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Hosted IVR Solutions

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This page lists providers of Hosted IVR solutions.

Hosted IVR Solutions Providers


Acclaim Telecom Services, Inc.

1300 E. Arapaho Richardson, Texas 75081
Suite 206
Richardson, Texas 75081
866.324.6416

Acclaim Telecom delivers the following solutions for your IVR needs:

  • IVR Hosted Solutions / Software as a Service (SaaS) / Cloud Computing Solutions
  • Custom IVR Application Development and Support
  • IVR Managed Services / Managed Premise IVR Solutions
  • Acclaim Telecom Mobile Solutions
  • IVR Extended Life Maintenance Programs

We Extend the Value of your IVR Systems

It’s not just about technology; it’s about solving your business needs. Acclaim Telecom Services helps our clients achieve greater top line revenue, improved customer satisfaction, increased company productivity, and reduced operating costs.

For more info: http://www.acclaimtelecom.com
For more info details on IVR Solutions:
IVR Hosting - http://www.acclaimtelecom.com
Hosted IVR Solutions - http://www.acclaimtelecom.com
IVR solutions - http://www.acclaimtelecom.com/solutions/



Arcutel - Enterprise Hosted IVR

https://www.arcutel.com

We provide an automatic telephony service to businesses of all sizes. You setup and have full control over your own IVR via our web based application. There is no investment required, we don't charge extra for premium features, and the first month is free.



AVOXI from AVOXI


AVOXI virtual call center solutions provide virtual call center products like SIP trunking and VoIP gateway solutions, with international toll-free numbers. Contact AVOXI at +1-770-937-9735 to learn more about these products and services.


Other Services:
Cloud Based Contact Center
Freephone Numbers
Hosted PBX Providers
International Toll Free Number
VoIP Call Center

Bright Pattern Interactive Voice Response


Bright Pattern delivers next generation cloud contact center solutions, including Interactive Voice Response. Gracefully build your IVR logic flow using our drag-and-drop interface, Scenario Builder, part of ServicePattern, our platform. Quickly build sophisticated interactive voice response applications that follow sequential or branching flows. ...

Atlanta VoIP

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This is a list of VoIP providers in Atlanta. Atlanta VoIP companies typically support multi-line telephone systems, small gateways and hosted VoIP. Please add VoIP Providers in Atlanta to the list below.

softswitch

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Softswitch is a central device in a telecommunications network which connects telephone calls from one phone line to another, across a telecommunication network or the public Internet, entirely by means of software running on a general-purpose system. Most landline calls are routed by purpose-built electronic hardware however, soft switches using general purpose servers and VoIP technology are becoming more popular.
Nowadays, many telecommunications networks make use of combinations of softswitches and more traditional purpose-built hardware.

A softswitch is also a VoIP server, providing a soft switch platform with full IP PBX call features. The most difference from IP PBX is its enormous numbers of users.

Typical application networking diagram
Typical softswitch application.jpg


See also


Softswitch Manufacturers and Providers


iTel Switch
iTel Switch is a single Softswitch platform for global Retail, Wholesale, Calling card & Call shop business. Being a highly customizable and scalable VoIP Softswitch with integrated billing, it serves as an ideal platform for all the VoIP service providers who want to provide a wide range of VoIP services.

Key Features

● Rate Plan Management
● Flexible & Advanced Routing
● Live Call Monitoring
● Multilayer Security
● Analysis & Reporting
● Behavioral Based Alarm System
● Call Simulator for Route Checking
● Client Management
● DID Management
● Payment Management

Get a Free Demo

  • SyncSwitch SyncSwitch Standard Platform is a web based interface which gives users a wide variety of business model implementation possibilities including Calling Card, Call Shop, Call Back, PC to Phone, Mobile Dialer, Device to phone, On net calling etc. Supports wide variety of free and proprietary codecs. SyncSwitch Standard Platform is complaint with Session Initiation Protocol(SIP) & IAX2 and facilitate conversion among them. Provides CDR and Billing. Enhanced security measures.


MediaCore Softswitch

mediacore-133_31.gif

Download overview

MediaCore is a high-performance Class 4 Solution for VoIP and SMS services developed by Speedflow.
It has the following features:
● Switching
● Routing
● Billing
● Transcoding
● Unique revener assurance mechanism 'Guardian'

Speedflow provides buyout and hosted/rented options. FREE TRIAL PERIOD is available.

Read moreat Speedflow web-site
Get demo:http://speedflow. ...

ZONE Limited

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just ZONE-smaller2.gif
http://www.zonetel.com


Established in 2000, ZONE Limited is a licensed telecom service and solution provider in Hong Kong. ZONE specializes in VOIP/SIP/Asterisk PBX/Call Center/IVR and have a number of successful implementations in Asian regions. ZONE endeavours to provide customers with best products and professional services.

ZONE is the distributor of Elastix in Asia and also an approved Avaya Partner.

NEW

NewRock OM20/50 IP PBXs


  • All-in-one IP PBX with built-in FXO + FXS + SIP support
  • 16G builit-in storage
  • Assistant tool in PC for quick setup of contact center
  • Self developed WeWei Apps to maximize productivity on mobile
  • User-friendly Web GUI
  • Flexible and easy for branch expansion
  • Remote accessibility via the NewRock Cloud

For further details, please visit here


Services

DID numbers/Hong Kong Virtual Phone Number/Shanghai DID/Singapore DID

ZONE is a licensed telecom operator providing Hong Kong/HK DID numbers (HK virtual phone numbers) for numerous SIP platforms like Asterisk, Elastix, FreePBX, Microsoft Lync and even SIP-compliant apps on Android or iPhone. You can get HK presence without all the expensive overhead. The DID comes with optional outbound dialing, voice recording, voice mail, call forward and concurrent call support.

Many call centers and virtual offices are using our HK DID services.

In addition to HK DID, we provide Singapore and China DID for you to develop the Asia markets.


IP-PBX

We carry IP-PBX products from selected brands which we believe to represent the best mix to serve SME/SOHO segments. We provide on-site installation, integration, customization and training so that customers can make full use of the IP technology.

We recommend the Elastix miniUCS for SME :

  • SIP/T1/E1/FXO/GSM connectivity
  • Small size
  • Low power consumption
  • Complete PBX functions : Auto-attendants, Voicemail, Ring Groups, Queue, Conference, Fax Detection, Remote Management
  • Office extensions on Android/iPhone and remote offices
  • Conversation recordings and many more..

For further details, please visit here

IP-Phones

Our IP-PBX can work with SIP-based IP phones from leading brands such as Alcatel, Digium, Elastix, Grandstream, Polycom, Snom, Xorcom, Yealink and many more,

Asterisk consultation and turnkey solutions

Asterisk is known to be the leading open-source PBX attributed to its powerful features and deployment flexibility. We have intensive exposure to Asterisk and could help you to implement various business applications such as calling card, IVR, voice recording, fax2email, auto-attendant, dialers, etc. ...

voip-info.org

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Welcome to the VOIP Wiki - a reference guide to all things VOIP.


This Wiki covers everything related to VOIP, software, hardware, VoIP service providers, reviews, configurations, standards, tips and tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony.

Your contributions are welcome, please read the How to add information to this wiki page and the Posting Guidelinesbefore you post.


NEWS


News Resources


Getting Started

Old News

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REVE Systems

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REVE Systems started in 2003 with a focused approach to serve the IP based communication industry. A Telecommunication and Software Solution provider, REVE Systems has a wide assortment of products, ranging from backbone infrastructure to peripheral products, including middleware. The company today holds a leadership position in Mobile VoIP, Softswitch& Billing and Bandwidth Optimization solutions.

iTel Mobile Dialer

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iTel Mobile Dialer is a SIP Softphone, which allows users to make long distance VoIP calls from mobile phones. Compatible with all standard SIP Softswitch, this mobile dialer has been designed keeping VoIP service providers requirements in mind. Being a service provider, you can run this softphone software under your own brand name as iTel Mobile Dialer is completely customizable.

iTel Switch

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iTel Switch is a single Softswitch platform for global Retail, Wholesale, Calling card & Call shop business. Being a highly customizable and scalable VoIP Softswitch with integrated billing, it serves as an ideal platform for all the VoIP service providers who want to provide a wide range of VoIP services. iTel VoIP Softswitch has been designed to meet the highest needs of carriers. This VoIP Softswitch also ensures most reliable and cost effective solution that can help VoIP service providers grow as a giant global carrier in VoIP industry. Multilevel reseller support, easy end user interface, integrated billingintegrated billing, intelligent routing and class 4 & 5 Softswitch features are some of the many unique competencies of iTel Switch.

iTel IM Dialer

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iTel IM Dialer through its cutting edge Instant messaging (IM) service allows its users (registered under service providers' brand) to communicate instantly with others which is available in both android and iOS platforms. After registering and authenticating his/her mobile number, users can instantly exchange messages with others whenever, from wherever. Users can also make audio & video call, share files & location, mobile top up and many more. Utilizing the complete customizability and white label branding facility of iTel IM Dialer, service providers can differentiate them from competitors and eventually expand their business successfully.

For more details:www.revesoft.com

How to start a VOIP Business

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The first thing to do is decide what part of VOIP marketplace you want to serve. Here are some possibilities:

  • VOIP Provider services
  • VOIP consulting
  • Independent Sales/Service Agent for existing VOIP service providers
  • Value Added services with VoIP
  • etc.

Some general suggestions:

  • Pick an area that plays to your strengths. For example, if your strength is sales and marketing, pick an area where you can leverage those abilities
  • Learn all you can about the maketplace
  • Attend industry tradeshows
  • Read industry magazines, blogs, forums, etc
  • Read books
  • Do market research - talk to your potential customers
  • Ask questions
  • Test the waters — to the extent possible try before you buy, test the waters before making large commitments of time or money

Value Added services

If you have experience with VoIP or already in VoIP business, you can get benifit / new customers by introducesing some value added services on VoIP. Few value added services are mentioned in following, Within each service there are many choices.
  • PBX sales and service
    • Hosted PBX
    • Virtual Numbers
    • Hosted IVR / Auto attendents
    • etc

  • Message broadcasting / Call Center Solutions

  • Prepaid Cards
    • Retail prepaid cards from existing wholesale providers
    • Start your own brand of prepaid cards using services from existing wholesale providers
    • Start a new prepaid card provider company
    • Create new software package for prepaid card services
    • Create a Free Phone Booth
    • etc.


Asterisk SIP Trunking - US — Wholesale Reseller User Portal and Admin Portal. With this service you can customize your own Admin and User portal to sell SIP Trunk Services to your end users. This means that you can set your own rates to your customers, your admin portal will charge your customers in real-time so you never have to chase your money. All services are turned up in real-time for your end users. This includes DID ordering, SIP Trunks, vFAX and all other telecom related services. This is the ultimate reseller portal program. As for the requirements you must have an Authorize.net Account and subscribe to the CIM for remote PCI protection of credit card numbers. When your customers pay the money is sent to your bank for each nightly batch settlement. The funds are transferred directly to you and our services are billed to your account at a wholesale rate. Call now: 877-686-4787 or visit us:

Mobile VoIP

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Mobile VoIP is an efficient, low-cost way to communicate using your cell phone and the services provided by your home or business VoIP provider.

How Does Mobile VoIP Work?


Mobile VoIP works with a cell phone’s 3G, 4G, GSM, or other Internet service to send voice calls as digital signals over the Internet using voice over IP technology. Mobile VoIP phones can also take advantage of WiFi hotspots to eliminate the calling costs of a cellular voice or data plan.

By using VoIP, mobile VoIP phone users — especially smartphone users — can benefit from lower costs when calling, texting, or other common smartphone activities. Digital data transmission using VoIP is also typically faster, as the data is spread out over multiple packets, each taking the fastest route to its intended destination.

Using a mobile VoIP phone with WiFi hotspot access can also reduce a mobile VoIP phone user's costs by sidestepping the carrier's expensive 3G service altogether. For instance, with a cellular carrier's monthly data plan, callers can easily exceed bandwidth maximums, incurring overage charges. Tapping into WiFi hotspots with mobile VoIP software reduces that risk and extends the lifespan of the monthly data allotment.

A mobile VoIP phone service can eliminate the need for a basic voice plan, as well as optional (and costly) text add-ons. With a mobile VoIP phone, cell phone users can enjoy more flexibility in calling times than a cellular voice plan provides, with fewer restrictions. VoIP mobile phone service means that a mobile VoIP user can make unlimited inexpensive or free calls using voice over IP technology at any time.

Mobile VoIP users don't need to worry about the limitations associated with cell phone calling plans, such as:

  • Anytime minutes
  • Night or weekend minutes
  • Rollover minutes
  • Roaming charges
  • Incoming call charges
  • Messaging limits
  • Mobile-to-mobile calling (check with your mobile VoIP provider, some do treat in-network calls differently)

Mobile VoIP phone users can also take advantage of the additional, integrated features a mobile VoIP app supports. This includes high-bandwidth activities such as group chat and video chat. Accessing these functions without mobile VoIP software (by fring or Talkonaut, for instance), typically requires a separate app, and using it could impact or exceed monthly text and bandwidth maximums.

Accessing Mobile VoIP

Cell phone users can use mobile VoIP service on their phone with the addition of mobile VoIP software. These are apps offered by VoIP phone service providers customers may already be using at home or at work, such as Vonage, or standalone mobile VoIP apps such as Skype, Vyke, or Truphone.

Some services, such as Truphone, also offer an entire mobile VoIP network by combining a SIM (Subscriber Identity Module) card and an app together. ...

VoIP Providers Canada

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This is a list of VoIP providers in Canada. These companies typically support multi-line telephone systems, small PBX gateways and hosted VoIP. Please add VoIP Providers Canada to the list below.


Business VoIP Providers in Canada

  • VoiceMeUp Largest Coverage for Canadian DIDs for business usage. Channel agregation available. Wholesale and Affiliates program. T.38 Supported In/Out. Supports SMS, Smart Caller ID. User accessible interface.
  • Netfone: Specializes in Cloud Hosted Business Phone Systems, Business VoIP Lines, SIP Trunks, and On-Premise Phone Systems using only premium business grade service. The company has by far the best cell phone integration with their service. Netfone provides service throughout Canada. Offer managed business ADSL internet in western Canada ensuring that voice is traveling in our private network and ensuring quality of service.
  • Zaplee LLC - Zaplee is a robust, reliable cloud phone option for small business, startups and budding entrepreneurs. Create a new account in under 10 minutes and get started creating new local, international, 800 and toll free numbers. You can forward these lines to the device of your choosing. It has 14 days free trial on membership account. It has very low price on Canada local & tollfree number.
  • CHECKBOX, VoIP Carrier, Wholesale A-Z, CALL CENTER CARRIER - We provide VoIP connection specially for call centers / marketing companies who needs High stress / CPS routes, We have USA CC, Canada CC, Australia CC, China CC, UK CC and others Direct Routes to many countries.
  • VoIPInvite: VoIPInvite is FCC licensed Wholesale VoIP/SIP termination service provider in Canada at very low rates and best quality.
  • RingOffice: Specializes in Cloud Hosted Business Phone Systems, Business VoIP Lines, SIP Trunks, and On-Premise Phone Systems using only premium business grade service. Coverage extends throughout Canada and USA. Offers a comprehensive customer portal with 24/7/365 support.
  • FFB VoIP Services | FFB Services specializes in small business VoIP solutions helping your business make a big impression. FFB Services works with you to get the right solution every time. Over 50 standard features on every line and no matter if you're 1 or 1,000 seats, you get the same quality customer service 24/7. Locally owned and operated, proudly serving all of Saskatchewan. Check us out at http://ffbservices.com/
  • AstraQom AstraQom Business Solutions include Hosted PBX, Live Answering, Computer Telephony Integrations and Business Internet. The AstraQom enterprise-grade VoIP solutions included: SIP Trunking, DID Numbers, A-Z Termination & VoIP Colocation.

Ecessa

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Ecessa Corporation manufactures network control hardware and software for data and telecommunications. Since 2000, the company has focused on Internet, cloud and wide area network technologies to give business networks greater reliability, redundancy and the ability to confidently use more affordable types of bandwidth. Specific to VoIP, Ecessa has developed several products to improve call quality and reliability.

Ecessa PowerLink™ http://www.ecessa.com/powerlink/ balances up to 25 WAN lines (T1, xDSL, Cable, ISDN, Wireless, Satellite, fibre, etc.) into a greater pool of available bandwidth while providing business continuity through ISP failover and redundancy. The appliance comes in various models that support bandwidth from 60 Mpbs to 20 Gbps. It was recently updated to include network security with a built-in firewall and VPN gateway.

ClariLink™ http://www.ecessa.com/clarilink/ incorporates all of the above features, plus provides real-time call failover for hosted VOIP. ClariLink optimizes call performance by managing SIP traffic among multiple Internet links to avoid jitter, latency and congestion. ClariLink monitors incoming and outgoing calls in real time. In the event of a WAN failure or call degradation (i.e. high latency, excessive jitter, packet loss, etc) it automatically moves active VoIP calls to better performing WAN links, without the call dropping or the user even noticing. This means that VoIP users whose providers support these features never have to worry about another dropped or low-quality VoIP call. Ecessa is the only service/product offering this advanced feature.

The new Ecessa Edge™ combines failover, load balancing and SIP features for small offices and branch offices at an affordable monthly price point. It accepts up to three WAN/ISP links and supports up to 150 Mpbs bandwidth.

Ecessa's most advanced product, WANworX™ http://www.ecessa.com/wanworx/ software-defined wide area network (SD-WAN) solutions, are the most scalable and flexible technology for improving WAN performance and ensuring Never Down™ network availability. WANworX enables enterprises with multiple locations to seamlessly connect all their locations and connect to the data center, Internet and cloud-based networks.

By integrating multiple WAN links from different carriers, WANworX supports multiple paths for data to travel, and continuously monitors links and routes traffic over the best-performing links, ensuring not only seamless uptime but also predictable application performance, even when carrier service level fluctuate. This is particularly important for real-time applications, such as VoIP, videoconferencing and VDI, where even a slight delay can result in dropped calls, glitchy video and dropped sessions - plus wasted time while all those services must reconnect.

Ecessa products enable network engineers to strategically design wide area networks they way they want to, add links, add rules, and not shy away from complex multi-layered networks that will accomplish their performance goals. And it allows them to manage the whole thing confidently with cloud-based controls.

Within the Ecessa suite of products, you'll find solutions for:
  • VoIP Failover and Monitoring
  • Multi-ISP WAN Failover
  • Intelligent Inbound and Outbound Load Balancing
  • Proactive monitoring and reporting of WAN network traffic
  • Network Traffic and Application Visibility, Traffic Shaping and Application Prioritization
  • Firewall and VPN
  • WAN Virtualization and VPN Tunnels between sites

Company News


ECESSA SD-WAN TECHNOLOGY GIVES SUMMIT PARTNERS A HIGH RETURN ON INVESTMENT
http://www.ecessa. ...

Call Center Software

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Call center software is the software system that allows a company or organization to run a call center. This page lets you compare call center software providers.

There are hundreds of different providers of call center software across the globe, and every call center software system has its pros and cons. When selecting the right call center software for your business, contact center, or call center, it's important to decide which features you want your phone system to have.

Types of Call Center Software


ACD helps productivity by assigning inbound agents to incoming calls. The automatic call distributor uses a set of instructions to determine who gets the call in the system. The algorithm can route calls based on agent skill or whoever has an idle phone. ACD can use caller ID or automatic number identification, but usually interactive voice response is enough to help the system determine the reason for the call.

An automatic call distributor can also take advantage of computer telephony integration. Agents can receive relevant data on their computers along with the incoming call.

Computer telephony integration is a broad category of software that connects telephone and computer systems. Computer telephony integration software can have both desktop and server functions. Various applications make up a system that can control phones, display call information, and route and report calls.

Interactive voice response allows callers to route themselves to the appropriate department or use the company’s database for assistance. More sophisticated interactive voice response systems can access accounts and perform certain tasks, such as activating a credit card through a bank’s phone system. IVR involves using dial tone multi-frequency or voice commands. In the VoIP industry, a PBXauto attendant is near interchangeable with IVR. However, auto attendants are not capable of speech recognition.

A predictive dialer calls a list of phone numbers at once. Outbound agents are then connected to the numbers that answer. A predictive dialer uses calculations to minimize the idle time of agents and the potential of losing answered calls when no agents are available.

Contact Center Software

For contact centers, software includes applications for chat, email, and web interaction in addition to telephony functions.

Call Center Software Providers

This is a list of call center software providers and developers. Please keep this list in alphabetical order.
  • Thirdlane Call Center software delivers a reliable and highly-customizable contact center solution with sophisticated features and functionality for a broad range of call centers - on-premises or in the cloud. Thirdlane Call Center is an add-on that is integrated with Thirdlane PBX products and offers set of features supporting environments ranging from professional contact centers to workgroups requiring more advanced queuing, call distribution, and call metrics features.

BrightPattern

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We help call centers to raise both customer experience and agent productivity using our excellent Multi-channel Cloud Call Center Software platform, with the help of our all-star customer success management team.

Our solution features true multi-channel functionality, non-disruptive update and HTML5 agent UI working in all major browsers.

Our customer success team is getting perfect survey scores from customers.

Salesforce Integration
Zendesk Integrations with Bright Pattern contact center, including Zendesk chat integration
RightNow Contact Center Integration
In-Mobile-App Customer Support
Predictive Dialer
SMS call center software


Asterisk CRM Integration

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Asterisk, being open source, can be integrated with a whole lot of different software solutions to enhance, improve and facilitate general users. One of such technological marriages is Asterisk-CRM integration. Asterisk-CRM Integration can provide your Asterisk with the ability to manage your customers directly. The CRM integration to Asterisk provides notes uploading, call recording, customer management, number lookups and a number of other benefits that a standalone Asterisk or independent CRM can not provide.

There are the number of products available in the market to achieve this goal. This page lists all of such products and solutions with alphabetical sorting.




i6net

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VoiceXML for everybusiness | www.i6net.com

I6NET provides a VoiceXML browser for Asterisk :

Products


VoiceXML Browser for Asterisk VXI*


The VXI* VoiceXML browser for Asterisk® gives operators and solution providers the ability to rapidly develop and deploy innovative voice and video applications via IP, PSTN, and 3G-324M networks. VXI* is fully compliant with the W3C's VoiceXML 2.0+ specification and is integrated with automatic speech recognition (ASR) and text-to-speech (TTS) software to enable advanced voice and video solutions, and real-time video calling applications. VXI* can be installed in common hardware configurations, providing a highly scalable base system to meet all customers' business and technical VoiceXML requirements.

VXI* VoiceXML interpreter works directly with the Asterisk® PBX software supported by Digium®. Not only can users of the open source PBX run VoiceXML applications in the same server, they can now offer these powerful, scalable IVR / IVVR solutions at an affordable cost.

Written in C, Linux Operating System, License Commercial

Key features and benefits

  • VoiceXML V2.0/V2.1 compliant
  • CCXML replaced by Call Control functions of Asterisk
  • Base on OpenVXI voice browser template
  • Application for Asterisk PBX (app_vxml)
  • VoiceXML accounts managment configuration (for hosting services)
  • Support plugging objects with the VoiceXML tag, SDK with API available
  • Fax support (send and receive)
  • Video silence parameter with VoiceXML syntax
  • Text-to-Speech (TTS) connector included
  • Automatic-Speech-Recognition (ASR) connector included
  • Text-to-Video (TTV) with HTTP connector (option)
  • Voice-Silent Detection (VSD)
  • MRCP v1 and v2 thru uniMRCP for TTS and ASR.

Text-to-Speech engines supported:

  • Flite - free (TTS) from CMU Speech Group
  • Voxygen (TTS)
  • Cepstral (TTS)
  • Verbio (TTS)
  • Nuance Scansoft (TTS)
  • Acapela (TTS)
  • Universal HTTP connector for any TTS engines
  • Universal MRCP connector for any TTS engines (uniMRCP)


Automatic-Speech-Recognition engines supported:

  • Lumenvox Speech Engine (ASR)
  • Verbio Speech Server (ASR)
  • VoiceInteraction Speech Engine (ASR)
  • Vestec Speech Engine (ASR)
  • Loquendo Speech (ASR-MRCP)
  • Nuance Speech (ASR-MRCP)
  • PocketSphinx (SpeechToText)
  • universal MRCP v1 and v2 (ASR) (uniMRCP modified with extra features)

Asterisk requirements:

  • Asterisk 1.4 / 1.6 / 1.8 / 11

Linux OS requirements

VXI* is built with a Debian Linux with GCC3 / libstdc++5 or GCC4 / libstdc++6 environment. To install, configure and use the the plugin for Asterisk® PBX for other distributions, please read the documentation.

Distributions:
  • Debian 6 / 7
  • Centos 5 / 6



Hosted VoIP

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Hosted VoIP is another term for Hosted PBX. "Hosted" means to say that the hardware and PBX is hosted at an off-site location from where the VoIP telephone service is being used. An office can have VoIP telephone service that powers their phones in the office, but their PBX could be hosted at their VoIP providers data center, thus the term: hosted VoIP.

Benefits of Hosted VoIP


There are many benefits to using Hosted VoIP rather than a traditional phone system, or an on-premise VoIP system. The main benefit is cost- a Hosted VoIP system costs much less to set-up than an on-premise PBX. In many cases, there are no set-up fees for a hosted VoIP system. Hosted VoIP phone systems fall under operational expenditure rather than capital expenditure, which also make hosted VoIP systems attractive to businesses.

Another benefit to a hosted VoIP system over an on-premise PBX is that hosted VoIP providers will take care of all the set-up and installation, meaning you do not need to be a telecom or VoIP expert in order to get hosted VoIP.

Hosted VoIP Service Providers


Please list hosted VoIP service providers below in alphabetical order.

1Pipe Telecom | OnePipe is a licensed carrier providing Hosted PBX and SIP Trunking in over 50 countries. We have regional network facilities spread across the globe.
  • Actual US CLEC
  • Branded customer portals
  • Multiple geographic locations on one Hosted PBX
  • Coverage in over 50 countries
  • Unlimited inbound on each channel
  • Great for inbound call centers
  • No fee's per extension


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Bicom Systems

Bicom Systems is the creator of Telco-in-a-Box, the most complete IP PBX Unified Communications solution for ITSPs, Service Providers, CLECs, and more. Telco-in-a-Box meets all of your telecom needs including telephony, billing, mobility, security, and more in one tidy package. With unmatched compatibility, stability, and reliability, Telco-in-a-Box is the tool to build and grow a telecom. The product suite includes:
  • SERVERware is a Cloud IP Services Delivery Platform
  • PBXware is an IP-PBX turnkey business communications platform
  • TELCOware is a provisioning platform that performs billing, accounts, and more
  • sipPROT is an advanced SIP security module
  • sipMON is a SIP monitoring module that analyzes and improves VoIP calls
  • gloCOM is a desktop and mobile business softphone


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VoIP Softphones

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What is a VoIP Softphone?

A VoIP softphone is a program that installs to and runs from your computer. A VoIP softphone enables you to make calls with just your computer using a VoIP service. Skype, iChat, and GoogleTalk are some of the more popular services, but there are many different ones available for you to choose from.

A VoIP softphone is accessible wherever you have a computer with you and there are both paid and free VoIP softphones. Companies often offer their own proprietary softphones that are configured to work with their service like Cisco and Counterpath.


The VoIP softphones are designed to be intuitive to use and most resemble an actual phone handset. Or you can choose to have the layout show your contacts if that makes calls easier. You can either click the buttons on the interface to dial or use the number pad on your keyboard.

VoIP softphones are only a program on the computer, so a headset with a microphone or an internal microphone and speakers are also needed to make the calls. Headsets prices start at a very affordable $5-$10 for a standard configuration and can get up in the $100-$200 area for a high-quality wireless headset with a long battery life and interchangeable ear pieces.

Who Can Benefit From VoIP Softphones?

Softphones can be used by anybody with a computer. There are a few types of users who can really benefit from the features of a VoIP softphone:

  • VoIP beginners
  • Heavy travelers
  • Telecommuters
  • Call center employees
  • Small businesses
  • Frequent long-distance callers


VoIP beginners can quickly and cheaply explore how the service works by downloading a VoIP softphone to make free computer-to-computer or computer-to-phone calls. Heavy travelers can avoid racking up large bills on their mobile phones or at hotels by using a low-cost VoIP service with a VoIP softphone. Telecommuters can register a VoIP softphone with their office PBX system to enjoy the same call features available to them at the office while they are on the move. Call center employees and small businesses can save on costs by pairing a VoIP service with a softphone to avoid purchasing and maintaining desk phones. International and long-distance rates are much lower when using a VoIP softphone, so those making regular or frequent calls out-of-state or country can cut some major costs.

VoIP Softphone Features

VoIP softphones offer the same features that traditional phones offer and more:

  • Call forwarding
  • Call conferencing
  • Hold capabilities
  • Call transferring
  • Voicemail
  • Greeting capabilities
  • Text, IM, and video capabilities
  • Echo cancellation to improve sound quality
  • Contact list/address book


Softphones also use less energy than phone and phone system hardware, which saves on costs and is useful for green companies.

VoIP Softphone Protocols

VoIP service uses different protocols to determine how the data is processed and transferred over the network. Your VoIP softphone needs to support the same protocol your VoIP service uses. ...

WebRTC

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Synopsis

The practical implementation of VoIP was started on hardware based IP Phones. The idea was well received and was transferred into the concept of Soft Phones or software based IP Phones. These softwares always required some additional installation to the native Operating System. Most common examples of Softphones or Software based SIP client is Counterpath's X-Lite and Bria.

The Evolution of Software Development made it possible to translate or formulate equivalent of almost every desktop based application to web based application. This brought major shift in Software Industry as the web browsers are integral part of almost every Operating System. SIP clients, were also transformed into Web Extensions. Most of the time, Flash was used to develop such extensions however, it always required extra plugin installation, thus decreasing system performance, and increasing chance to troubleshoot as it required additional resources to be deployed. And this problem gave rise to the concept of WebRTC.

Overview

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WebRTC provides the functionality of realtime multimedia applications without any installation of additional plugins, downloads or extensions. The ideal form of WebRTC describes such web based Real Time Communication independent of Browser being used by user. It's a Javascript based API originally being developed to develop browser to browser communication applications for Voice, Video and Peer to Peer File Sharing tasks.

Architecture

The architecture of WebRTC, as described by W3C looks something like this:
WebRTCpublicdiagramforwebsite (2).png


Design

Major components of WebRTC include:

  • getUserMedia, which allows a web browser to access the camera and microphone
  • PeerConnection, which sets up the audio/video calls
  • DataChannels, which allow browsers to share data via peer-to-peer

Support

Chrome WebRTC Development Team

Discussion List: https://groups.google.com/group/discuss-webrtc
Google Plus Page: https://plus.google.com/113817074606039822053
Chrome WebRTC Issue Tracker: http://code.google. ...

WebRTC vs VoIP

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Theres is a bit of confusion in the telecommunication industry as to whether or not WebRTC is compatible with or runs against VoIP. WebRTC is a viable Internet Protocol (IP) communications system that parallels and runs alongside the internet-based phone system VoIP. VoIP services and WebRTC solutions are both heavily promoted in the business and residential fields. So the confusion lays here: are VoIP and WebRTC providers friends or foes? Can the two systems coexist, do they overlap, and how does the client benefit from these?

The similarities

WebRTC and V.VoIP are similar in that both aim to enhance the user experience and enable any consumer device (whether it be mobile phone, fax, internet etc.) to effortlessly connect from anywhere and on any network internationally.

The differences

The primary difference between the two services is that VoIP uses a multitude of variants such as VoIP over DSL/cable modem, voice over Wi-Fi/3G (VoWiFi/3G), voice over LTE (VoLTE), and Rich Communication Suite (RCS), while WebRTC is solely focused on browser-based communications.

VoIP

VoIP is an online telecommunications system which offers simpler and more efficient technology than traditional phone service. VoIP uses advance phone technology in order to make phone calls from the office or home more cost effective and with more features. Standard telephone systems uses telephone lines to transmit phone calls, using physical circuits for connection. Since VoIP is cloud-based, calls are sent as digital data and no cables are needed to send the call so any kind of Internet connection can be used to make calls and from a plethora of devices. Millions of people and businesses have switched to VoIP in order to save money as well as to be able to access the same lines from any place and any device.

Benefits that most VoIP providers include are: around the clock customer service; reduced costs compared to traditional phones; no installation or service fees; free ad-ons including unlimited calling to the US and Canada, unlimited extensions, 1,000 free toll-free minutes, high-definition video-conferencing, desktop integration with popular CRMs, online PBX controls, virtual extensions, remote access, auto-attendants, and unlimited extensions for multiple office locations.

WebRTC

WebRTC (Web Real-Time Communication) is an API being drafted by the World Wide Web Consortium (W3C). Put simply, its a software intermediary that makes it possible for application programs to interact with each other and share data. WebRTC is used to enable browser-to-browser applications for voice calling, P2P file sharing, and video chat without plugins. WebRTC is an emerging technology that are accessed with JavaScript APIs and currently in development are an audio and video data stream as well as API which allow for two or more users to communicate browser-to-browser, real-time gaming, text chat, file transfer and other online based sharing.

Connecting vs. clashing

WebRTC makes it feasible for web developers to enable VoIP into their Web-based applications. Since WebRTC is in its early stages of development, it does not include any signaling protocol which leaves this choice and development and integration to the developer. By integrating a signaling protocol into WebRTC, a developer can create a full VoIP soft client on a browser.
One nice example of such VoIP soft client is CryptoVoIP SIP WebRTCDialer which uses SIP Protocol for signalling. The good part of CryptoVoIP web dialer is that it does not require web-sockets or webrtc support in SIP Servers or softswitch. ...

Unified Communications

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Unified Communications, often abbreviated as UC, is the combination of different types of communication mediums including telephone service (VoIP), chat, video and web conferencing, messaging, email, fax, and other types of communications. Unified Communications systems can be sold and implemented with a select amount of individual communication mediums, or many integrated into a single unified system.

Unified communications consist of integrating various real-time and non-real time communication services into a convenient package for the user. Examples of real-time services commonly used include instant messaging, telephony and data sharing. Non-real time communication services focus mainly on messaging services like e-mail, SMS, voicemail and fax.

With a unified communications solution, the user can receive a message through one communication medium and use another one to access it. One common example of this would be receiving a voicemail message on an office landline phone and using a mobile phone to retrieve it. ...
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