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iamjackmorris · 18 days ago

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How FreeSWITCH Modules Streamline Custom VoIP Development

In today’s business environment, flexibility and scalability are not advantages — requirements. With more organizations, large and small, seeking to improve their voice solutions, FreeSWITCH development has emerged as a powerful way to build customized, future-proof VoIP systems. And underpinning that flexibility are FreeSWITCH modules, which allow programmers to tailor VoIP platforms to precise business requirements.

Learning About FreeSWITCH and Modular Architecture

FreeSWITCH is an open-source telephony system used and envied for its flexibility and performance. What really sets it apart, though, is the way that it is modular. Instead of a rigid one-size-fits-all solution, FreeSWITCH allows you to pick from an incredibly large selection of modules, each providing some features such as call routing, conferencing, IVR, etc.

If you want to get a step-by-step walkthrough on how these modules work and the types that are available, don’t overlook this in-depth tutorial on FreeSWITCH modules. It’s a great tutorial on how to learn about FreeSWITCH's modular mechanism and practical application.

Major Benefits of Using FreeSWITCH Modules in VoIP Development

1. Unlimited Customization

With FreeSWITCH modules, you can design a VoIP system that fits your company like a glove. Do you require advanced call recording? There’s a module for it. Do you wish to connect with your CRM or billing system? Modules make it possible. Such customization is a game-changer for companies with unique workflows or regulatory requirements.

2. Scalability on Demand

As your business grows, your communications needs evolve. FreeSWITCH’s modular architecture allows you to add new functionality or expand call density merely by enabling or building additional modules. Your VoIP solution can easily scale without costly rebuilds.

3. Faster Development and Deployment

Modules are plug-and-play. Developers can leverage existing modules and accelerate project timelines, and enjoy the flexibility of designing custom modules for specific needs. This blend of speed and agility is a major advantage in today’s fast-paced market and environment.

4. Cost Efficiency

Since FreeSWITCH is open-source and modular, you only pay for what you need. You don’t need to invest in cumbersome, do-everything-but-the-kitchen-sink solutions that contain features you won’t ever use. This targeted approach to FreeSWITCH development maximizes budgets for businesses while delivering access to enterprise features.

5. Future-Proofing Your VoIP System

Telecom technology is ever-evolving. FreeSWITCH modules make it easy to adapt to new standards, protocols, and integrations. Be it the support for the latest codecs or facilitating WebRTC capabilities, modules keep your system up-to-date and competitive.

In-Demand FreeSWITCH Modules and Their Uses

mod_conference: Provides support for advanced conferencing support for team collaboration, and virtual meetings.

mod_voicemail: Provides solid voicemail support.

mod_cdr_csv: Provides call detail records in detail for analytics and billing.

mod_xml_curl: Dynamic configuration and web service or external database support.

mod_webrtc: WebRTC support for browser-based video and calling.

For a more technical overview of these and all other FreeSWITCH modules, check out the FreeSWITCH modules blog

Real-World Example: Custom IVR with FreeSWITCH Modules

Imagine a growing customer support center that needs a custom IVR system to route calls based on language, department, or customer status. FreeSWITCH modules allow developers to quickly create a custom IVR solution that connects with databases, CRM systems, and even voice recognition through AI, all without having to rebuild from the ground up.

Why Expert FreeSWITCH Development Matters

Whereas modularity in FreeSWITCH makes it accessible, its full potential can only be achieved through profound technical acumen. Changing, incorporating, and supporting modules — particularly in advanced or heavy-traffic scenarios — takes expert developers familiar with the platform as well as your company’s objectives.

If you need to develop a tailor-made VoIP solution or improve your existing one, it makes sense to hire FreeSWITCH developers with expertise in module development and integration. The correct experts will help you unleash the full potential of FreeSWITCH and ensure your communications are secure, dependable, and scalable.

Conclusion

FreeSWITCH modules are the secret ingredient in some of the most cutting-edge and cost-effective VoIP solutions out there today. With a modular freeswitch development strategy, firms can create personalized communication systems that are tailored to their requirements, cost less, and deliver better performance.

Ready to make your VoIP system something incredible? Start by finding out what the power of FreeSWITCH modules can bring — and work with specialists at Hire VoIP Developer, we can turn your vision into reality.

#FreeSWITCH Modules #FreeSWITCH Development #FreeSwitch #VoIP Solutions #VoIP Developer #Hire VoIP Developer #FreeSWITCH Solutions

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crmvoipsoftwaresolutions · 2 years ago

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FreeSWITCH Development Guide for Aspiring FreeSWITCH Developers

FreeSWITCH is a free and open-source telephony platform designed to facilitate the creation of voice and messaging applications launched in 2006. It provides a cost-effective alternative to proprietary PBX systems. It can be used as a standalone or integrated with other IP-based systems. FreeSWITCH supports many TDM and VoIP protocols and can be deployed on physical hardware or virtualized environments.

It is a powerful and flexible open-source telephony platform that can be used to build various communications solutions. The FreeSWITCH architecture is designed to be highly modular and scalable, making it ideal for building everything from simple PBX solutions to large carrier-grade deployments.

In this section, we will take a high-level look at the FreeSWITCH architecture, including its core components, its modular design, and how it can be scaled to meet the needs of any size deployment.

Advantages and Disadvantages of FreeSWITCH Development

There are many reasons to use FreeSWITCH for your next VoIP development project. We have enlisted some of the key advantages and disadvantages of using this solution:

Advantages:

FreeSWITCH is a compelling VoIP platform that can efficiently handle many calls.

It is very stable and has been battle-tested in production environments.

The FreeSWITCH community is large and active, so you can get help and support when needed.

Disadvantages:

FreeSWITCH can be complex to install and configure properly.

The learning curve may be steep for new users, as there is a lot of documentation to read through.

Setting Up a Development Environment for FreeSWITCH

Assuming you are starting with a fresh Ubuntu installation, the first thing you need to do is install some basic dependencies required for compilation:

sudo apt-get install -build-essential autoconf automake libtool curl make unzip

With the dependencies out of the way, you can now proceed to clone the FreeSWITCH repository from GitHub and checkout the “stable” release branch:

git clone https://freeswitch.org/stash/scm/fs/freeswitch.git FreeSWITCH cd freeswitch git checkout v1.8.3

Now that you have the source code, all you need to do is to compile it. Depending on your system resources, this process can take a while, but once it’s done, you will have a working FreeSWITCH installation.

Writing Your First Application in FreeSWITCH

Assuming you have already gone through the process of installing FreeSWITCH, the next thing you need to do is write your first application.

There are two ways to write applications in FreeSWITCH

Event Socket Library (ESL)

mod_lua module.

This article will focus on using mod_lua to write our applications.

When you write an application in mod_lua, you create a file with a “.lua” extension and place your FreeSWITCH installations in the “scripts” direction. For our example, we will create a file called “helloworld. lua” in this directory.

The first thing you need to do is tell FreeSWITCH to load the mod_lua module. You can do this by adding the following line to our “modules. conf” file:

load-module mod_lua

Once we have done this, we need to edit our “hello-world.lua” file and add the following code:

This function gets called when the Lua script is loaded:

function hello()

— Print *** to the console

freeswitch.consoleLog(“notice”, “Hello World!”)

— return control to FreeSWITCH

return true

end

How to Plan Your Approach?

When you’re ready to start developing for FreeSWITCH, it’s essential to have a clear plan and strategy in place. This will help you stay organized and on track as you work on your project. Here are a few tips to keep in mind for successful VoIP development in FreeSWITCH:

Familiarize Yourself with the FreeSWITCH Documentation

The first step is to familiarize yourself with the FreeSWITCH documentation. This will help you understand the basics of how FreeSWITCH works and what you need to do to get started. The documentation is extensive, so take your time and read through it carefully.

Choose the Right Language for Your Project

FreeSWITCH development supports various programming languages, so you’ll need to choose the one that’s right for your project. When deciding, consider factors such as performance, ease of use, and support. Once you’ve selected a language, be sure to spend some time learning the basics before diving into development.

Use Version Control from the Start

Troubleshooting Common Problems with FreeSWITCH Development

Version control is essential for any developer, especially when working on FreeSWITCH projects. Be sure to set up a version control system (such as Git) at the start of your project and use it regularly as you work. This will make it easier to track changes, revert if necessary, and share your code with others.

Troubleshooting Common Problems with FreeSWITCH Development

If you’re having trouble with FreeSWITCH development, there are a few common problems that you can troubleshoot. First, make sure that you have the latest version of FreeSWITCH. You can check for updates by running “git pull” from the FreeSWITCH directory. If no updates are available, check for any new releases of FreeSWITCH by visiting the website or checking the mailing list archives.

If you’re still having trouble, try checking the debug logs. The debug logs contain detailed information about what FreeSWITCH is doing and can be very helpful in troubleshooting problems. To enable debug logging, edit your log file (freeswitch.log) and change the “level” setting to “9”. Once you’ve done this, restart FreeSWITCH and reproduce the problem. After reproducing the problem, check the log file for any clues about what might be causing it.

If you’re still having trouble, try posting on the FreeSWITCH mailing list or IRC channel. Be sure to include as much information as possible, such as your operating system, version of FreeSWITCH, and steps to reproduce the problem. Someone can help you solve your problem more quickly if they have all this information.

Conclusion

In conclusion, FreeSWITCH development is a great way to get involved in the open-source software community. With its powerful features and simple setup process, it’s easy to see why FreeSWITCH is quickly becoming one of the most popular VoIP platforms. We hope this article has given you the knowledge to start your journey into FreeSWITCH solutions development and that you’ll succeed as you progress with your project.

Are you looking for the best FreeSWITCH development company?

AC InfoSoft is one of the leading FreeSWITCH development companies that offer different FreeSWITCH related services like FreeSWITCH installation, support, custom development, and more. To learn more about this FreeSWITCH development services, please visit

https://www.acinfosoft.com/freeswitch-services-solutions/

#FreeSWITCH Development #voip development

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voiptechsolutions · 7 years ago

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VoIPTech is a leading international VoIP provider in India associated with the improvement and plan of VoIP services in simultaneousness with the requests of the customer endeavour. We are considered as one among the head and trusted VoIP service provider for all the communication needs.

VoIP Development Services

FreeSWITCH

WebRTC

Asterisk Development

OpenSIPS

To know more visit http://www.voiptechsolutions.in/voip-development/ today or call us at +91-7008220621 .

#VoIP service Providers #VoIP provider #Cloud based Phone system #pbx phone system #IP PBX solution #voip services for business #PHP Development Company #Bootstrap web development #Magento Development Company #Freeswitch module development #Opensips module development #Conferencing solution Provider #Leading International VoIP provider

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terabitweb · 6 years ago

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Original Post from Rapid7 Author: William Vu

Towards a more reliable BlueKeep exploit

zerosum0x0 recently improved the reliability of our BlueKeep exploit after a little soul searching and a helpful cue from Worawit Wang.

In short, the exploit was developed in a lab without the Meltdown patch, which meant more frequent crashes in the wild against targets that have the patch installed — a high likelihood. You can read zerosum0x0’s full analysis on his blog. We’re just glad it wasn’t the lizard people causing those crashes.

Gaining access to Pulse Secure VPN servers

Earlier this year, Orange Tsai and Meh Chang were on a rampage through VPN server software, having discovered more than a few vulnerabilities in popular VPN solutions, such as Palo Alto Networks, Fortinet’s FortiGate, and Pulse Secure. They were even able to compromise Twitter via their bug bounty!

Starting with a contribution from Alyssa Herrera and Justin Wagner that exploits a file disclosure vulnerability in Pulse Secure’s VPN server, we created a finished module that will download any credentials, hashes, and sessions from a server, allowing an attacker to authenticate to the VPN, potentially as an administrator. A manual mode is also supported to download arbitrary files.

This leads us to the next phase, which uses a valid administrator session from the file disclosure to authenticate a post-auth, remote root RCE against the server, bypassing the software’s application whitelisting by using the env(1) command — which is happily permitted. The module can pop a root shell or run an arbitrary command on a vulnerable target.

A major overhaul of password cracking integration

The ever-reliable h00die graced us with a complete and total overhaul of our password cracking integration, notably adding new support for hashcat. Check out the pull request. It’s a doozy, and we can’t do it enough justice in this wrap-up alone!

New modules (14)

Xorg X11 Server Local Privilege Escalation by Narendra Shinde and Zack Flack, which exploits CVE-2018-14665

Bludit Directory Traversal Image File Upload Vulnerability by sinn3r and christasa, which exploits CVE-2019-16113

Pulse Secure VPN Arbitrary File Disclosure by wvu, Alyssa Herrera, Justin Wagner, Meh Chang, and Orange Tsai, which exploits CVE-2019-11510

Pulse Secure VPN Arbitrary Command Execution by wvu, Meh Chang, and Orange Tsai, which exploits CVE-2019-11539

CMS Made Simple Authenticated RCE via object injection by Daniele Scanu, which exploits CVE-2019-9055

FreeSWITCH Event Socket Command Execution by bcoles

FusionPBX Command exec.php Command Execution by bcoles

FusionPBX Operator Panel exec.php Command Execution by Dustin Cobb and bcoles, which exploits CVE-2019-11409

Password Cracker: AIX by hdm), h00die, and theLightCosine

Password Cracker: Databases by hdm, h00die, and theLightCosine

Password Cracker: Linux by hdm, h00die, and theLightCosine

Password Cracker: OSX by h00die

Password Cracker: Webapps by h00die

Password Cracker: Windows by hdm, h00die, and theLightCosine

Enhancements and features

PR #11695 by h00die is a complete transformation of the cracking system, adding support for additional applications and hash types to be utilized during reversing of stored credential details. JtR has been migrated and Hashcat has been added using this pattern.

PR #12556 by bcoles bumps the maximum size for ASCII art banners to 65,535 bytes.

Bugs fixed

PR #12543 by layderv fixes several modules to use myworkspace_id instead of myworkspace.id, the former of which will check if the database is connected first, whereas the latter will crash if not connected.

PR #12570 by timwr changes the Msf::Post::Linux::Compile mixin to use the correct Failure class.

Get it

As always, you can update to the latest Metasploit Framework with msfupdate and you can get more details on the changes since the last blog post from GitHub:

Pull Requests 5.0.59…5.0.60

Full diff 5.0.59…5.0.60

We recently announced the release of Metasploit 5. You can get it by cloning the Metasploit Framework repo (master branch). To install fresh without using git, you can use the open-source-only Nightly Installers or the binary installers (which also include the commercial editions).

#gallery-0-5 { margin: auto; } #gallery-0-5 .gallery-item { float: left; margin-top: 10px; text-align: center; width: 33%; } #gallery-0-5 img { border: 2px solid #cfcfcf; } #gallery-0-5 .gallery-caption { margin-left: 0; } /* see gallery_shortcode() in wp-includes/media.php */

Go to Source Author: William Vu Metasploit Wrap-Up Original Post from Rapid7 Author: William Vu Towards a more reliable BlueKeep exploit zerosum0x0 recently improved…

#Rapid7

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vindaloo-softtech · 2 years ago

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Asterisk vs. FreeSWITCH: How Are They Different?

Asterisk and FreeSWITCH are the two most widely used platforms for developing VoIP solutions in the VoIP industry. They are both powerful enough for developers to create any intricate VoIP solutions for teamwork and communication in addition to being open-source technologies.

What is Asterisk? In 1999, Mark Spencer created Asterisk, a piece of software that distributes calls like an expensive PBX. His tiny group created the Asterisk call distribution and handling software for their business, Linux Support Systems, which was eventually renamed Digium.

What is FreeSWITCH? The Asterisk platform has drawbacks, which led to the creation of FreeSWITCH in 2006. Anthony Minessale, a well-known Asterisk developer, made the decision to create a software from scratch to address some of the alleged problems with the Asterisk platform. This was later known popularly as FreeSWITCH.

If you are wondering why to switch from traditional to VoIP calling, here is an article on “PSTN v/s VoIP”

How Does Each Technology Work? The heart of every Asterisk system is the dialplan. It is a scripting language, and the modules are used to give instructions to the Asterisk system through the configuration directory. Developers may implement a variety of capabilities, including call reception on a particular SIP channel, call connection to IVR, and dial plan-based call routing. Asterisk's configuration files are stored as standard text files.

FreeSWITCH adopts a different strategy; the system was created in C, and the core programming was better organized. FreeSWITCH uses processing threads that operate uniformly throughout memory, in contrast to the Asterisk approach, which gives each channel its own thread and memory space.

Basic Functionality On the most fundamental level, both technologies offer the majority of the same functionality. Voicemail, call recording, and IVR menus should be available on every FreeSWITCH Development or Asterisk-based switch on the market. With any design, the steps involved in constructing extensions and gateways are quite similar. Nevertheless, depending on the server's Memory and core performance, different numbers of users may be supported. For communication with other Asterisk systems, Asterisk needs the proprietary IAX protocol, but FreeSWITCH is not constrained in this regard.

MultiTenancy One of FreeSWITCH’s features is its capacity to support several tenants. As a result, many user branches may use a single FreeSWITCH system as unique entities under their own domains or subdomains. For multi-tenancy to function on Asterisk systems, costly proprietary solutions must be constructed on top of it at the time of Asterisk software development.

Clustering Asterisk is primarily made to operate on a single system. Using a single PBX server to install all roles is equivalent to placing all of your eggs in just one basket. Thus, it is advised to divide each system, performing a given function, into a distinct server for large-scale or enterprise- level PBX servers, i.e., distributed architecture. While FreeSWITCH solutions makes it simple for separate systems in the cluster to execute individual duties, accomplishing this with Asterisk is difficult.

IM abilities Systems like Asterisk and FreeSWITCH provide cutting-edge communication features including conference, video calling, and chat. Unfortunately, the majority of Asterisk systems require on an add-on for IM functionality, which businesses must pay extra fees for. With FreeSWITCH, the only requirements are that the XMPP service be enabled and that the end devices be correctly set up for IM.

Device Deployment Capabilities On either FreeSWITCH or Asterisk networks, device deployment varies greatly. Several endpoint management modules for IP phones and softphones are supported by Asterisk, however, access to the provisioning software costs around $100. Compared to Asterisk, FreeSWITCH offers a significantly smaller selection.

Wrapping Up There is no discernible difference between a well-setup system running Asterisk or FreeSWITCH for the end user. In fact, FreeSWITCH provides a wider variety of modularity and is a better option for various VoIP solutions. With more than six years of experience developing commercial communication solutions using either open-source VoIP communication technology, Vindaloo Softtech is an accomplished VoIP development firm.

#FreeSWITCH vs asterisk #asterisk vs FreeSWITCH #Asterisk Development #Asterisk software development #Asterisk Solutions #FreeSWITCH development services #FreeSWITCH solutions #open source VoIP communication technology

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elisionvoipsolutions · 3 years ago

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Major Services Offered by Elision Technologies

Elision Technologies Pvt. Ltd, more popularly known as Elision, is a leading VoIP company. For more than twelve years, it has been offering different VoIP solutions and services. There are hundreds of businesses all across the globe, which use its different solutions or services. In this article, I will share the details of the top five services the company offers in the VoIP sector.

1. Hosted Call Center Solution

After offering an omnichannel call center solution for many years, Elision has launched a hosted call center solution. It means the company offers call center services based on the SaaS model. It has many amazing offerings for its customers that use the call center services of Elision. Some of the offerings are listed below:

Contact center suite

Unified communication channels

Pay as you go model

Telco connectivity

Unlimited calling

And more

2. VICIDial Customization

Elision has a team of VICIDial experts that offer customization and custom development for VICIDial. VICIDial open source call center solution users can take advantage of any of the below mentioned services from Elision:

Customization of an existing module

Ready to use modules

Custom development

Performance enhancement

And more

3. vTiger Customization

Along with various VoIP services, the company also offers services in one of the most popular open source CRM solutions, called vTiger. The specific mention of vTiger related services in the list of VoIP services offered by Elision is made because there are many services that are related to VoIP solutions. Some of the major offerings of Elision for the vTiger CRM solution are listed below:

Custom module or feature development

VICIDial vTiger integration

Custom call center software integration into vTiger solution

And more

4. Custom Development

As mentioned earlier, for more than 12 years, Elision has been catering to its customers with different VoIP solutions and services. The company has a team of VoIP experts that have been performing development and other similar jobs in major VoIP technologies. Thus, Elision offers VoIP development services in all major VoIP development technologies such as:

Asterisk

FreeSWITCH

WebRTC

OpenSIPs

Elision has a team of VoIP experts that can build any kind of VoIP solution with custom features using any one or more than one VoIP development technology.

5. Hire a VoIP Developer

Along with VoIP development, Elision also has a resource hiring model for its customers. As per the nature of the project or depending on the business needs, customers of Elision can hire one or more VoIP developers from the company. Some of the well-known hiring models offered by Elision are as below:

Hire an Asterisk developer

Hire a FreeSWITCH developer

Hire an OpenSIPs developer

Hire a WebRTC developer

Hire a vTiger developer

Hire a VICIDial developer

Hire a FusionPBX developer

Hire a FreePBX developer

Hire a VoIP project manager

And more

These are the top 5 VoIP services offered by Elision. There are many more services and solutions it offers in the VoIP sector. To know more, visit

https://www.elisiontec.com/

#Hosted Call Center Solution #VICIDial Customization #vTiger Customization

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asteriskservice-blog1 · 6 years ago

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Custom Asterisk Development Solutions in Australia For Businesses and Service Sectors

The Australian economy and GDP show a healthy growth trend with support from government and industries such as shipbuilding, mining, insurance, aviation and IT are growing apace. Communications forms the backbone of any business and the Asterisk platform is the best for small to enterprise grade VoIP communication applications despite the rise of Freeswitch. It is open source and has an active community of developers contributing to its evolution. Asterisk has modules that can be customized and put together to develop tailored communications systems that align with a user’s needs. It can be compared to Lego in which how you put together the bricks determines the shape you achieve. In the same way custom asterisk development by experienced experts makes all the difference to your communications platform.

Custom Asterisk development areas

As an open source platform with a huge library of modules and contributions from an active community, Asterisk offers standard options and customizations in various areas such as system, software, applications and modules in addition to incorporating external features like WebRTC through APIs. For instance, in the case of an application, all modules in a file system will load and slow down operations. Experts in Asterisk development configure and define relevant modules to be loaded in context which immeasurably speeds up operations and reduces load. Channel implementation is another area where extreme customization is possible due to experience and thorough knowledge of the features, desired outcomes and adoption of right channel technologies for super smooth call operations. Similarly, dialplan is extensive in Asterisk and are loosely defined. It needs an expert to implement internal APIs and include scripting for sound performance. Another area where knowledge, expertise and experience come together in custom Asterisk implementation is codec translations and negotiations. Given the variety of codecs in use transcoding plays a vital role in connection quality, clarity of audio and smooth flow. This is just touching the complexities of Asterisk where only custom Asterisk development can deliver solutions that hum along. This is especially evident when it comes to scaling a system involving multiple servers where a scalable communication framework implementation is necessary.

Representative Asterisk customizations

IVR: Virtually every other business implements an Asterisk based IVR but unless it is customized to be dynamic and configurable the IVR does not deliver expected results.

Dialplan programming: Crucial to inbound and outbound calls, dialplan programming needs to be customized to the user environment such as call center or broadcasting.

Call capacity enhancement: Asterisk does have a limitation in the way it can handle concurrent calls. Again, custom Asterisk development can provide workarounds to increase concurrent calls and call per seconds.

Choose a specialist in custom Asterisk development, one that has a thorough understanding of the critical architectural concepts and experience in delivering such solutions. It works out cheaper and better since they will have a lot of reusable modules that can be implemented fast for a tailored solution.

Source: https://voipconferencingsolutions.wordpress.com/2019/06/28/custom-asterisk-development-solutions-in-australia-for-businesses-and-service-sectors/

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danevo-blog1 · 6 years ago

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Importance of EBS Server Pro in the Present Scenario

In today’s era, people prefer using digital means to work at their offices, companies, or stores. They are in search of a platform where they can find a way to get connected with a lot of people from around the globe. Their method of connection can be through a video, audio, text, or any other form. As the latest technology, Electric Board Series Server Pro does it all for the people.

EBS Server Pro is a product that has brought revolution in the market due to its efficiency and the ability to connect with people at long-distance via the Internet.

EBS Server Pro

EBS Server Pro is an appliance which has been designed for the best possible practice of the integrator which is used to integrate computer and telephony. The product has the same application as that of a gateway, along with the maintenance of the range of resources involved in the gateway. It aims at the markets where there is a requirement of:

strong hardware for not allowing any application to disturb the system,

any possibility of loading proprietary application, and

an operating system, if desired

Advantages of an EBS Server Pro

The telephony boards such as PCI or PCI-E buses were developed to meet the requirements of the market for voice solutions. Instead of connecting these telephony boards, the EBS is connected to the Ethernet network, which eliminates the need for servers.

The above-given feature allows the scalability of the product which means the product should continue to function well even when there is an increase in the demand of client flow, without a negative impact on the existing operation.

There is a possibility of creating a redundancy system, which duplicates the functioning of the system and enhances the availability, reliability, and flexibility.

It is compatible with Windows, Linux, FreeBSD, and pfSense.

Features of an EBS Server Pro

As with the EBS family, the EBS Server Pro has well-equipped hardware for

Echo Cancelling:

It is a method used to improve the clarity of voice by removing the echo from the sound. The echo is prevented from traveling the network.

Digital Signal Processing:

Various techniques are used to improve the accuracy and reliability of the signals from the digital communication.

Audio Signal Processing:

It is concerned with the manipulation of audio masking at different times.

In addition, EBS has the capacity of transferring in the digital trunk, its own signaling protocols, and treatment of signaling information. It can be integrated with any open source applications available in the market, such as Asterisk, FreeSWITCH, Snap, and Elastix.

The Process of Operating the Product and Integration

It takes up to 8 modules of the EBS line for the integration with telephony interfaces, making the EBS Server Pro apt for scenarios, where the EBS models are required. The integration is achieved through the channel drivers and supplied along with the equipment.

The EBS Server Pro has an integrated switch, which drops the requirement of other equipment connected to the network. It allows the user to use single compact hardware, which is present with a clean design, and an excellent cost-benefit ratio. This equipment is installed in a standard 19” rack which occupying 1U.

Conclusion

The summary hereby concludes that EBS Server Pro acts like a gateway which establishes a connection between a fixed telephony operator and a VoIP telephony operator and converts Integrated Services Digital Network protocol (ISDN) to Session Initiation Protocol (SIP) and transmits the signal using IP interface without connecting to any of the telephony boards.

For any technical queries, we recommend you to contact a professional assistant who can guide you with the working of the EBS Server Pro and help you with the installation as well.

#EBSserverpro #EBSservermodule #digitalsignalprocessing #audiosignalprocessing #danevo #data #network #voice

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freeswitchbilling-blog · 7 years ago

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Meet ASTPP Community Leaders at GITEX 2018

ASTPP is one of the most popular open source solutions in the VoIP industry. It is a VoIP billing software that is designed and developed for FreeSWITCH solutions. It can be used by any FreeSWITCH based VoIP softswitch, gateway, IP PBX, etc. ASTPP has a large community that keeps on contributing to it to make this FreeSWITCH billing platform better. The ASTPP community is led by two Indian technopreneurs, namely, Mr. Arpit Modi and Mr. Samir Doshi. They have been consistently contributing in this platform since 2013 when they had adopted this platform completely.

To take this open source billing solution to the next level, the leaders of ASTPP community have decided to take some strategic steps and as part of it, they are participating in the largest expo and trade show in Africa, Asia and the Middle East region, namely, GITEX Technology Week 2018. It is the 38th inning of this technology trade show and it is expecting 100,000+ visitors to the expo. The ASTPP community leaders grabbed this opportunity to generate awareness about this most powerful opens source billing solution in front of the masses.

The representatives of this community will be demonstrating the strength of this billing solution to the booth visitors. This billing software has following modules which will be exhibited to the booth visitors:

· Accounts management

· Calling cards

· Billing / Invoicing

· Rate group

· Tariff

· Rates

· Least cost routing

· DIDs Management

· Extensive Range of Reports

The ASTPP also has some add-ons which will be showcased to the interested prospects at the expo. The ASTPP billing solution add-ons are listed below:

· Mobile Dialer

· PC Dialer

· Monitoring Module

· AP Faxer

· TrueCNAM Display

· WHMCS Integration

The ASTPP, open source billing solution is stable and scalable billing software and the leaders of this open source community determined to increase the users and community members for it. As part of this, they are inviting users and community members to come forward in spreading this news and making this participation of ASTPP successful.

The leaders of ASTPP are also willing to meet users and community members in Dubai during the GITEX Technology Week 2018 to get their feedback to improve the platform and community.

The ASTPP community invites all of you to give your valuable presence to the Booth No: B1 – 20, Hall No: 01, Dubai World Trade Center, Dubai during 14 – 18 Oct, 2018. You can also book a personal meeting to discuss possible association opportunity with the community. For more details,

Visit

https://www.astppbilling.org/

#GITEX 2018 #FreeSWITCH billing platform #VoIP softswitch

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cloudtelephonysoluion · 8 years ago

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Now You Can Have the VoIP software development cloud IVR solutions FreeSWITCH development company Of Your Dreams – Cheaper or Faster Than You Ever Imagined

Hello and welcome to my new article about “Now You Can Have the VoIP software development cloud IVR solutions FreeSWITCH development company Of Your Dreams – Cheaper/Faster Than You Ever Imagined.” In this article, I will let you distinguish between these things and recommend you a website where you can get these services cheaper and faster. So, let’s see what is waiting for you.

When you look at VoIP Software Development Cloud IVR solutions FreeSWITCH Development Company online, you will get many results, but all of them are not trustworthy. So, it becomes harder for you to take the right decision and ask yourself which way I should go? I am telling you this is a Company (Gventure Technology) which you can trust and walk further with them. I am going to talk about these topics below

· VoIP software development

· Cloud IVR solutions

· FreeSWITCH development company

VoIP Software Development

Gventure is an entity associated with the advancement and plan of VoIP arrangements in simultaneousness with the requests of the customer venture. They are considered as one among the chief and trusted VoIP solution provider for all the undertaking correspondence needs. They endeavor to put the best foot forward and guarantee to give VoIP business solutions for the customer with the most surprising request of precision. Their relentless attempt toward this path has empowered the productive conveyance to provide demanding outcomes that work to profit the client.

As one among the esteemed VoIP organizations, their specialty lies in the way that we make utilization of open source VoIP platforms, for example,

1. FreeSWITCH,

2. Asterisk install,

3. Asterisk IVR,

4. Opensips, and

5. Kamailio

To address the different VoIP requirements.

For VoIP software development Cloud IVR solutions, Cloud Telephony Solution go to Gventure Technology.

Cloud IVR solutions

Cloud IVR gives you the opportunity to automate and control your most essential client touch point or business forms via telephone without managing complex communication framework. Gventure facilitated IVR solution and advancement devices control a scope of voice applications, which enhance the client encounter.

Gventure has an experience of creating cloud hosting architectures, which are high performance and highly trustworthy. Cloud deployment makes it simple to deliver a customer experience, which can retort quickly to changing client demands while controlling your overall costs, making it easier to retain customers and stay modest in today’s business climate.

Their cloud-based deployment options allow your business to leverage latest solutions, without bearing the burden of implement comprehensive cloud-based interaction management, significant upfront capital or additional IT investments, and workforce optimization technologies including inbound, outbound, and blended voice communications. Enjoy peace of mind with total severance and no single point of failure – backed by their world-class uptime service-level contract.

For VoIP software development Cloud IVR solutions FreeSWITCH development company go to Gventure Technology Pvt. Ltd.

FreeSWITCH Development Company

FreeSWITCH is an extensible exposed source cross-platform telephony platform designed to route and interconnects open communication protocols using text, audio, video or any other form of media. It can be utilized as a soft-client, carrier-class Softswitch or even as PBX. It is the comfort of installation and configuration has made it a very user-friendly PBX solution nowadays. It has a standard design which means that new features can easily integrate into the system as additional modules. Unwanted modules can be disabled at the same time.

FreeSWITCH Softswitch can be installed and work with no trouble in any possible framework stage including Windows, has made it a desirable alternative to the VoIP PBX engineers. In spite of the fact that, FreeSWITCH can be managed through a GUI, the structure of its setup indexes and documents influences the next record to get to the administration more easy to use and straightforward to deal with notwithstanding for the beginner. Setup documents are XML-based. The XML composition is apparent and can be effectively caught on. No XML ability is required.

For VoIP software development Cloud IVR solutions FreeSWITCH Development Company goes to Gventure Technology Pvt. Ltd.

Conclusion

Thank you so much for reading this article. For any VoIP software development Cloud IVR solutions FreeSWITCH development company go to Gventure. They will always be at your service. Don’t put out of your mind to share your opinion about the article in the comment section.

#FreeSWITCH development company #VoIP software development #cloud telephony solution #cloud IVR solutions

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loveperfectcreationcollecto-blog · 8 years ago

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Kingasterisk technology provides Best Voip Solutions

Asterisk Based Voip Solutions

We are very innovative with opensource voip technology based product, we are heartly invite you to contact us and get more details about our products and marvelous applications.

We are working on voip based opensource plateform since 8 years, we are providing our solutions, services and supports on several applications like asterisk, freepbx, a2billing, dialer, freeswitch, opensips ,kamailio, callweaver, hylafax, elastix, EPABX, IVR, Predictive dialers, Voice Mail, Voice Logger, Video & audio conferencing solutions, web conferencing solutions and lots more.

We are KingAsterisk Technologies where we are developing lots of voip based solutions and applications. Application development, research and issues resolution supports are like our blood in vains, we are keep supporting to our client to achieve their goal in their own decided plateform and models.

IPPBX Solution

IP PBX stands for Internet Protocol Private Branch Exchange. IP PBX is a telephone system that is aimed at delivery of the information including voice, video, etc over the network. IP PBX has become a real breakthrough in the modern technologies as it allows transferring various types of data. IP PBX is especially useful for business enterprises that need to maintain constant contact with customer and affiliates that may be far away. IP PBX is the way to monitor your business throughout the world.

It's the ability to make free calls that makes IP PBX so popular today. International phone calls are becoming much cheaper nowadays but the considerable part of expenses that business companies have goes to cover the cost of international and long distance calls. IP PBX offers a cheap telecommunication service that lets you stay connected with people on the other part of the world. Since IP PBX technologies were introduced hundreds of companies have managed to cut down their expenses and have become more profitable.

VOIP Call recording

Call recording

is a feature present in both client and server side software’s provided by KingAsterisk Technologies. This is a use ful feature especially when there are law obligations or for quality control. On the server each call can be recorded (selectable by route/user) regardless of the codec used (all common codec's are supported). The recording is done using separate low priority background thread which doesn’t affect the call quality in any way. These files are usually stored on a separate hard disk to not affect the I/O speed on the primary disk where the VoIP server is installed

The recorded voices are stored in compressed and encrypted format allowing for easy later playback or export by the administrators from an easy to use interface. To enable voice recording for a user, just open the "users and devices" form from the KingAsterisk Manage, select the user(s), switch the "Functions" tab and tick the "Recording" checkbox. You can listen to recorded conversation anytime later from the "CDR records" form using the by selecting the "Recorded Conversations" radio box.

Click to Call

Click-to-call

is a service which lets users click a button and immediately speak with a customer service representative or interconnect two or more telephone "line". The call can either be carried over VoIP, or the customer may request an immediate call back by entering their phone number. One significant benefit to

click-to-call

providers is that it allows companies to monitor when online visitors change from the website to a phone sales channel. Click To Call is the solution for all site owners that like to offer a free phone call to their visitors.

The advantage over other proprietary solutions is that these are true VoIP calls that will work with any phones (voip terminals, softphones, your mobile or landline number) through any telecom service provider. For example you can receive the incoming calls on your VoIP phone or softphone (free calls) and if you are away, then the call can be automatically forwarded to your mobile number.

Sound Box Dialer

offers a call center web based application which can help you to increase your productivity, It will give portabilities to your agents and people who can perform smarter and faster way in your teams.

We have experience in dial applications, so we can keep thinking a lot on this field too. It will give your clients more benefits and more knowledge about your performance so you can get more and more business. This is will give you more effective sound systems where your team can perform better, it will give best and accurate results for your performance.

It will allow to transfer calls, manual way as well as faster way as well. You can also monitor all the stuff on live systems too while agents are in actions. You can use this application in various ways and run in various purposes.

CRM For Call Center

Vtiger CRM is a fully open source CRM application. vtiger CRM is widely trusted by thousands of businesses to effectively manage leads, identify quality sales, track marketing campaigns and monitor inventory. Its features include..

Customer support & service functions, including a customer self-service portal

Marketing automation (lead generation, campaign support, knowledge bases)

Inventory Management / Analysis and reporting

We handle VTiger projects and customize the CRM for various industry-specific needs. Our customer-centric approach makes us proficient in VTiger CRM implementation for various industry verticals. We have expertise in integrating CRM & Telephony.

Voice Broadcast

The KingAsterisk Technologies provides an automated mass dialling solution where hundreds, or thousands of numbers are simultaneously dialled, and an automated connection to an IVR occurs when answered by a person.

For more complex solutions, where the IVR needs to transfer calls to a Live Agent, rather than dialling a fixed number of outbound lines the system can be told how many Live Agents are available and it decides how many numbers to dial based on how many Live Agents are busy. There is a further option for these Agents to log into the predictivedialler.net system, which means that the system knows exactly how many agents are available and whether they are ready to take calls. Using this latter mode there is no need to manually adjust the dialling rate on the Broadcast dialler at all.

Custom IVR Service

Interactive voice response(IVR) is a computerized phone system that enables a person, typically a telephone caller, to make a selection from a voice menu. The selection is made using phone keypad entries or voice responses. This interaction allows the individual to communicate with the phone system and thus the computer system. The phone system plays pre-recorded voice prompts and the person typically presses a number on a telephone keypad to select the option associated with the voice prompt

The KingAsterisk Technologies VoIP server has a built in sophisticated IVR module capable of handling all your business needs including callback and forwarding options, phone to phone calls, answering for SMS initiated actions, announcements, etc. The IVR module is associated with campaigns, which can be set to run different scripts (functions).

Billing Solution

The KingAsterisk Technologies VoIP server built-in billing was designed with carrier grade customers in mind. We offer a complete suite of billing and switching solutions that support the whole range of common VoIP business models. Our VoIP Billing platform will allow you to sell advanced features and its configuration is handled by a simple to use user interface. The softswitch allows service providers to efficiently manage and accurately bill all aspects of their end-users’ VoIP usage.

The KingAsterisk Technologies VoIP billing software is built using 100% multi-threaded C++ code, making it a high performance billing engine that can handle millions of calls. The billing process is running on lower priority threads, never affecting the call quality when the server usage is high. The remote management application offers multiple customizable reports including, accounting, revenue, expenses, call history by user and others.

SMS Broadcasting

SMS Broadcaster is a software which allows you to send/broadcast SMS messages to a list of phone numbers. You can type in the SMS Message and SMS Broadcaster will read in a list of phone number from a file on the root of your SD card and broadcast you’re message to everyone in the file.

This is a lot quicker than do it one by one. Especially, it is very useful when you want to send group SMS to your hundreds or thousands friends or clients. It is an effective tool to improve your efficiency and save your time. You put a file named numbers.txt with your numbers on the root of your SD card using the USB cable, Bluetooth, email or a File Manger then you type in your message and click two buttons and SMS Broadcaster will do everything else.

Video Conferencing

Conducting a conference between two or more participants at different sites by using computer networks to transmit audio and video data. For example, a point-to-point (two-person) video conferencing system works much like a video telephone. Each participant has a video camera, microphone, and speakers mounted on his or her computer. As the two participants speak to one another, their voices are carried over the network and delivered to the other's speakers, and whatever images appear in front of the video camera appear in a window on the other participant's monitor.

The KingAsterisk videoconferencing allows three or more participants to sit in a virtual conference room and communicate as if they were sitting right next to each other. Until the mid 90s, the hardware costs made videoconferencing prohibitively expensive for most organizations, but that situation is changing rapidly. Many analysts believe that videoconferencing will be one of the fastest-growing segments of the computer industry in the latter half of the decade.

Fax Solutions

Fax For Asterisk provides two components: res_fax and res_fax_digium. Res_fax is an Asterisk resource module that adds fax termination and origination functionality in Asterisk. It provides Asterisk dialplan functions and dialplan applications to enable the user to build highly-customizable fax solutions. Res_fax_digium provides core fax processing functionality in the form of several supported fax modems — V.21, V.27ter, V.29, and V.17 — to achieve speeds up to 14400bps.

Fax For Asterisk provides the functionality to send and receive faxes to / from TDM and IP channels — TDM channels are established across Digium telephony boards and IP channels can use regular G.711 audio encoding or T.38 encapsulation.

Faxes transmitted and received by Fax For Asterisk begin and end as TIFF image files. TIFF files may be readily converted into or from other formats using standard Linux command-line utilities.

Call Center

The KingAsterisk Technologies Callcenter can handle huge amount of inbound and outbound traffic, in a secure, reliable way. The KingAsterisk callcenter combines maximum efficiency with easy to use and intuitive interfaces. Separate campaigns can be setup each of them running separately assigned scripts with graphical user interface for both the operators and the supervisors. By defining quotas, you can restrict your calls to well defined target groups (called clients).

All the call-center related statistics can be viewed in real-time. One of the features of the callcenter is predictive dialing. To restrict the operator wait times, the Calls can be prepared on the server side and dropped to operators when they are waiting for it.

Tele Marketing

We provide you the correct telemarketing services which you need for your business in manner of successful inbound and outbound telemarketing campaign. We can provide you the best applications and services to conduct tele marketing businesses

Our experienced telemarketing program managers design, manage and implement highly customized and flexible customer depended telemarketing campaign that meet your business objectives. Leading organizations and startups rely on us for their inbound telemarketing and outbound telemarketing campaign.

You can create any kind of IVR scripts using a graphical user interface from the remote management client. With more than 30 built-in actions it is very easy to build your custom IVR menus within minutes.

Multi tenants

Multi-tenant management is the ability for a cloud tenant to have omnipotence over the instances, data, and networks in their cloud-hosted solution. In terms of an SP VDI solution this means the vDesktops, the master images, the application distribution mechanism (if applicable), patching, user data, vDesktop networks, access policies, pool size, et cetera. Essentially, the tenant’s management portal needs the ability to perform the primary tasks performed in the VMware View Admin Console or XenDesktop Desktop Delivery Controller console.

In addition, the multi-tenant management solution needs to have the ability to securely provide this level of access to multiple tenants. Unfortunately, this is the first hurdle the major players, Citrix XenDesktop, VMware View, and Microsoft VDI trip over. These solutions have one primary console that’s used to manage the entire environment.

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