore, Asterisk PBX is highly scalable so it can grow with your business as and when the need arises

At Globussoft we offer complete Asterisk installation service including installation of prerequisites, necessary packages and all Asterisk packages such as zaptel, libpri, Asterisk, Asterisk addons and Asterisk sounds. We can install Asterisk on most of the unix like systems such as Linux, BSD and MacOSX

We can install Asterisk on your server remotely with all prerequisites and necessary packages. You just need to give us your server details, root access and relax. We will do Asterisk installation on your server within less time. A complete peace of mind service for customers.

Asterisk Configuration

Globussoft offers complete Asterisk configuration services. We offer following Asterisk configuration services.

Configuration of ZAP channels:
We can configure various analog interface cards and digital interface cards for Asterisk..
Configuration of VoIP providers for PSTN termination and origination
Generally phone service via VoIP costs less than equivalent service from traditional sources. Therefore people use VoIP providers to terminate international calls. We have experience in configuring various SIP, IAX2 and H323 termination providers. We can also suggest you few good VoIP termination and origination providers.
Configuration of extensions
We can configure SIP/IAX2/H323/ZAP extensions for you. To connect appropriate softphone or hardphone, we need to create extension for it in Asterisk. After creating extension/user, user can connect soft or hard phone to asterisk and start calling.
Configuration of ACD, Agents and Queues
Asterisk has built-in support for call center features. If you want to start an inbound or outbound call center, we can help you with it. With Asterisk setup and operating cost of the call center goes down to earth. We can configure ACD with IVR and create various queues according to your needs. We can configure agents for call center and assign them to different queues.
Configuration Find me, Follow me
One nice feature of Asterisk is follow me. You can receive calls to your extension. If you are not available, it will ring your cell phone or PSTN phone and finally go to voicemail if no one answers the call.
Configuration of Meet me
Asterisk is a very nice conferencing platform. We can configure various conference rooms with and without pin numbers. We can also configure extensions for administrative operations in particular conference rooms.
Configuration of Voicemail
Asterisk has the ability to store voicemails and send them as an attachments to configured email addresses. We can configure Voicemail accounts with email addresses and format of the emails so that you can have customized voicemail to email features.

Dialplan Programming

Dialplans can be defined in extensions.conf or database using realtime Asterisk configuration in Asterisk. Basic Asterisk dialplans can be defined easily in no time. But if you want to create Asterisk dialplans for billing solutions, call center solutions, IVR or any other custom application then creating the dialplans is not so easy. We have expertise in creating complex Asterisk dialplans. We have created many Asterisk dialplans for billing solutions, IVRs, conferencing systems and many other applications.

We can configure Asterisk dialplans for your billing solution which can consist of authenticating users from agi scripts, checking balance, getting rates for the destination, insert custom CDRs (Call Data Records), etc.
We can create simple to complex IVRs according to your needs. We can even use text to speech and speech to text capabilities to offer outstanding services from Asterisk PBX.
We can create more complex customized Asterisk dialplans for various other applications such as conferencing, click to call, notification systems and many more.

Freepbx Installation

FreePBX is a full-featured PBX web application. If you’ve looked into Asterisk, you know that it doesn’t come with any “built in” programming. You can’t plug a phone into it and make it work without editing configuration files, writing dialplans, and various messing about.

FreePBX simplifies this by giving you pre-programmed functionality accessible by a user-friendly web interfaces that allows you to have a fully functional PBX pretty much straight away with no programming required. Some of the features that FreePBX supports out of the box are:

Unlimited number of Voicemail boxes

“Follow Me” functionality

Ring Groups with calls confirmation (so if, eg, a cellphone is out of range and diverts to voicemail, all the other phones keep ringing)

Unlimited number of Conferences (limited by available CPU power – about 300 simultaneous users in conferences on a P4 3ghz – 600 with a dual core!)

Paging and Intercom functionality for man SIP phones that support it.

Music on Hold (via MP3s, or streamed off the internet)

Call Queues

And many other features

FreePBX is built on the LAMPA™ stack (Linux, Apache, MySQL, PHP and Asterisk). It’s a modular system, with click-to-install plugins downloadable over the internet from the online module repository.

FreePBX Features at a Glance:

Add or change extension and voicemail accounts in seconds

Native support of SIP, IAX, and ZAP clients (other endpoints are supported through custom extensions)

Supports all Asterisk supported trunk technologies

Reduce long distance costs with LCR

Route incoming calls based on time-of-day

Create interactive Digital Receptionist (IVR) menus

Design sophisticated call groups

Manage callers with Queues

Upload custom on-hold music (MOH)

Search company directory, based on first or last name

Detect and receive incoming faxes

Share administrative duties

Backup and Restore your system

Save audio recordings of calls

View call detail reporting with asterisk-stat

View extension and trunk status with Flash Operator Panel

View conversation recordings with Asterisk Recording Interface (ARI)

AGI Programming

One of the cool features of Asterisk is AGI scripting. AGI is the Asterisk Gateway Interface, an interface for adding various functionalities to Asterisk with many different programming languages such as Perl, PHP, C, Pascal, Bourne Shell, etc.

Basically there are three types of AGI scripts:

AGI:
Using AGI, you can control dialplan and do various operations with database, system, manager api and almost everything.
EAGI:
EAGI gives the application the possibility to access and control the sound channel in addition to interaction with the dial. EAGI can be used to plan. EAGI can be used to handle voice stream for recording voice or doing speech to text.
DEADAGI:
DEADAGI gives access to a dead channel, after hangup. DEADAGI is generally used to do cleanup operations when call is hang up.

We have expertise in programming AGI scripts. We generally use Perl as a programming language for AGI scripting. We have created many AGI scripts for various applications such as billing solutions, IVR systems, conferencing system, click to call, etc.

Compiling the applications

SK — Sun, 06 Dec 2009 21:32:53 +0000

The order we compile the applications is important. Compiling them out of order would mean that some of the dependancies of an application might not be fulfilled and things wont work as expected. You will need to be inside the actual directory of the source you are trying to install

Compiling libpri
Libpri dnt have a configure script so just it needs Make and Make Install
cd /usr/src/libpri make make install

Compiling Zaptel
cd /usr/src/zaptel ./configure make make install make config

Compiling Asterisk
cd /usr/src/asterisk ./configure make make install make samples make config

Compiling Asterisk-Addons
cd /usr/src/asterisk-addons ./configure make install make samples

Setup the build environment

SK — Sun, 06 Dec 2009 21:21:22 +0000

Because we are installing from source, they (the source files) will rely on other components of the operating system. The build environment sets up those other components so there wont be errors during the building of the source.

Install build environment
aptitude install linux-source aptitude install kernel-package aptitude install g++ aptitude install libncurses5-dev aptitude install linux-kernel-headers aptitude install sqlite aptitude install libnewt-dev aptitude install libusb-dev aptitude install zlib1g-dev aptitude install libmysqlclient15-dev aptitude install libsqlite0-dev

Now we configure the linux-source that we just downloaded, the other apps have been already been configured by Debian.

configure source
cd /usr/src tar -xjf linux-source-(version) ln -s /usr/src/linux-source-(version) /usr/src/linux cd /usr/src/linux make oldconfig make prepare make prepare scripts

Congratualtion..! on setting up the build environment. As well begin is said to be Half Done.Lets Move on to next module and compile the application in Milestone 3

Asterisk Installation: Future of Telephony

SK — Sun, 06 Dec 2009 20:54:08 +0000

We will be using the default install of Debian and installing Asterisk from source and compiling.

Debian in my opinion is the best Linux distribution, Asterisk is the best PBX available to us. I have used both of these things in the past and find them a great combination as do many others.

We really need to appreciate all the people who give us this great software free of charge, they are truly generous.

Consider that if you use a setup like this in a business with 5 or more phones, you can save thousands of dollars a year in call costs, hardware and licensing.

Prerequisites
You will need

Internet connection
Spare computer with at least 10gb hard disk space
Pentium class machine somewhere above 500mhz at least or AMD equivalent

Some will disagree on the CPU requirements mentioned here and I am not going to argue about that. What I will say is that for a 100 person call center, this pc is not going to cut it. In all cases, monitor the cpu and memory usage on your machine under different call loads. The easiest way to do that is to login to the console and run “top”

Timeline

This howto was written from a bare metal install and took many days to research, install and document. It should take you considerably less time to get running. Somewhere around the 3 hour mark you should be able to dial a couple of extensions. Following this, the configuration fun is endless.

What will I end up with?

This howto is not a magic bullet. Finishing this howto will ensure you have a working VOIP server that you can add extentions and communicate between them. There are no hardware components required (other than the machine itself) You will also be able to add conference rooms and communicate with the outside world.

Can I use this for a company or business?

Yes, this is a solid and stable build suitable for deployment in an office environment. Be aware there is no security in its basic form. Things such as firewalls and usernames and passwords should be added to protect the box from bad things. That is a whole other document.

Getting Started

Install Debian Etch

If you are not familiar with Debian, have a look at the step by step Debian install document installing_debian then come back here and continue.

Installing Asterisk
After doing the above, we need to get Asterisk from the Digium website. We use the source and compile it ourselves.

Downloading the source files
At this point we are going to get the source files so make sure you are logged in as root and change to the linux src directory

cd /usr/src

Get Asterisk
wget -c http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.21.2.tar.gz

i am using Asterisk-1.4.21.2, as i found this highly stable.

Get libpri
wget -c http://downloads.digium.com/pub/libpri/releases/libpri-1.4.7.tar.gz

Get Zaptel
wget -c http://downloads.asterisk.org/pub/telephony/zaptel/zaptel-1.4.12.1.tar.gz

Get Asterisk-Addons
wget -c http://downloads.digium.com/pub/asterisk/releases/asterisk-addons-1.4.7.tar.gz

This is what we need to install asterisk. Lets Move on Milestone 2

Installation

SK — Sun, 06 Dec 2009 20:32:17 +0000

Let’s begin our journey into the world of ASTERISK.

While Asterisk is the most widely used open source PBX solution, it hasn’t been easy for the average user to take advantage of it. A good, basic knowledge of Linux is a fundamental requirement, as is the ability and nerve to handle the telephony hardware and Asterisk configurations.
Broad support is available through Web-based developer forums for those who take this do-it-yourself approach, but even Bill Miller, vice president of product management and marketing for Digium Inc., Asterisk’s main developer, describes this as the “techie” route for potential Asterisk users.

Before we can start, make sure you have the following:
1. Computer with installed Linux OS (For this please check our other posts).
2. Asterisk’s installation packages.
3. AsteriskGeekz

Note : You do not need any special hardware to install and run Asterisk, not even a sound card is required.

Before downloading the Asterisk source code, the installation requires users to already have a Linux operating system with kernel version 2.4 or higher, and the bison, ncurses, zlib and openssl libraries with their associated developer packages.
The Asterisk code and other packages needed to install Asterisk can be downloaded in several ways.
- As tar formatted files (tarballs) from ftp.digium.com
- From Digium’s CVS server (password anoncvs) using the commands:
# cd /usr/src
# mkdir asterisk
# export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot
# cvs login
- From Digium’s SVN server (once an SVN package has been installed) using commands such as:
# svn checkout http://svn.digium.com/svn/asterisk/branches/asterisk-current
If Digium’s own hardware is being used with the installation then both the zaptel and libpri filed should be downloaded along with the Asterisk files.
Installing Asterisk for most Linux distributions is relatively straightforward. Once in the folder in which the source codes are kept issue the following commands:

# cd../asterisk
# make clean; make install
If the full installation with both zaptel and libpri is needed, those should be installed in order as:
# cd../libpri
# make clean; make install
# cd../zaptel
# make clean; make install
# cd../asterisk
# make clean; make install

Seems Interesting but this is what other documentations of Asterisk quoted, Lets Dive into the Future of Telephony with AsteriskGeekz.
I have divided Asterisk Installation into different Milestones MileStone 1