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crmvoipsoftwaresolutions · 3 years ago

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Omnichannel Contact Center Software for E-commerce Industry

Ecommerce does need to have a contact center solution in place not only to serve existing and potential customers but to also carry out campaigns. Given that people these days use their mobiles and desktops and a variety of communication tools, it follows that a simple voice channel is not sufficient. Omnichannel is the solution.

When it comes to omnichannel contact center software it is likely that ecommerce enterprises may think of getting a customized solution that will serve their needs or opt for proprietary systems from leading brands. In both cases cost is high and so is maintenance. Customization may not yield the desired outcome and it may be time consuming as well. Ecommerce industries would do well to look at fully packed and ready to go call center software with a host of virtually all features they are ever likely to need.

It is best to select an omnichannel call center solution based on an open source platform like FreeSWITCH and look for features such as predictive and auto dialer, intelligent call routing, proactive outbound calls, voice blast, skill based mapping and automatic call distribution on the technical side. From the management perspective the omnichannel contact center software should naturally include real time analytics and reports. From the employee perspective your omnichannel software for communication should have a dashboard that gives employees a pipeline into backend CRM, access to any channel through which a person initiates a call and the ability to switch between channels. The key differentiator to a modern contact center solution is whether it incorporates WebRTC or not.

WebRTC in Omnichannel for E-Commerce Industry

WebRTC in omnichannel call center software can be a game changer for ecommerce enterprises in being able to give a very personalized experience and higher levels of satisfaction. A caller does like to see a face, not talk with a disembodied voice. Being able to see the agent does engender confidence as well as trust. WebRTC also makes it easy to share on the screen documents such as invoices. A caller with an issue can show the package he has received and the defect in the product helping the agent to take a decision on the spot. WebRTC is open source too and blends easily with FreeSWITCH platform.

Look for open source platform based call center solutions and you get the best in terms of price and performance, especially when the call center solution is developed specifically keeping in mind needs of ecommerce segment.

To know more about Omnichannel Contact Center Software for e-commerce Industries, visit

https://www.acinfosoft.com/ac-smart-call-center-software/

#omnichannel call center solution

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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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watertowerloft-blog · 6 years ago

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VOS3000 Web Based Billing With Anti Hacking Solution

All in one at Dashboard, Reseller Level 4, Account, Rate, Exchange Rate, Package, Reseller, IVR, CDR & Gateway Management, Ip Trace & Block, Reboot, Everything can be Monitor in a VOS3000 single Web Based Billing. Get comfortable, Monitor and manage your clients With our VOS3000 Based Billing systems anywhere.

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usedappliance-blog · 6 years ago

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vos3000 server

We provide best class vos300 server. For audio, data and multimedia services control, connection management, protocol adaptation, address resolution, routing, authentication, billing and so on.

#vos3000 vos3000 rent vos3000 softswitch vos3000 server voipswitch rent voipswitch ViciBox Dialer Asterisk FreeSWITCH FreePBX

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arystonstechnologies · 5 years ago

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Sip Trunk

On-Demand Cloud SIP Trunking

Low Cost – Self-Service – Flexible – Reliable Arystons SIP is a business-class SIP trunk solution provider for IP-PBX systems and analog/digital telephone adapters. our solution operates your own broadband Internet connection, and that we offer unlimited rate plans. Our easy setup, T-1 network, and powerful auto-serve SIP control panel have made us the leading on-demand SIP provider. We’re so confident that you’ll love our service, we provide free trial accounts which include an hour of outbound calling to anywhere within the lower 48 states to check our service. It’s easy to get started. Once your email authorization got completed, Our SIP Solution automatically creates an SIP trunk for you and deposit test credit. You can then configure it to your system and begin making calls immediately. The Ultimate SIP Platform Arystons SIP provides SIP trunking service for IP-PBX systems and analog/digital gateways. Arystons SIP Trunk solution works together with your existing Internet connection and supports any SIP-compliant device including; IP-PBX, EPBX, SIP softphones, Analog telephone adapters or T1 digital gateways. Our SIP Trunking service is a perfect fit open source systems like Asterisk, FreeSwitch, Elastix, PBX during a Flash and other popular Graphical User Interfaces to configure and control Asterisk. GET SIP Trunks Immediately Our SIP trunks are provisioned immediately and are available with a free trial that has up to an hour of calling, allowing you to form sure everything works to your satisfaction. We won't ask for a credit card for your free trial. We have an easy, yet powerful online instrument panel which enables you to simply manage your SIP trunks and telephone numbers. Everything is completed during a fully automated, self-service environment. Powerful Control Panel We offer a powerful easy-to-use web-based dashboard to manage all aspects of your account. You can purchase and manage phone numbers; add, change or discontinue service; review and export real-time call data records and manage to the bill. T-1 Redundant Network Arystons SIP Trunk utilizes only T-1 upstream providers to route our SIP traffic. We do this to provide the highest quality voice experience and a redundant network for outstanding reliability and performance. Simple Pricing Arystons SIP solution is prepaid with no-contract. The rate is manageable with our high-level solution plains. Add channels or cancel anytime. Upgrade channels online in the Control Panel. Bring Your Own Bandwidth Unlike other SIP Trunk providers that require you to get high bandwidth from them, we offer you the flexibility to run sip trunk on any isp provider & with manageable bandwidth as our sip trunk work with VoIP functionality. Widespread DID Coverage Arystons SIP Trunk offers a large database of DID numbers across the US. We also offer International DIDs around the world. Most DID number orders are provisioned instantly. Telemarketing Support If your company uses automated dialing for telemarketing, Arystons SIP Trunk can connect you with upstream carriers who welcome this type of traffic and help you avoid costly dialer surcharges. Read the full article

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jpc-us · 5 years ago

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JPC Technologies

JPC Technologies is our passion to deliver business growth through Information Technology. Idea to Innovation is our mantra to succeed. We have a in-house innovation center to prepare us for tomorrow's technology & business needs. We offers a wide range of innovative IT services and solutions to enterprises and technology driven businesses. We have combined IT experience of over 100 man-years, with delivery of 100+ business-critical apps. We serve wide-range of industries from MNCs to startups, from government organizations to big shipping companies, from VoIP companies to major Banks. We understand business is dynamic and we are flexible to your needs. We are highly customer focused putting our customer's interest before us. We are a strong team of technology and process enthusiasts, who are willing to take-on new business and technological challenges. We broadly classify in 3 areas: 1) IT Services » Mobile app: Native iOS, Android and Windows. Cross-platform using Phonegap, Titanium, Ionic UI framework, SDK development » Web app: Script: PHP, ASP.NET, Node.js, Java, HTML5, Angular JS Server: Apache, Nginx, Tomcat, IIS Mongodb, MySQL, MS SQL, Postgre » WebRTC: Native iOS, Android and Windows app, HTML5 based app audio, video, IM » IoT: Working on specialized M2M signalling protocol that consumes low power, low processor and low bandwidth » VoIP/SIP We provides Setup, customization, integration and maintenance services for Asterisk, FreeSwitch or any other proprietary soft-switch. Expertise to integrate Mobile applications with soft-switch 2) IT Solutions Our Solutions are available on white-label basis. [email protected] for details. » zutesipdialer.com - SIP based mobile dialer for iOS and Android » zoro.im - WebRTC based native and cross-platform 3) IT Consulting Project Management, Lean, Six Sigma

Source: LinkedIn Public Company Page

#JPC Technologies #domain #3l #tld #company #jpc

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freeswitchbilling-blog · 6 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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gventuretech · 10 months ago

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The FreeSWITCH Dialer is a powerful tool that enables businesses to make outbound calls efficiently. It is built on the FreeSWITCH platform, which offers a modular architecture for customized solutions that can manage voice, video, and data on a single line, supporting communication between PRI and SIP lines. The dial plan is a decision tree that provides routing services to bridge call legs together, execute dial plan applications, and invoke custom scripts that you write. The mod_dialer module is a FreeSWITCH module that allows for making mass calls and sending them to an IVR, message, or whatever. It is designed to meet the needs of businesses that require making mass calls and when connected, send them to an IVR, message, or other destinations. The dialplan is not a scripting language and is not a single entity. Instead, it is a unique, independent method through which you can access information.

#freeswitch #freeswitch dialer #freeswitch software #pbx freeswitch #freeswitch pbx #freeswitch gui #gui for freeswitch

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t1e1j1pricardformumbai · 7 years ago

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crmvoipsoftwaresolutions · 3 years ago

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Asterisk Development Services to Build the Best Call Center Solution

VoIP stands for voice over internet protocol. The VoIP has different development technologies such as Asterisk, FreeSWITCH, WebRTC, Kamailio, and OpenSIPs. Each VoIP development technology can be used to build different types of communication and collaboration solutions.

The Asterisk is one of the most robust and scalable VoIP development technologies. It has been in the industry since 1999. In fact, it was the first VoIP development technology. It is fundamentally a PBX solution and that is why it has all features of the PBX software. One can use the power of expert Asterisk development to build any type of VoIP solution. Once the VoIP solution is developed via Asterisk development, then that solution can work on both, SIP aka VoIP and PRI aka traditional lines with ease.

A call center solution is one of the most widely used VoIP solutions. It is used in large BPOs, KPOs, call centers, banks, insurance companies, automobile companies, and other industry verticals to run inbound, outbound, and blended calling campaigns. The call center solution comes with a wide array of features to automate and enhance the calling campaigns in the call centers.

Asterisk development can be used to build the most robust, reliable, and scalable call center software. Of course, FreeSWITCH development is in trend and there are many developers that insist using FreeSWITCH over Asterisk, but Asterisk is tried, tested, and proven technology. One of the most renowned call center solutions, VICIDial, is developed in Asterisk. It is an open source call center solution and it has amazing features. However, being an open source solution, it has certain limitations. Thus, people seek for other more feature rich call center solution. Asterisk development can be really helpful here.

With Asterisk development service, one can get the following features in his or her call center solution:

Web based access panel

Dashboard

Live call view

Call forwarding

Call transfer

3-way conference calling

Call recording and playback

Barge-in

Whisper

Soundboard Avatars

Disposition bucket

Preview dialer

Power dialer

Predictive dialer

Auto dialer

Manual dialer

Advanced call distribution (ACD)

Skill based call routing

Sticky agent

Priority call routing

Call queue

Interactive Voice Response (IVR) solution

MoH (music on Hold)

Reports

And more

If you take the help of an expert Asterisk development company like AC InfoSoft, you can build any features you want in your call center software. Also, this call center solution will be easy to scale up. It means if you get the call center solution with 50 agent seats, it can be extended to 5000 seats with ease.

In a nutshell, Asterisk development is perfect to build a reliable call center solution. If you are looking for a reliable partner, you can take the help of AC InfoSoft. The company has developed the best call center software which is available as a ready-to-use solution and the company also offers custom call center software development. To know about its Asterisk development services, visit https://www.acinfosoft.com/call-center-software-development/

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jennabrileyus · 7 years ago

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Kamailio Consultants

Add your entry here (Alphabetical order by company): This page is growing large. Please don't post logos!!

Page Contents

Asipto

CryptoVoip Technologies

dOpenSource

Dr. SIP (Zürich, Switzerland)

Edvina AB, Sollentuna, Sweden

Evariste Systems

Eurolan VoIP Solutions, Sofia, Bulgaria

FavaJam Co

Fred Posner

HM Kias

Jeremy McNamara

Klarya srl, Modena, Italy

Laszlo B.

LOD Communications

OpenXg Inc.

Oralnet

Sipwise

Team Forrest (The Palner Group, Inc.)

VoIP Embedded Inc.

VoIPxSWITCH

Consultant:

Asipto

Web site: http://www.asipto.com

Daniel-Constantin Mierla, co-founder OpenSER/Kamailio

Elena-Ramona Modroiu, co-founder OpenSER/Kamailio

OpenSER/Kamailio services

training - SIP Router Masterclass

consultancy

technical support

development

OpenSER/Kamailio solutions

Internet Communication/Telephony Platform (Voice, Video, Instant Messaging and Presence)

SIP Prepaid System

Standalone Nat Traversal Box

LCR, Load Balancer and Traffic Dispatcher System (Scaling PSTN Gateways, Asterisk, FreeSWITCH, etc...)

CryptoVoip Technologies

Web site: http://www.cryptovoip.in

VoIP and SIP Anti Block Solution

Mobile Dialers

PC Softphones

Web based Softphones and Click to Call

Anti Block Tunnel Server

Kamailio/ PJSIP /Freeswitch services

training

consultancy

technical support

development

Kamailio/ Freeswitch/ PJSIP/ RTP PROXY/ TURN PROXY/ VoIP BILLING solutions

24x7 Servers Monitoring and managed data services.

Use efficient tools to monitor all VoIP Servers critical parameters like Bandwidth, Memory, Processes, etc

Report or raise alert if any of the service goes down or if any issue

Resolves in very short time to ensure almost no impact on customer business. ... from Updates & News http://www.voip-info.org/wiki/view/Kamailio+Consultants

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watertowerloft-blog · 6 years ago

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We offer all type of Softswitch hosted package with excellent performance server and service. We will setup everything and make ready to go for you.

#vos3000 vos3000 rent vos3000 softswitch vos3000 server voipswitch rent voipswitch ViciBox Dialer Asterisk FreeSWITCH FreePBX

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bestvoipdevelopmentcompany · 7 years ago

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Best VoIP Development Company | Gventure Technology

Gventure Technology is one of the fastest growing VoIP Development company in the USA, India and beyond. We offer all VoIP Software development like FreeSWITCH development, Asterisk Development, IVR Solution, IBD/OBD Dialer and more at very affordable price range with the best service.Our Primary aim is to get full customer satisfaction on behalf of our work. Visit our website today to know more about us.http://www.gventure.us/

#best voip development company

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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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