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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. ...

Predictive dialer

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What Is A Predictive Dialer?


  • "A predictive dialer is a computerized system that automatically dials batches of telephone numbers for connection to agents assigned to sales or other campaigns. Predictive dialers are widely used in call centers." - Wikipedia

"Definitions of Predictive dialer on the Web:

  • A predictive dialer is an outbound call processing system designed to maintain a high level of utilization and cost efficiency in the contact center. The dialer automatically calls a list of telephone numbers, screens the unnecessary calls such as answering machines and busy signals, and then connects a waiting representative with the customer.

note: above text may be copyrighted by it's respective owners)

A VOIP Predictive Dialer, a.k.a. soft predictive dialer, is a software product capable of predictive dialing using VOIP service directly. Besides computer and internet connection, there is no equipment needed in order to use VOIP predictive dialer.


Software Only Predictive Dialer

New predictive dialing technology, together with faster computers and bigger broadband bandwidth, enables software only predictive dialers to work as good as or even better than hardware based dialers. Software based solution avoids expensive telephony board and associated hardware maintenance cost. It is easy to install and configure. For example, it is very easy to setup remote agent (at home agent).

SOFTWARE ONLY PREDICTIVE DIALER

"Start your call center technology search with a 3G Callnet Call Center Advisor that can recommend the best solution for your business. 3G Callnet offers more configuration and budgetary options than any other provider in the world."

About US
With an experience of over 10 years in a dynamic field of technology equipped with talented Quality engineers having multi-skill expertise, wide experience and well-tailored professionalism which is a basic requirement of a Professional company, in addition to Analytical Capabilities, Engineering Skills and Innovative approach. GMS promotes itself as a Quality and Customer Oriented company with a dream to win hearts of its clients and continue a lifelong relationship by assuring world-class IT solution & services.

The company has shown consistent organic growth in the field. The company specializes in providing turnkey solutions through the vast knowledge gathered in the past years. We are providing services to some of the leading corporate sector and are also considered the preferred vendor by most of the telecom IT enabled Service providing companies in the region.

Dialer Major Features:

Computer Telephony Integration (CTI) with Customer Relationship Management (CRM)
Outbound, Inbound and Blended Telephony
Web-based agent and administrative interfaces
Predictive, Power, Preview and Manual Dialing Options
VoIP and Analog Telecommunication Options
Multiple ACD and IVR Features
Compliance Management
Custom Integration
Quality Assurance
Reporting
Recording
Scripting
Agents operate remotely
Call Monitoring, Coaching, Barging, Conferencing
Our Experties:

• Call Blasting (Agentless Dialing)
• Text to Speech - T2S (Agentless Dialing)
• IVR – Interactive voice response system
• Additional Customized feature required by customer.


Contact For More Details : Gurjeet Singh : +91 98880 62718

Call Center Solutions

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Call center solutions include any call center provider or kind of software that supports call centers. Call center solutions are products and services that solve the specific need for customer relationship management (CRM).

A call center solution is often a total package like unified communications or turnkey product, such as an outsourced call center. Software that runs a complete call center system would also be considered a call center solution.

Types of Call Center Solutions


Inbound call center solutions help companies with incoming calls, such as with customer service and tech support. Outbound call center solutions can help businesses with market research, lead generation and customer outreach. Telemarketers use outbound call centers.

Blended call center solutions can provide both inbound and outbound services.

A virtual call center solution can work for businesses with no central location or several locations. Call center agents can work from home.

Offshore call centers are outsourced companies outside of the business’ home country. Offshore call center solutions can save businesses on overhead expenses.

Web-enabled call center solutions allow customers to call in from their computers. A web-enabled call center is usually a contact center.

In many cases, call center solutions are actually contact centers with the ability to communicate in a wider variety than voice. See call center vs contact center.

Call Center Software

Software is always a part of modern call center solutions. A call center needs a server to run its phone system (such as a PBX) and software to integrate telephony and computer systems.
Not all software is all-in-one. For example, a company wishing to integrate CRM with its call center technology requires additional software.

Call center software can have useful features such as phone controls from the computer and screen popping a client’s profile/information.

Many call center services come with their own, proprietary software. When a company runs its own call center or contact center there are software solutions, both proprietary and open source, for the below functions:


Call Center Solutions

3G Callnet is the one of call center solutions providers and software. ...

Business PBX

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PBX systems were designed for businesses. The private branch exchange system serves organizations, offices, and businesses by establishing an internal network of telephones. Business PBX is a redundant term in this sense, though today residential VoIP services may be based on PBX systems.

In the 1990s innovations led to new PBX technology, namely VoIP and hosted PBX. VoIP and hosted PBX allowed smaller businesses to take advantage of complex PBX functions and features that were not practical before. In the past, PBX systems were incredibly complicated and expensive to setup and maintain, making it difficult for small businesses to use.

There are other business telecommunications systems such as Centrex, but business PBX systems have become versatile enough to offer equal or greater features for similar cost.

Small Businesses

Business PBX is available for SOHO, SMB, etc. Business PBX providers have a variety of options and scalable systems.

Hosted PBX

Hosted business PBX is housed off-site with the provider. No hardware beyond phones and an Internet router are required. A hosted PBX is charged per user, which is cheaper for smaller businesses than installing expensive equipment to run a PBX on-site.

Virtual PBX

Virtual PBX for business is a stripped down hosted PBX, and is usually designed for very small enterprises. Outbound calling is not usually included.

Larger Businesses

IP PBX

On-premises IP PBX is useful for large businesses, where paying for each user is impractical. Large business PBX systems need far more maintenance, but an IP PBX can save money by using software to replace PSTN equipment. An alternative is managed PBX, which is still on-site but maintained by the service provider.

3G Callnet

91- 98880-62718 or info@3gcallnet.com

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Business Phone Systems

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Business phone systems are an essential part of any business operation. Businesses can opt for a traditional phone system or a business VoIP system. Typically, employing a VoIP in a business phone system has the potential to save businesses up to 70% of their monthly phone bill when compared to a traditional phone system. Compare business phone system service plans, features, pricing, and customer reviews below.

Business Phone Systems by Country


3G Callnet

91- 98880-62718 or info@3gcallnet.com

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IVR

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What Is IVR?


IVR, or interactive voice response, is a what allows phone systems to process touch tones or voice waves during a telephone call. IVR technology is responsible for the menus people hear and respond to when they call up a company or business and hear the words: "press 1 for sales, press 2 for marketing, press 0 to speak to the operator," for example. IVR systems can be fully customized to play back dynamic audio, or pre-recorded menu options.

IVR is not necessarily related to VOIP, however, a VOIP IVR is. Most VOIP IVR systems or software support SIP based VOIP, but Skype IVR also support non-standard based Skype service.

Computer Telephony Component

IVR is an automated computer telephony integration CTI system which allows providers to create complex menus which the caller can navigate by using touch-tone key-presses or via spoken commands. IVR systems can be used as a Voice portal to access remote information such as bus scheduling where the caller can select the route for which they require information, or for billing or customer service systems which allow the caller to enter information such as their account number or credit card details without the need for operator assistance.

IVR and ACD Integration

IVR solutions are often integrated with an ACD, which routes incoming phone calls to agent work groups. This integration can be both a front end and back operation.

  • Most typically, an ACD system can route callers to an IVR program based upon DNIS or other parameters such as time of day or day of the week.
  • A smart IVR can transfer callers back to an ACD system to route the call to the next available agent within an agent hunt group.

One important task of an integrated IVR and ACD is to display Screen Pop information from the caller on the agent's workstation so that the agent has caller information readily available without the need to prompt the caller again.

IVR and Voice Broadcasting

IVR applications are typically associated with inbound calling programs. However, IVR technology can be applied to outbound calling campaigns and are most commonly used with Voice Broadcasting and touchtone responses. Examples of the application of this technology include the option to speak with an operator, opt out of a calling campaign, or taking an outbound survey.

Here is an example of IVR implementation in Voice broadcasting

Graphical Design Tool for IVR Applications

Recent IVR systems usually use high level scripting languages such as VoiceXML, an open standard for interactive voice response systems. For most users who lack technical training, developing an IVR system using scripting language, even high level language, are not feasible. The good news is there are design tools that are based on graphical user interface for the techies and none-techies alike. By using a GUI tool, a user can simply drag-and-drop components and create and deploy an IVR system in minutes. The whole design is a call flow diagram, much like a voicemail system user manual.

See Also (Vendor Information)

IVR Information


  • CCXML standard markup language for IVR / call control applications
  • IVR System Simulation Model - estimates resources required for an inbound calling campaign.
  • IVRS World - Blog about IVR

Virtual PBX providers

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Virtual BPX is a service offering functionality of a PBX without the need to install switching equipment at the customer location. Only VOIP phones need to be installed at the customer site. This makes supporting distributed workers very easy as each requires only and internet connection and a VOIP phone. A business virtual PBX phone system can reduce your monthly phone bill significantly compared to a traditional business phone system.

What Is a Virtual PBX?

A PBX, short for private branch exchange, is a telephone system with the capacity to switch calls between different users on local lines while still relying on the same number of external phone lines. With a virtual PBX system, the system is posted and software based without all of the traditional hardware of a physical PBX.

Virtual PBX Primary Function

A virtual PBX is used by businesses in a variety of ways. Primarily, companies utilize the system as an auto-attendant to establish preset call transfer options without needing an operator or receptionist. This type of system is capable of performing tasks that include auto-attendant settings, time of day or day of week functions, or even find or follow me sequences.

One of the most important functions of a virtual PBX system for companies is the software’s ability to establish pre-determined sequences. For example, in some businesses it may be appropriate for the phone to ring to a receptionist or operator first. If the receptionist does not answer in a predetermined number of rings, however, the call is then transferred to a secretary. Again, if the call is unanswered, it can be set to forward to an assistant. Left unanswered by these two individuals, the call can be forward to a manager or even an owner. These call settings are completely customizable and can be based on any number of sequences.

This type of software is also able to facilitate customized answering menus and sub-menus. The system can be modified to establish appropriate dial prompts leading to a number of different departments within the business, including different sequences on different days. PBXs are used by the vast majority of businesses to establish advanced call routing services.

Virtual PBX Cost

A virtual PBX is a complex service; however, that doesn’t mean that it is expensive. In fact, a virtual PBX is typically more cost effective than a physical PBX. The main reason that a virtual system saves on cost is that it does not require the same investment in capital to establish or set-up the call system. Because a virtual PBX is a software or hosted system, it is typically an operational cost, or a low monthly payment rather than a large upfront investment. This aspect alone generally makes a virtual or hosted PBX a less expensive, or at least more cost effective, option compared to the traditional PBX.

Virtual PBX Benefits

Aside from offering an effective call system, a virtual PBX presents a number of added benefits for users. As a whole, virtual PBXs lead the industry in business communication choices. This type of system seamlessly integrates the call management system with any existing phones to affordably and effectively deliver better call management. These systems also feature several innovative call features to meet the needs of any business. These systems offer various functions including call routing, follow and find me call forwarding, voicemail notifications, call recording, and more.

The benefits aren’t limited to the features, though. Virtual PBXs offer virtually limitless application for one or hundreds and even thousands of employees. Likewise, there is not hardware to maintain or constantly upgrade. Considering that benefit, the system is also more cost effective and generally provides for a variety of flexible billing options. The limited maintenance, web-based management, and hassle-free setup alone are often enough to convince a company to switch over to this option.

PBXs are an important tool in any business that makes and receives nearly any volume of calls. A virtual PBX can dramatically increase the efficiency of a business by effectively managing calls. This efficiency combined with the other numerous benefits of a virtual PBX can virtually transfer the communication capabilities of any company.

List of Virtual PBX Providers




OnePipe

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

Elastix miniUCS appliance

The Elastix miniUCS is a compact all-in-one IP PBX featuring :

  • 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

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.

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.

Add voice recording and IVR to your existing PBX

Want to impress your callers with a professional IVR ? Or think of keeping audio files for conversation between your agent and customer ? We could enhance your PBX with these features by VoIP technology. Please contact us for quote.

Cloud/Hosted PBX and IVR service

Looking for a hosted PBX service? We provide elastix/asterisk PBX hosting environment with DID SIP trunks and support of remote extensions. Pay-as-go-you. ...

VOIP Software

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Billing

See Open Source Billing Systems& VOIP Billing

Call center monitoring


Computer Telephony Integration (CTI)

SIP Phone Service Providers in Canada

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This is a list of SIP Phone Service Providers in Canada. Also check out VoIP Providers Canada to see VoIP service providers in Canada.

ClearSwitch - ClearSwitch provides SIP trunks to major corporations in North America. Pay-as-you-go payment structure, DID's available in over 7000 ratecenters, guaranteed Premium US/Canada routing, Premium international routes, multiple North American servers, crystal clear connections. Superior solution for softphone, Fring, etc.


  • nurango - SIP Trunking for Canadian and U.S. based Businesses. nurango provides Encrypted and Geo-Redundant VoIP Solutions including; SIP Trunks, Per-minute VoIP and DID's in Canada, the United States and over 60 Countries. Also check out our Wholesale and Call Center SIP Termination. Located in Montreal Canada, serving globally! Follow us @nurangotel

Net2Phone Net2Phone offers high quality, low cost SIP Trunking and Business VoIP Solutions - no charge per channel, no contracts, pay only for the minutes you use.

Hosted Business Telephone Systems by Taridium - Taridium Canada offers fully featured hosted business telephone systems for small, medium and large enterprises. Taridium also offers Hosted PBX Software for Service Providers.


VoIP Innovations VoIP Innovations is a Wholesale VoIP Services Provider, interconnected with the industry's top telecommunications carriers. VoIP Innovations provides high quality and low cost VoIP services to ITSPs and resellers through our unique industry leading BackOffice, Titanium III. Titanium III provides you with online access to the industry's largest footprint and number warehouse.

Twitter VoIPUser Directory

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This is a page that can contain entries of Twitter users and their business or VoIP interests.

Please insert into correct alpha place and please leave things clean enough to be helpful to people on Twitter who want to find VoIP Users

Voip-info.org Twitter: @Voip_info

VOIP sites

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Links to other Non-Commercial VoIP sites

This page is intended for informative and/or community related websites only. Postings of websites from manufacturers, VoIP networks, consultancy, free calling, calling cards and other commercial websites will be removed. See also Asterisk news and blogs. Links out of URL order are also subject to deletion. (hint, make your display tag the same as your domain name)

A

VOIP Service Providers Residential

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Asterisk Professionals

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The list of Certified Asterisk professionals around the globe.

United Arab Emirates / Dubai


DVCOM logo.png

DVCOM Technology is the only Digium Authorized Training Partner in the Middle East.

DVCOM Technology is the Authorized Distributor of Digium Asterisk in MENA region.

With a 360 degree solutions and product portfolio, DVCOM has become a prominent name to compete with proprietary IP based communications products.

DVCOM provides a complete range of professional products and services and assists organizations with the assessment, design, and implementation of professional IP Communications solutions including Unified Communications(UC) and Converged Communications (CC) for ensuring high availability, reliability, performance, scalability, and manageability within their intranet environments.

DVCOM is recognized in the industry for its experience and expertise in developing enterprise systems essential for any organization to ensure maximum efficiency which are delivered through DVCOM’s highly experienced team of certified and experienced professionals and the extensive partner/channel network.

Visit us at - http://www.datavoiz.com
Location Map - http://www.datavoiz.com/images/dvcom_map.JPG
JAFZA - Dubai
Tel: +971 4 88 733 70
Fax: +971 4 88 730 20
Email : mailto:sales@datavoiz.com


Peru


  • Celular : 956010103 - #956010103
  • Email : ventas@asternicperu.com
  • Expertos en Vicidial
  • Certificados dCAP
  • certificados dCAA
  • Soporte onlie - remoto
  • IPPBX
  • IVR
  • Asterisk dCAP
  • Support, Solutions and Services.

Australia


Darley Stephen

  • MOBILE 0401597387
  • E-mail: darsdsd@gmail.com
  • all Asterisk / digium installation & support, small / Large scale deployments, Asterisk configuration, Multiple location deployments
  • Predictive Dialer
  • Call Center deployment
  • Industry based telephony installation
  • Global PBX setup
  • IVR
  • Multi site Maintenance
  • E1/PRI/ISDN setup
  • red5 installation & config
  • flash phone customization

Asia (ASEAN) / Myanmar


Asterisk/FreePBX consultant
Asterisk based IP PBX professional service
      • The first Asterisk Consultancy in Myanmar ***

1. IVR
2. FXO/FXS
3. Dial plan
4. Conference Phone configuration
5. Voice Mail.
6. Android/iphone/Nokia Phones with native SIP support for Asterisk using wifi PBX
7. Linux Server Setup. ...

SIP training

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SIP Training Classes:


A VoIP and SIP Training Course: Hands ON classes TrainingCity Delivers two VoIP Hands ON classes, Course 784: VoIP Hands ON Workshop Part 1 and Course 2004: Advanced SIP Fundamentals Workshop. Public classes are scheduled in a variety of locations including Washington, DC, San Diego, California every month and both classes are available for onsite delivery.
SIP Training and SSCA® Certification Visit for Demos and Pricing and see why this is the school with 16500+ students and over 4200 certified SSCA® professionals. TIA Endorsed, Bicsi Credits and Recognition by USTelecom, CompTel, ITSPA, Avaya, Cisco, Mitel, Toshiba, Panasonic, NEC and Many many more!
Learn SIP by thoroughly studying the SIP protocol through a process of lecture and hands-on training. Learn what SIP is, how it works, and get a practical guide on how to use it. Training is offered at the customer’s site, online through our virtual platform, or at our training facility in Harrisburg, Pennsylvania.

New Software Releases

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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

customLogo.gif.png

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. ...

voip server monitoring

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Online voip server monitoring


Online voip server monitoring providers users SIP messages to see whether VOIP service is operational in VOIP servers.

  • Zoom Soft - We are providing VoIP billing Dedicated Server with cheapest price and 24/7/365 services. VoIP Soft Switch, VoIP Billing, Standard Server, Turbo Server, Premium Server, Pioneer Server all are available for any country.

  • SysNetx - We deliver fully monitored High-Availablity clusters & Failover solutions for VoIP carriers and Enterprise VoIP deployments. We deliver true solutions with excellent customer & technical support.

  • QualiStream - QualiStream's Monitoring service allows you to launch call scenarios at regular intervals, continuously or on demand, and indeed supervise the Quality of Experience of given services’. Benefit from relevant indicators & alerts of Quality of Experience from end-to-end.

  • VoIP Spear - global VoIP monitoring and testing service that monitors your VoIP quality 24x7x365. Free for personal use and very affordable for commercial use. http://www.voipspear.com

  • SIP NMS - SIPNMS is an online service that uses different sip messages (ie invite, options, sip simple message) to monitor voip nodes. Full functional demo account can be upgraded to Free (with limited functionality) or Paid account after the 30 day trial.

  • Dotcom-Monitor - SIP Monitoring is an online monitoring service that proactively monitors the ability of VoIP infrastructure components to establish and m aintain VoIP calls. It proactively monitors VoIP services using Session Initiation Protocol (SIP) the signaling protocol typically used for VoIP

  • Humbug Labs - Monitor and view real-time analytics, and benefit from fraud detection and alerting capabilities. Includes support for multiple PBXs.

  • StarTrinity SIP Tester - Freeware SIP and RTP monitoring tool with G.107 MOS/R-factor, RFC3550 global max jitter, realtime charts and reports, email alerts.

VoIP Origination

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Please add information to this page about VoIP Origination.

What is VoIP Call Origination?


One of the terms most often used when talking about VoIP communications is call origination, which basically has to do with how a telephone call starts and how it travels to the receiver once it does. The following will provide an in-depth look at what call origination is, how it works and what type of hardware is required.

What is Call Origination?

VoIP stands for "Voice Over Internet Protocol". This means that phone calls utilize a technology that allows the calls to be sent directly over internet networks, which is a much cheaper way to make calls. Call origination refers to the point where the call starts, which takes place over the PSTN telephone network and transferred to their destination through the internet. It's important to note that a phone call through VoIP starts between the initiation point and the destination point, which are referred to as the originator and terminator respectively. There are typically different types of call origination depending on the services provided by the ITSP involved. The only way for call origination to work is if the VoIP originator has a call termination arrangement with a VoIP terminator.

Required Hardware

The best aspect of VoIP services is that there are hardly any noteworthy hardware requirements. All that is necessary is a gateway, which essentially transfers phone calls from the internet and onto PSTN lines. Since a gateway must interact with both the internet and standard PSTN lines, there are two interfaces necessary for a gateway, including a telephony interface that takes digital and analog lines and an Ethernet interface as a connection between the gateway and the internet. It's important to understand that a digital line can support a large amount of calls at one time, which can range anywhere from 20 to 30 depending on the type of line that is chosen. In comparison to a digital line, an analog line can only support one phone call at a time.

How Call Origination Works

Call origination is a fairly simple process that is a bit more complicated to explain. In essence, the gatekeeper mentioned previously will receive the calls and requests from the dialer. When a user makes a connection to the dialer, the gatekeeper will ask a Radius server to check if the user has input the correct password and username. The Radius server will then answer the gatekeeper with a yes or a no. If a yes answer is received from the Radius server, the user the has the ability to make a phone call from the origination point. Once the number has been placed, the gatekeeper once again receives a request for the phone call to be made. It is at this point where the gatekeeper will interact with the Radius server again to see if the user in question has enough money to make the phone call. The Radius server will then connect to the billing server to ascertain how much money has been provided by the user for this specific call, in order to nail down how long the call can last before being cut off.

The billing server will then take a look at the location at which the call is originating from in order to measure the current rate of pay, which all depends on whether the call is being made in off peak or peak hours. Once this has been determined, the relevant information will then be sent back to the Radius server. Once the Radius server has received this information, it will be sent back to the gatekeeper. The user is then provided with the IP address for the gateway at the destination to which the call is being placed. This gateway will send the call to the final destination. If no other issue arises during this time, the call can take place and the two users can talk for however long the duration of the call is. If the call takes place once the maximum time allotment has been reached for the call, it will be disconnected and more money will need to be provided in order to make another call and continue the conversation. When the call has come to a conclusion, whether by the callers themselves or because they were disconnected, the initial user will have the total price of the call deducted from their payment source by the billing server.

Types of VoIP services

There are two basic types of VoIP services, including PC to phone and phone to phone. The provider for the PC to phone service will place a dialer on the internet that the user of the service can download and install. This dialer allows for an account to be created with any type of payment source that is allowed by the service provider. Once this is done, the user can make and receive calls. ...

CNAM

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CNAM is an acronym which stands for Caller ID Name.

When phone calls are made, there are usually two user-facing identifiable pieces of information: a phone number and a Caller ID Name (usually a 15-character string). CNAM can be used to display the calling party's name alongside the phone number, to help users easily identify a caller.

There are numerous CNAM lookup services which allow you to pay a small fee to lookup the CNAM of a specified caller (by phone number).

CNAM Lookup Services List:


http://www.bulkcnam.com/ Cost: Only $0.005 per query for carriers or $0.009 for hobbyists! No catch, guaranteed with easy paypal integration. Sign Up for a FREE Account and we will credit you 30 FREE CNAM queries to try http://www.bulkcnam.com/.

http://www.calleridservice.com No monthly fees or account minimums and 20 free queries to test our service when you open an account ( instant setup ). Simple HTTP API or Fast AGI that can be placed in your Asterisk dial plan. Also native support for Switchvox PBX systems. Results are never cached so you get up to the minute real-time results. Retail prices are $.006 per query and bulk pricing is available with a volume commitment of at least 25,000 queries per month. Free support and installation assistance is available.

www.callwithus.com offers both CNAM ($0.006) and LRN ($0.0003) look ups. No minimums and monthly charges. Simple HTTP API, easy to integrate to Asterisk dial plan.

CID(name) Professional CNAM (Caller name) delivery

  • EVERY LOOKUP IS LIVE FROM THE SS7 (direct from the carrier owning the number)
  • NO CACHING... EVER!
  • NO 3rd party data sources
  • NO monthly fees
  • NEVER pay full price for unavailable results
  • Carrier grade, multi-redundant platform
  • Simple to integrate HTTP API
  • 99.7% caller id name accuracy
  • Lightning fast query responses (under 500ms)
  • Volume pricing as low as $0.002 per query
  • Try before you buy, 100 free dips with every new account
  • You choose the output, TEXT/JSON/XML
  • Track sub-accounts
  • Easy integration with Freeswitch, Asterisk, OpenSIPS, and other open source voip platforms
  • Easy access and daily downloads to your account activity
  • Thousands of happy customers

Get CARRIER GRADE CNAM at http://www.cidname.com


www.cnam.info offers both CNAM and a pseudo-CNAM service at a fraction of the cost. Integration with asterisk is as easy as downloading the AGI and adding a single line to your dial plan.

http://www.data24-7.com CNAM service for $12 per month (membership fee) and $0.005 per query ($0.004 for over 500,000 transactions per month). Simple HTTPS/XML-based API with examples, plus batch file upload and manual entry available via website. Asterisk support. We also support SIP protocol so your switch/dialer can communicate with our service directly. Members have full access to Data24-7's other services, such as carrier lookups and email-to-SMS gateway address lookups. Free trial!

http://www.multitel.net/ We have multiple SS7 interconnects and are able to provide you with some of the most accurate and up to date results. Pricing is $0.004 per query for our Free tier and it goes to $0.003USD and $0.0025USD for our Standard and Professional tiers. We do provide free testing - limited to 60 queries per hour. No commitment , no hassle. ...
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