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sheerbittech · 17 days ago

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Sheerbit: The Top VoIP Development Company for Custom, Scalable Solutions

Introduction

In today’s fast-paced digital landscape, clear and reliable communication is no longer a luxury—it’s a business imperative. Voice over Internet Protocol (VoIP) technology has revolutionized how organizations connect, collaborate, and serve their customers. However, not all VoIP development companies are created equal. Selecting the right partner can mean the difference between a smooth deployment and ongoing technical headaches. This is where Sheerbit shines. As a leading VoIP development company, Sheerbit combines deep technical expertise, bespoke solutions, and unwavering customer support to deliver communication platforms that scale with your business.

Understanding VoIP and Its Business Impact

VoIP enables voice calls, video conferences, and multimedia data to traverse IP networks rather than traditional telephone lines. This shift reduces costs, boosts flexibility, and integrates seamlessly with cloud-based and on-premise systems. Organizations that adopt VoIP enjoy features such as advanced call routing, click-to-dial, call analytics, and integration with CRM or helpdesk platforms—empowering teams to work smarter and respond faster to customer needs.

Common Challenges in VoIP Deployments

Even with compelling benefits, VoIP projects can falter if not handled by seasoned professionals. Organizations often face:

Quality of Service (QoS) issues that lead to dropped calls or latency

Security vulnerabilities exposing voice traffic to eavesdropping or fraud

Complex integrations with legacy PBX systems or third-party applications

Scalability hurdles when call volume spikes or new offices come online

Ongoing maintenance and lackluster support after go-live

Addressing these challenges demands a partner who understands both the networking fundamentals and the unique needs of your business.

Why Sheerbit Stands Out

Sheerbit has built its reputation as the best VoIP development company by focusing on three core pillars: technical excellence, client-centric customization, and comprehensive support.

1. Technical Excellence

Every Sheerbit engineer brings extensive experience with leading VoIP platforms—Asterisk, FreeSWITCH, OpenSIPS, Kamailio, and WebRTC frameworks. Whether you need a robust SIP trunking solution or a cutting-edge WebRTC application, Sheerbit’s team writes clean, scalable code and adheres to industry best practices for network performance and reliability.

2. Custom VoIP Solutions

Off-the-shelf VoIP packages rarely fit every business scenario. Sheerbit specializes in tailor-made development services, from crafting custom dial plans and interactive voice response (IVR) systems to integrating advanced call-center features like predictive routing and real-time analytics. With Sheerbit, you can hire VoIP developers dedicated to understanding your workflows and delivering solutions that align perfectly with your objectives.

3. End-to-End Support

The deployment of a VoIP system is just the beginning. Sheerbit offers full-lifecycle services: consulting and needs assessment, architecture design, development, testing, deployment, and post-launch maintenance. Their DevOps-driven processes ensure seamless updates, continuous monitoring, and rapid resolution of any issues—minimizing downtime and safeguarding call quality.

Key Service Offerings

VoIP Development Services: Sheerbit engineers build feature-rich VoIP applications, including softphones, mobile VoIP apps, and web-based conferencing tools. They ensure interoperability across devices and browsers, delivering user experiences that mirror or exceed traditional phone systems.

Custom Integrations: Leverage your existing investments by integrating VoIP with CRMs like Salesforce or HubSpot, helpdesk platforms such as Zendesk, or bespoke databases. Sheerbit’s APIs and middleware ensure call data syncs accurately with your business systems.

SIP Trunking & PBX Migration: Whether you’re migrating from a legacy PBX to a modern SIP-based infrastructure or establishing new SIP trunks for international call routing, Sheerbit’s proven migration framework guarantees minimal service interruption.

Security & Compliance: Voice services must be secure. Sheerbit implements TLS/SRTP encryption, robust firewall configurations, and fraud-detection modules. They also assist with regulatory compliance (e.g., GDPR, HIPAA) to protect sensitive communications.

Success Stories

Global Retail Chain Enhances Customer Support A multinational retailer struggling with call center overload engaged Sheerbit to deploy a scalable Asterisk-based IVR with predictive call routing. Post-launch, average wait times dropped by 40%, and customer satisfaction scores rose significantly.

Healthcare Provider Integrates VoIP with EHR Sheerbit developed a HIPAA-compliant FreeSWITCH solution for a healthcare network, integrating audible call prompts directly into the electronic health record system. Clinicians saved an average of 10 minutes per patient, boosting operational efficiency.

How to Hire Sheerbit’s VoIP Developers

Engaging with Sheerbit is straightforward. After an initial consultation to assess your needs, you’ll receive a detailed proposal outlining scope, timelines, and pricing. You can choose to hire VoIP developers on a project basis or onboard them as part of your extended team. Flexible engagement models include fixed-price projects, time-and-materials contracts, or dedicated-team arrangements.

Pricing & Engagement Models

Sheerbit offers transparent, competitive pricing tailored to project complexity and resource requirements. Typical engagement tiers include:

Standard Package: Core VoIP deployment with essential features

Advanced Package: Custom development, integrations, and analytics

Enterprise Package: Full-scale solutions with ongoing support and SLAs

The Implementation Process

Discovery & Planning: Define objectives, technical requirements, and success metrics.

Design & Architecture: Create network diagrams, call-flow maps, and infrastructure plans.

Development & Testing: Build features in agile sprints, perform comprehensive QA, and conduct pilot testing.

Deployment & Training: Roll out the solution, configure networks, and train your IT staff and end users.

Support & Optimization: Provide 24/7 monitoring, periodic performance reviews, and iterative enhancements.

Conclusion & Call to Action

Selecting the best VoIP development company can transform your organization’s communications, delivering cost savings, operational agility, and superior customer experiences. With Sheerbit’s proven expertise in custom VoIP solutions, end-to-end support, and dedication to quality, your business is poised for seamless, future-ready communications.

Ready to elevate your voice infrastructure? Contact Sheerbit today to schedule a free consultation and discover how you can harness the power of a tailored VoIP solution built by industry experts.

#voip #webrtc #voip development #hire voip developers #voip development company

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bytebranllp · 11 months ago

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Selecting the Ideal Tech Stack for Your VoIP Project

Software development involves various technologies and processes that are constantly evolving with new tools and frameworks. While these advancements simplify development, they also add complexity when selecting the right technologies for a project. Beyond the technical aspects, successful software development requires strong teamwork, project management and communication among diverse teams. These skills are crucial for keeping projects on track and within budget. Our blog article aims to give you a clear understanding of the crucial factors to consider when choosing the right technology stack for your software development project.

Understanding the tech stack

A tech stack is a set of tools and technologies used to build and run an application to cover everything from servers and databases to frontend and backend frameworks. Here’s a quick look at its main components:

Frontend: Handles the user interface and experience with technologies like HTML, CSS, JavaScript, and frameworks like React, Angular and Vue.js.

Backend: Manages business logic, database interactions, and server setup using languages like Python or Ruby and backend frameworks.

Databases: Store and manage application data, including relational databases like MySQL and NoSQL databases like MongoDB.

DevOps and Cloud Services: Ensure your application is integrated, scalable, and well-maintained.

VoIP Platforms

FreeSWITCH, Kamailio, OpenSIPS, and Asterisk are top platforms for creating VoIP services.

FreeSWITCH is an open-source platform that connects and routes communication protocols.

Kamailio and OpenSIPS are high-performance SIP servers for managing voice, video, and real-time communication.

Asterisk is a free framework that turns a standard computer into a powerful communication server.

Tech Stacks in Practice

Have you ever wondered what technology stacks power your favorite brands? Let's explore the tech stacks some industry leaders use and see what they rely on to deliver their services.

Uber Tech Stack:

Web Servers: NGINX, Apache

Databases: MySQL, PostgreSQL, MongoDB

Server-side Framework: Node.js

Programming Languages: Python, Java, JavaScript, Objective-C

Uber's tech stack combines robust web servers with versatile programming languages and frameworks, enabling smooth communication and data management across its platform.

Instacart Tech Stack:

Server: NGINX

Databases: PostgreSQL, Redis

Server-side Framework: Rails

Programming Languages: Ruby, Python, Objective-C

Instacart uses a streamlined stack focused on efficiency and reliability. It strongly emphasizes fast database operations and a robust server-side framework.

Reddit Tech Stack:

Server: NGINX

Databases: PostgreSQL, Redis

Server-side Framework: Node.js

Programming Languages: JavaScript, Python

Reddit's stack is optimized for handling large amounts of user-generated content. It uses powerful databases and a modern server-side framework to maintain its performance and scalability.

These examples highlight brands' different choices in building their tech stacks tailored to their specific needs and operational demands.

Choosing the Right Tech Stack for VOIP App Development

Selecting the right tech stack for app development can be overwhelming, especially for non-technical entrepreneurs. While developers often have their preferred stacks, it's wise to seek guidance from a subject-matter expert. An expert can provide valuable insights and help you avoid future issues. If you're short on resources, start with these simple steps:

Understand Your Users' Needs: Focus on what your users need. For example, if most of your users are on mobile devices, consider a "mobile-first" tech stack. If your app is "mobile-only," your tech stack will likely differ from other types of applications.

Define Your Development Needs: The complexity and scale of your app will influence your tech stack choice:

Small Projects: Well-defined stacks like Python-Django or Node.js-React are ideal for simple apps like MVPs or one-page apps.

Medium Projects: E-commerce stores or mid-market apps require more complex stacks with multiple layers of programming languages and frameworks.

Large Projects: Complex marketplaces or social apps need a robust tech stack designed to handle high-volume use and maintain performance.

Evaluate Your Resources: When choosing a tech stack, consider your available resources. Even the best tech stack won't be effective without the right developers to implement it. Many frameworks and tools are open-source and free to use, but remember to account for costs related to servers and hosting when making your decision.

Essential Components for a VoIP Software Technology Stack

Let’s understand these components.

Real-Time Communication Protocols: SIP (Session Initiation Protocol) and RTP (Real-Time Transport Protocol) are essential for managing voice, video, and messaging sessions. WebRTC enables real-time communication directly in browsers and mobile apps, making development easier.

Codecs: Codecs like G.711, G.722, G.729, and Opus compress and decompress voice data, balancing call quality and bandwidth usage.

Programming Languages: Python and Java are commonly used for their simplicity, readability, and strong support for multithreading and concurrency.

Web and Mobile Frameworks: Angular and Node.js are popular for web development, while Flutter and React Native are favored for cross-platform mobile apps.

Databases: SQL databases like MySQL and PostgreSQL handle structured data, while NoSQL databases like MongoDB offer scalability for unstructured data.

Server Technologies: Cloud services like AWS, Google Cloud, and Microsoft Azure provide scalable, secure environments with various development tools.

Testing Tools: Selenium for UI testing and Apache JMeter for load testing help ensure the software performs well and offers a good user experience.

Take Away

A tech stack is essential for a startup or business's success, so selecting the right technologies, languages, frameworks, and tools is essential.

Choosing the right technology stack for VoIP software development depends on your project's specific needs and goals. To make the best choice, follow these three rules: prototype and test, seek expert advice both within and outside your team and stay flexible. Doing so will help you build a strong and scalable product. Consulting experienced developers or software development companies can help you get tailored recommendations. With the right choice, your VoIP application will meet your current needs and be adaptable to future changes.

#voip technology #voip solutions #voip development #VOIP technology stack #Custom Software Development

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huntterri · 3 years ago

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Openser xlog

Openser xlog code#

loadmodule textops.so loadmodule maxfwd.so loadmodule xlog.so. loadmodule textops.so loadmodule siputils.so loadmodule xlog.so. KAMAILIO This config file implements the basic P-CSCF functionality - web. # opensips-cli -x mi subscribers_list E_RTPPROXY_STATUS unix:/tmp/event. KAMAILIO define WITHMYSQL define WITHAUTH define WITHUSRLOCDB define. # opensips-cli -x mi subscribers_list E_RTPPROXY_STATUS If the socket is also specified, only one subscriber information is returned. By setting the usrloc’s parameter dbmode to 2 we tell OpenSER to use mysql for storing contact information (and not the memory). We need mysql module to store user locations in a database. We xlog() function for logging the processing details on the screen. If the event is specified, only the external applications subscribed for that event are returned. We run the OpenSER server in a debug mode as a terminal process. The goal of the implementation is to load balance requests from my SIP provider to a farm of 10 asterisk servers for media processing. Output: If no parameter is specified, then the command returns information about all events and their subscribers. hey, i am new to OpenSIPS/OpenSER and just finished writing my first config. In Kamailio, we often may wish to add headers, view the contents of headers and perform an action or re-write headers (Disclaimer about not rewriting Vias as that goes beyond the purview of a SIP. socket (optional) - external application socket The SIP RFC allows for multiple SIP headers to have the same name, For example, it’s very common to have lots of Via headers present in a request.pid (optional) - Unix pid (validated by OpenSIPS).Section 1.2, Implemented Specifiers shows what can be printed out. A C-style printf specifier is replaced with a part of the SIP request or other variables from system. level (optional) - logging level (-3.4) (see meaning of the values) This module provides the possibility to print user formatted log or debug messages from OpenSIPS scripts, similar to printf function.If pid is also given, the logging level will change only for that process. OpenSIPS is a GPL implementation of a multi-functionality SIP Server that targets to deliver a high-level technical solution (performance, security and. If a logging level is given, it will be set for each process. If no argument is passed to the log_level command, it will print a table with the current logging levels of all processes. Get or set the logging level of one or all OpenSIPS processes. Output: an array with one object per connection with the following attributes : ID, type, state, source, destination, lifetime, alias port. Updated to latest upstream version: 1.0.1 Added support for multiple modules, including accounting, mysql, sms, xlog. Pseudo-variable marker - represents the character '' 3.2. The list of pseudo-variables in OpenSER Predefined pseudo-variables are listed in alphabetical order.

Openser xlog code#

As a special service "Fossies" has tried to format the requested source page into HTML format using (guessed) INI source code syntax highlighting (style: standard) with prefixed line numbers.Īlternatively you can here view or download the uninterpreted source code file.The command lists all ongoing TCP/TLS connection from OpenSIPS. Pseudo-variables can be used with following modules of OpenSER: avpops - function avpprintf () xlog - functions xlog () and xdbg () 3.

#Openser xlog

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vindaloo-softtech · 3 years ago

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Outstanding IT Services by Vindaloo Softtech Outdo Its Competitors: GoodFirms

Vindaloo Softtech Pvt. Ltd. is a foremost IT company that offers exceptional tech solutions. Vindaloo provides VoIP software development, MEAN stack development, database services, digital marketing services, e-commerce development, custom CRM development, web app development, cross-platform, and UI/UX design services. The company was founded in 2016 with headquarters in Ahmedabad, Gujarat, India.

Vindaloo focuses on cutting-edge technologies with an innovative approach to deliver excellent results. The dedicated team is highly experienced in the IT and telecommunication industries. They guarantee robust & reliable solutions for every size of business and a reasonable budget. The company extended its services and served its clients worldwide.

The company was established to serve VoIP app development solutions. In 2016, Vindaloo started to offer its software development services using open source VoIP technologies, FreeSWITCH, OpenSIPs, Kamailio, Fusion PBX, etc. Gradually, they increase their services by adding outstanding web apps, mobile apps, and cross-platform development services. They offer full-service customized development solutions according to their client’s business requirements.

They comprehended the importance of UI/UX and concluded that an attractive design is a must for an app. Further, Vindaloo is determined to provide unmatched UI/UX design in the IT sector. The company possesses expert designers & developers to satisfy their global clients to succeed in it. Today, the company has delivered numerous projects and shares a solid bond with its clients. The team believes in maintaining the relationship with each client that drives massive success. It can also be an opportunity to help them again through Vindaloo’s services & solutions.

The company’s facts & figures:

5+ Years Experience

10K+ Hours Experience

100+ Quality Project Delivered

40+ Experts

GoodFirms is a B2B organization that connects IT service seekers with service provider firms. The platform helps service seekers collaborate with the most appropriate partner through exceptional research on IT firms.

Moreover, the company’s team of experienced researchers and reviewers seeks client satisfaction, market penetration, the overall experience in the market, and quality of deliverables. GoodFirms studied all the registered companies based on the three essential parameters: Quality, Reliability, and Ability.

Similarly, GoodFirms also evaluated the services of Vindaloo Softtech. According to researchers and reviewers, the firm proves to be promising in providing distinctive IT outsourcing solutions.

Vindaloo is a trusted company that offers affordable prices yet zero compromises on quality. The company’s mission is to supply suitable services and develop reliable products for its clients. The team has a tremendous vision for the company to deliver a hassle-free experience to their clients during the product development process.

Vindaloo’s VoIP services include consultancy, troubleshooting, installing, configuring, and developing different open-source VoIP software. The company’s VoIP technology service helps in client service communication. Most startups and medium to large enterprises implement VoIP technology and provide improved ROI by cost control & time. The size is not an issue for VoIP technology to produce excellent customized software development solutions.

Why Choose Vindaloo Softtech?

Top-Grade tech expertise

Budget-friendly

Excellent project management service

Flexibility & Reliability

Absolute confidentiality with a non-disclosure agreement

The company’s core solutions are Hosted PBX, IP PBX, FreeSWITCH solutions, OpenSIPs Development, Kamailio Development, WebRTC Development, IVR, Call Center Software, VoIP Billing, Click to Call, Softswitch, A2Billing, and Mobile VoIP Application, etc.

This expertise has helped Vindaloo Softtech attain a leading position among Ahmedabad’s top IT services companies at GoodFirms.

About the Author

Working as a Content Writer at GoodFirms, Anna Stark bridges the gap between service seekers and service providers. Anna’s dominant role is to figure out company achievements and critical attributes and put them into words. She strongly believes in the charm of words and leveraging new approaches that work, including new concepts that enhance the firm’s identity.

#VoIP app development

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ob4by · 4 years ago

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ubuntu 18.04 安装opensips 3.1

OpenSIPS is an Open Source SIP proxy/server for voice, video, IM, presence and any other SIP extensions.OpenSIPS is a multi-functional, multi-purpose signaling SIP server used by carriers, telecoms or ITSPs for solutions like Class4/5 Residential Platforms, Trunking / Wholesale, Enterprise / Virtual PBX Solutions, Session Border Controllers, Application Servers, Front-End Load Balancers, IMS…

View On WordPress

#opensips #ubuntu #viop

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kingasterisk · 4 years ago

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Kingasterisk Technologies – Asterisk-VoIP Solution Provider

Kingasterisk Technologies is a company which is working on VoIP based open-source platform for ten years.

We are furnishing our solutions, services, and supports on various applications like an asterisk, FreePBX, 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.

Kingasterisk Technologies is an expert in developing several applications and solving issues of clients for making their business-wide. We continue to support our clients to achieve their goals and sharing good positive power.

Kingasterisk Technologies is a highly recommended company that consists of skilled and experienced developers. Moreover, Asterisk experts always have been successful in fixing all asterisk solutions and deliver optimum results.

Kingasterisk Technologies are giving the best services and supports to satisfy our clients. Therefore the client can interact more with us to increase their business. Our company has conserved its appellation in the business field by supplying trust-worthy on-time project delivery.

About Kingasterisk:

Kingasterisk provides the best Asterisk VoIP solutions. VoIP is Voice over Internet Protocol which allows you to make calls using a broadband internet connection over the world instead of using phone calls. To utilize VoIP, all the users need a high-speed internet connection and a VoIP phone service provider. Kingasterisk is the leading supplier and pioneer of VoIP solutions. The characteristic of Kingasterisk is beneficial client relation that emerges telephony companies derived from a large corporation. Our company grant 24*7 supports system and cost-effective services that are affordable for the client. We serve career-class telecommunication services with powerful communication technology.

For more information, please visit https://www.kingasterisk.com/

Media Contact:

Kingasterisk Technologies

Email: [email protected]

Telephone: +1(786)414-2610

#asterisk solutions #asterisk services #asterisk development #asterisk customization #asterisk module development #asterisk application development #asterisk customization services #asterisk and voip solution

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jennabrileyus · 7 years ago

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ICT Innovations

January 19. 2018 ICT Innovations released new version of communications development platform for web developers ICTCore Version 0.8.0 December 3 2017. New blog post at ICT Innovations about top open source CRM of year 2017 November 12 2017. New blog post at ICT Innovations about top open source Voip and Telecom applications of year 2017 ICT Innovations is a software development company having experienced and dedicated professionals with expertise in LAMP Stack and computer telephony integration (CTI). ICT Innovations has strong knowledge of Open Source communications technologies as well as in depth expereince of working with open source ICT applications such as Asterisk, Freeswitch, Opensips and RabbitMQ.

Products

ICTBroadcast

ICTBroadcast is a multi-tenant unified communications telemarketing autodialer software solution. It supports Voice broadcasting, SMS broadcasting, Email broadcasting & Fax broadcasting. It is suitable for small business owners, enterprises and Internet telephony service providers. ICTBroadcast is a smart auto dialer software with advanced autodialing multilingual supported features. ICTBroadcast is being offered in two editions , ICTBroadcast Enterprise Edition suitable for organizations / enterprises for their own telemarketing and mass communication needs and ICTBroadcast Service Provider Edition suitable for Internet telephony services providers (ITSP) / carriers to offer hosted telemarketing / mass communications services to their customers and integrate it with their existing infrastructure.

Features & Campaigns supported

Polling and automated Surveys through Voice Broadcasting

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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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gventuretech · 1 year ago

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OpenSIP development plays a transformative role in shaping the future of VoIP technology. Its scalability, flexibility, and interoperability make it an indispensable tool for building a wide range of communication solutions, from enterprise PBX systems to WebRTC applications and IoT integrations. As the demand for seamless and reliable communication continues to grow, OpenSIP development will remain at the forefront of innovation, driving the evolution of VoIP technology for years to come.

#opensips #opensip development service #opensips development #opensips services #opensip solutions #opensips development solution

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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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gventuretech · 1 year ago

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With OpenSIP doesn't have to be daunting, even for beginners. By understanding the fundamentals of SIP and VoIP, setting up OpenSIP, and leveraging its features and functionalities, you can create scalable and feature-rich communication applications tailored to your specific needs. Whether you're building a messaging app, a voice calling platform, or a video conferencing solution, OpenSIP provides the tools and resources you need to bring your ideas to life. So why wait? Start building your next communication app with OpenSIP today!

#opensip #opensips services #opensips development

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jennabrileyus · 7 years ago

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VOIP Event Calendar

May 2018

1-4: AMSTERDAM, OpenSIPS Summit - join fellow VoIP & RTC experts, developers and users from all over the world for 3+1 days of talks, inspiring presentations, workshops and trainings about OpenSIPS and the Open-Source ecosystem (FreeSWITCH, Asterisk, Homer, Janus and many more)

https://www.opensips.org/events/Summit-2018Amsterdam

October 2017

Oct 10:European VoIP Summit - Amsterdam 2017 Exploring the Future of Voice Communications

The day will consist in a series of panel sessions, keynotes and presentations. We find that this approach, mixed with ample time to network with your peers and colleagues in the industry breeds significant discussion and debate throughout the day. http://www.cavellgroup.com/index.php/events/evs-amsterdam-2017

September 2017

Sep 12-15: Asian Carriers Conference, ACC 2017, the largest telecommunication conference of Asia will take place at Cebu, Philippines at the Shangri-La’s Mactan Resort & Spa to bring together the present leaders of the telecommunication industry from all over the world.

Meet Inaani Pte Ltd along with over 1500 telecommunication professionals from about 600 companies spread across over 200 countries.

August 2017

7-10: CHICHAGO ClueCon 2017 A 4-Day conference full of demos and presentations from industry leaders. ClueCon features a full-day hack-a-thon devoted to IoT, Making, Coding Telephony and media applications.

http://cluecon.com

11: FreeSWITCH and OpenSIPS Training After the ClueCon conference there will be a training session available with the choice of FreeSWITCH or OpenSIPS training

http://cluecon.com/training.html

May 2017

2-5: AMSTERDAM Opensips three exciting days filled with VoIP and RTC presentations, workshops and design clinics bringing the latest updates from the OpenSIPS community

https://www.opensips.org/events/Summit-2017Amsterdam.html

8-10: BERLIN - Kamailio World Conference & Exhibition - real time communications event with sessions covering SIP, VoIP, VoLTE/IMS and WebRTC - https://www.kamailioworld.com

14-16: CHICAGO Come and meet Speedflow at International Telecoms Week, Booth#1006, Purple level, East tower.

17-18: LONDON Discover the solutions designed to provide flexible ways of delivering, managing and supporting communication that include major UC&C technologies: Cloud, Mobile, Customer, Video, Networks, Collaboration.

http://www.ucexpo.co.uk

24, 31*: LIVE STREAM JeraSoft, the innovative developer of carrier-grade billing platforms and VoIP solutions, announced today the release of a new version of their versatile billing platform – JeraSoft VCS 3.12. Be the first to hear about JeraSoft’s latest technology innovations. ... from Updates & News http://www.voip-info.org/wiki/view/VOIP+Event+Calendar

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