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VoIP Hardware

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This page lists information about VoIP hardware and VoIP hardware products. For phones and hardware to use with Asterisk, including VoIP phones (both hard and soft phones) and Analog Telephone Adapters, see Asterisk phones.

PSTN Interface cards (analog, GSM, ISDN-PRI and R2/MFC)


This section contains VoIP hardware for connecting analog or digital phone lines from the Public Switched Telephone Network to your Asterisk server. Please keep VoIP hardware providers in alphabetical order.

2-Day Direct

  • Cisco SPA303 3-line business-class IP phone; Connects directly to an Internet telephone service provider or to an IP private branch exchange (PBX)
  • Cisco SPA504G Full-featured 4-line business-class VoIP phone supporting Power over Ethernet (PoE)
  • Cisco SPA525G2 5-Line Business IP Phone with Enhanced Connectivity and Media for a New Level of Small Business User Experience; includes wifi and bluetooth connectivity

freePBX

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FreePBX is a full-featured PBX web application. If you've looked into Asterisk, you know that it doesn't come with any "built in" programming. You can't plug a phone into it and make it work without editing configuration files, writing dialplans, and various messing about.

FreePBX simplifies this by giving you pre-programmed functionality accessible by a user-friendly web interfaces that allows you to have a fully functional PBX pretty much straight away with no programming required. Some of the features that FreePBX supports out of the box are:

  • Unlimited number of Voicemail boxes
  • "Follow Me" functionality
  • Ring Groups with calls confirmation (so if, eg, a cellphone is out of range and diverts to voicemail, all the other phones keep ringing)
  • Unlimited number of Conferences (limited by available CPU power - about 300 simultaneous users in conferences on a P4 3ghz - 600 with a dual core!)
  • Paging and Intercom functionality for man SIP phones that support it.
  • Music on Hold (via MP3s, or streamed off the internet)
  • Call Queues
  • And many other features

FreePBX is built on the LAMPA™ stack (Linux, Apache, MySQL, PHP and Asterisk). It's a modular system, with click-to-install plugins downloadable over the internet from the online module repository.



FreePBX Features at a Glance:



  • Add or change extension and voicemail accounts in seconds
  • Native support of SIP, IAX, and ZAP clients (other endpoints are supported through custom extensions)
  • Supports all Asterisk supported trunk technologies
  • Reduce long distance costs with LCR
  • Route incoming calls based on time-of-day
  • Create interactive Digital Receptionist (IVR) menus
  • Design sophisticated call groups
  • Manage callers with Queues
  • Upload custom on-hold music (MOH)
  • Search company directory, based on first or last name
  • Detect and receive incoming faxes
  • Share administrative duties
  • Backup and Restore your system
  • Save audio recordings of calls
  • View call detail reporting with asterisk-stat
  • View extension and trunk status with Flash Operator Panel
  • View conversation recordings with Asterisk Recording Interface (ARI)

Project Sponsored by SANGOMA

Service Providers

FreePBX Services - AristoCraters offer a wide range of services for FreePBX Solution, including, setup and installation, configuration, customization, upgradation, troubleshooting, performance enhancement, and ongoing support to keep your FreePBX solution up and running.

VoIP / SMS / API - Unlimited Channels with measured rates and always free to test.

http://sipstation.com

http://www.asterisksiptrunking.us Asterisk SIP Trunking - US — Offers SIP Trunking for Asterisk. Over 500,000 DID's available in 9,500 rate centers. You can activate and setup service in minutes. TDM Enterprise quality. Fully qualified Asterisk consultants ready to assist you. Live customer service with 24/hr ticketing system. No per channel fees, we offer unlimited channels in with your SIP trunk. FREE API for your website. Use our API to leverage the power of our customer user portal on your own website. You can build your own back office admin panel with our API and also provide your customers the ability to order DIDs in REAL TIME along with setting up SIP Trunks. Automate everything and increase customer base.Our Asterisk SIP Trunks are not only for Asterisk. ...

VOIP Billing

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Hosted Billing Services (in Alphabetical Order).



  • 4PSA VoipNow fully featured, carrier-grade, multi-tenant edition for service providers and businesses, that can be installed on their chosen infrastructure or delivered as a UCaaS. VoipNow provides a fast, competitively priced go-to-market solution, from deployment and provisioning all the way to selling and billing.
  • Avatar Dialler Make your billing procedure easy with easy voip billing! avatar is offering you free voip billing software where you can also check your balance at any time.
  • Adore VoIP Billing Adore VoIP Billing Software comes with the enhanced functionality along with the architecture with class. It is fully compatible and gets integrated with all other VoIP related products. It is designed with all the present and future demands of booming telecom industry kept in the mind. The telecom industry is changing and developing with rapid speed and the VoIP products such as VoIP Billing comes as an excellent product in this time.
  • A2BILLING - VoIP Billing Solution / AAA / Class 5 Softswitch.
  • Aradial AAA for Billing Solutions
  • benotos offers free callshop billing system 4-level billing system: reseller-subreseller-callshop-customer, 2 different routes, nice easy to use interface, intelligent ratemanager, online payment, detailled reports, receipt printing with own logo, white labelled, use your own brand and domain name and much more features. About 9000 callshops around the world are using our excellent callshop billing solution already. Free signup - best rates on market - low payment amounts
  • BillCall - Telecom Resource Management for wholesale Voip Carriers Panamax’s Telecom Billing Solution BillCall provides solutions for End-User billing, Carrier Access Billing (CABS), CDR Mediation, Rating & Routing.
  • CloudAstrix SPE CloudAstrix SPE is such a VoIP Switch. Build on the world renowned WHMCS Billing Suite, the Soft-switch module brings all necessary functions to perform and provide a top class VoIP service.As a Carrier Neutral soft-switch, CloudAstrix has already proven to be a firm favourite among ISPs all over the world.
    • Note:CloudAstrix SPE Module works with FreeSwitch.
  • Cybercallshop ultimate callshop server Incredible advanced online callshop server 100% standalone able to handle many shop, simply the best software to get customer loyalty because it's many more than simple online booth billing.
  • CRM Sipit Enterprise CRM system with fully integrated pdf invoice billing, customer balance, export, import of priceplans. This is a fullblood CRM system with invoice capabilities. Compatible with Kamailio.
  • Cyneric Fully Integrated Billing Solution Platform. Compatible with: Cisco, Radius, Mera, SIP, SER, Asterisk, Quintum, SNOM, Audiocode.
  • DORETEL Communications, Inc. ...

HyperMedia Systems

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Hypermedia Systems


Hypermedia Systems develops and manufactures scalable, professional GSM Gateways and VoIP GSM Gateways for low to high volumes of traffic, starting from 8 ports, with up to 72 cellular channels/ 288 SIMs. Available interfaces: VoIP, E1/T1-PRI and Analog interfaces, as well as GSM, 3G UMTS, CDMA.

These carrier grade Voice Gateways provide opportunities for Alternative carriers, Call termination vendors, and Service providers, as well as for SMB and corporate enterprises.

Read our Testimonials page.


Visit Hypermedia's Website

We encourage you to visit and get to know it. Find many interesting new areas, like a new downloads page where you can find White Papers, Data Sheets, and Application Notes and more. See our latest addition: Clips & Presentaions.

We are proud to announce The Pinnacle of Quality: 2013 BID Award went to our distributor; HyperAccess Systems from Ivory Coast, who won the BID International Star for Leadership in Quality Award in Paris

Meet with us at WWC - Check our new Termination Systems!

NEW: Economical 320 SIMs Gateways320 SIM-based box for SMS/ VoIP Best performance ever!


Complete GSM/SMS device with three times more SIM storagecapacity and SIM Management features in one box.
Easy to deploy and quick to implement, this is the perfect all-in-one product solution which allows you to solve all your telephony needs with NO NEED for an additional SIM server!

• Enables to maximize SIM price plans usage and special dynamic tariffs.
• 10 SIM cards per port with individual settings that can be configured for EVERY SINGLE SIM card!

Ready to order? Be among the first 20 buyers and Get SIMGuard free

Professional SIM-based hosting server - NEW

Open your SMS service boundaries, with Global Long code 2 Way SMS!

HG-7000 SIM hosting server is an integrated platform designed for 2-way long code SMS messaging.

Unlike short codes (such as 7777, *2323) that can only handle local inbound SMS, long codes (ex. +33.678235.12345) allow inbound SMS messages from all over the world!

Integrating the HG-7000 server is the FASTEST WAY to start or expand worldwide Virtual Mobile Number (VMN) services of sending and receiving SMS messages between your users/customers databases and your email, website or database.

VOIP Providers Italy

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These are providers that offer voip services to Italy :

  • http://www.vohippo.com - Vohippo is one of most important provider that offer free call to landline destination in Italy and low-cost rates for mobile. It provides also free pbx features for company that need more extension or wholesale plans.

  • https://www.gloobobusiness.com - GlooboBusiness is one of the cheapest VoIP providers for small business and call centers in Italy. Is possible to create a customized package of minutes for every destination called having a discount from 5% to 25% on rate list and to activate a 30 days free trial DID VOIP number.

SS7

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Common Channel Signaling System No. 7 (aka SS7 or C7) is a global standard for telecommunications defined by the International Telecommunication Union (ITU) Telecommunication Standardization Sector (ITU-T). The standard defines the procedures and protocol by which network elements in the public switched telephone network PSTN exchange information over a digital signaling network to effect wireless (cellular) and wireline call setup, routing and control. The ITU definition of SS7 allows for national variants such as the American National Standards Institute (ANSI) and Bell Communications (Telcordia Technologies) standards used in North America and the European Telecommunications Standards Institute (ETSI) standard used in Europe.



Reference pages


SS7 Forums

SBO

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sbo multipath logo.jpg

What is SBO?

Synchronous Bandwidth Optimizer is generally known as SBO. It is a bandwidth optimization solution for VoIP call termination developed by Synchronous ICT, a world famous VoIP software provider. It is a pioneer Bandwidth Optimization Technology fully loaded with amazing features.

Why Bandwidth Saver is so important for VoIP?

You know, in VoIP termination, main expenditure is Bandwidth cost. Only for highly bandwidth cost many companies are struggling day by day. Minimizing Bandwidth consumption is the main purpose of SBO Solution, at the same time it helps to reduce business operation cost. As a perfect bandwidth saver SBO reduces more than 80% internet cost directly. However, the quality of service is never compromised. To run a quality VoIP business congestion free, telco grade bandwidth connection is required . Such type of bandwidth connection is very costly as well as extremely rare. So, it was almost impossible to run quality VoIP business basically for small and medium entrepreneurs. But, now things are different, SBO Multipath make its possible to run IP telephony business with low cost share internet connection by maintaining supreme service quality. It can combine multiple internet connections together and use all the available bandwidth creating a single connection.

How does SBO work?

Well, there are mainly two parts in SIP communication, one is payload and another is RTP header. For a SIP call with G.729 it consumes almost 31.5 kbps bandwidth. But, noticeable matter is that the payload size is only 8 kbps. Rest of the bandwidth is consumed by RTP and other headers. SBO works here. It has own proprietary VoIP protocol which can replace the RTP and only transmit payload size thus reducing bandwidth consumption.

Key Features of SBO:

  • The most important feature of SBO is, it can reduce bandwidth cost directly by 80% without degrading service quality.
  • SBO works behind any type of firewall and NAT. That means, it has anti-block feature.
  • Works with all commonly used codecs such as G.729, G.723 etc.
  • Multipath: SBO Multipath allows you to use multiple number of internet connections same time. These connections can be used simultaneously for your VoIP termination establishment which can balance load among available networks and it develop service quality expectedly.
  • Works with any type of internet connection i.e. ...

Asterisk click to call

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There are a number of different ways to implement "Click-to-call" functionality, which allows you to dial a phone number without actually dialing it. The convention for these scripts is to tell Asterisk to call your extension, wait for you to answer the call, and when you do, initiate a new call to the destination number.

Downsides of click-to-call include A) that you start to forget phone numbers that were once stored in memory, and B) when you're using foreign interfaces that don't provide you with the clickability, you often wonder how those around you live without it.

Click-To-Call FREE in your Website

Anveo ContactME widget is free Click To Call solution from Anveo. Stay in touch with the online community and keep your personal contact information private. Fully customizable;you can change look and feel of Anveo ContactME button; upload multiple pictures and music ringtones. Here is how to use Anveo ContactMe with your Asterisk: Click to Call from Anveo with Asterisk - HowTo
Click to call through Asterisk PBX Easiest Solution on the web for click to call with open source code
VoIP click to call solutions A wide range of web based click2call, callback and p2p solutions from Mizutech integrated with Asterisk or Voipswitch servers.


Yuboto ClicktoCall - Turn Key Solution - Turn visitors into customers

Click to Call by Yuboto
Create and add a fully functional and customizable Click to call button for your website in minutes.
Click to Call in online environments
Click to Call on printed materials

Superb voice quality, low calling rates. Register now for free and try the service with the test credits provided!


AsteriskC2D for iPhone

Rho
AsteriskC2D and AsteriskC2DPro allow you to route your calls via Asterisk or indeed any VoIP style PBX. AsteriskC2D is not a VoIP phone but a call initiator, and can be configured the route the call back to your iPhone or deskphone or maybe hotel phone!

Dialectic

http://www.jonn8.com/dialectic/
A small but fully functional click to dial app for OS/X Supports multiple dialers including mobile phone and Asterisk. Includes, call timer, notepad, log, and while you were away as well as extensive support for automation through applescript.

Asterisk Click To Call

Click To Call Solution
Asterisk click to talk software solution design, development & configuration as per custom requirements by Asterisk experts.

Click2Call

Anveo ContactME widget is free Click To Call solution from Anveo. Online visitors can call your business from the website and get the same IVR call experience as if they dial into your business line. ...

VoIP Wholesale

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Wholesale VoIP Market:


There is no doubt today that VoIP is taking over the telecom market, and every month increases penetration into services and industries. Competitive carriers are looking at the numerous ways to make money from this exploding technology, but there's a lingering question as to whether it is profitable to deliver VoIP in a wholesale model? Their customers, typically Service Providers, are looking for their ‘competitive advantage' into this ‘lowest price' race, leveraging within three key alternatives for packet telephony : “build” , “buy” or “rent”. Business aspect, there’s no need to invest tens of millions of dollars in wholesale VoIP to join in. Many Telecom Companies have done the work for you. They offer a complete, turnkey VoIP business service and equipment. Now you can start wholesale VoIP business with virtually no investment and yet reap great dividends.

Wholesale VoIP Resellers:


In today’s world, Service providers seeking to deliver VoIP to as wide a customer base as possible may find that becoming a wholesale VoIP reseller is the way to go. Wholesale VoIP may be sold to both other service providers and to enterprises or residential customers.

Reselling IP telephony as a wholesale VoIP company is becoming an increasingly popular business model. For many companies, becoming a wholesale VoIP provider hits the sweet spot between profit and market control. Any firm with a well-established customer base is a good candidate for reselling wholesale VoIP.
Becoming a wholesale VoIP reseller is not a decision that should be taken lightly. It does, however, offer the potential of being very lucrative if done right.

Wholesale Consumer Demand:


An important characteristic of the industry is the complex segmentation of consumer demand and rapid change in the characteristics that are being demanded, both at the end customer and in the intermediate ones (wholesale customers).
Demand coming from ‘packed customers'? will be significantly different of the conventional telecommunications one, were telephony was the unique service to provide and differentiation was based on tariff-distance paradigm, being today's service offerings closer to data applications rather than telephony. Voice communication (and not old POT telephony) becomes the common feature into several communications applications and devices, but not the unique one.
Messaging, conference, collaboration, web contact centres, etc … requires a common communication format between parties, which is voice, implemented through VoIP technologies. Heterogeneous and rapidly changing customer demands and products are important dynamic influences on the evolving structure of the telecom industry, resulting into a new value-chain.
Telecommunication markets evolution will be driven by ‘packed customers' demand rather than networks, technology or finance, changing many decades rules into this industry.

Finance in Telecommunication Industry:

Finance institutions had been influencing Telecomm Industry since the beginning, due the business itself was characterized by huge investments, big market shares and bigger capitalization, influencing in many cases top management, who addressed their strategy towards ‘stock' opportunities rather long term and solid business models. WorldCom crash has been an example of this ‘financial market' pressure and wrong business management.
Today, the networks has been deployed. New scenario in Telecoms enable new players to deploy services over broadband without proprietary network and this new generation business will not be anymore capital intensive, let's say these will be innovation intensive.

U.S. VoIP Market:

The US market for VoIP advanced dramatically in 2006-2007, adding 3.8 million VoIP households in 2006, reports In-Stat: As a result, wholesale VoIP revenues grows quickly, as MSOs, Skype, and a myriad of new entrants most lacking network facilities enter the market and drive demand for telephony features and applications, the high-tech market research firm says.
As retail VoIP expands, wholesale VoIP will accelerate quickly, says Bryan Van Dussen, In-Stat analyst. The largest segment remains international VoIP, but we expect the market for local services to surge from 12% of all revenues to 27% by 2010.
Recent research by In-Stat found the following:

  • Consumer VoIP adoption will drive wholesale VoIP revenues to $3.8 billion by 2010 from $1.1 billion in 2006.
  • In-Stat finds small businesses are driving the growth of hosted services in the U.S. Hosted VoIP seats in the U.S. ...

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

+1800-385-4895 Outlook Mail Helpdesk

I8OO-385-4895 Outlook Mail Support Number

Yeastar

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Easy Communication!


Yeastar specializes in the developing and manufacturing IP-PBX products and is committed to the distribution of new generation technology products in the field of enterprises' communications. In the mean time, Yeastar provides the cost- efficient solutions for ITSP to develop the enterprises ultimate purchase market. With professional and high quality team, Yeastar designs products to worldwide applications and maintains the long-term stability of products to greatly benefit users. Yeastar team gave an insight into users' requirements and ready to listen to their new demands for implementing in the product design and services. Yeastar welcomes the cooperation from various kinds of companies and will sincerely treat them to create the multi-win situation together. Yeastar is working hard to make better office communication experience for you.

Team
With professional and high quality team, many of us are seasoned experts in telecom network and VoIP/PBX. A close-knit organization that we are, we aspire, work and achieve together. At Yeastar, we have built a tradition of teamwork. Innovation being our chief motivation, as a Yeastar, every one strives to come up with newer, more stable software and hardware products suitable for the users.

Our Products:

1. MyPBX - IP PBX for SMBs

Yeastar - MyPBX

The new compact, feature rich PBX for every-day use.

MyPBX is a standalone embedded hybrid PBX for small businesses and remote branch offices of larger organizations (2-500 users per site). MyPBX also offers a hybrid solution (a combination of VoIP applications using legacy telecom equipment) alternative for enterprises who are not yet ready to migrate to a complete VoIP solution.

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VoIP Providers USA

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

VoIP Providers in the USA


  • .e4 SIP/PBX/API/SMS - .e4's hosted PBX and SIP products lead the way. Check us out for a 1 month trial.

  • ALTOTELECOMCall Center VoIP provider - AltoTelecom is VoIP company that provides SIP Trunking services for Call Centers, hotels, small and large business using a PBX or an Asterisk based predictive dialer such as Vicidial or Goautodial for telemarketing sales, wholesale rates to USA, Canada and UK www.altotelecom.com CHAT Support Available

  • 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.
  • Vox Connect - Cloud based phone systems, cloud hosted PBX, SIP trunks Vox Connect is a cloud-based phone systems provider serving Chicago’s home, small and medium-sized businesses. We provide cloud hosted phone systems, hosted PBX ( also known as cloud based PBX ), domestic and international calling services, local/toll free numbers for USA and Canada.Our high-quality cost-effective telecom services are custom-designed to fit your business – and your budget.

  • Zaplee - Seamlessly integrate with Skype, GoogleTalk, VoIP, SIP, landline and mobile phones to make Zaplee fit into your existing framework easily. ...

VOIP Service Providers Business

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Business VoIP Providers - Compare and Choose a Business VoIP Provider

Quality business VoIP providers today offer a wide variety of feature packages, services and prices. Selecting the ideal provider and service options will depend on your type and size of business, features needed and projected volume of usage. Even when working with top-tier providers, your basic monthly service charges per line may begin at rates as low as $20. Before choosing your VoIP provider, it is essential to first determine your company's precise telecommunications needs to enable timely and cost-efficient initiation of your service. By consulting your chosen Voice over IP service team and seeking their expert advice in advance, you can be prepared to take the following steps to facilitate the smooth, productive startup of your services:

  • Evaluate Your Internet Connection. - Determine the strength and capacity of your Internet connection and bandwidth. You need to ensure that your system has adequate speed to best accommodate your new VoIP installation for top quality service.
  • Assess Your Company Budget and Needs. - With knowledge of your company's current budget and VoIP needs, you can more easily select the service provider and feature options that meet your requirements.
  • Determine Your Equipment Needs. - Evaluate your current and near future VoIP equipment needs. Phones can be purchased from around $50 to $500 or more. Once you decide which feature options are immediate requirements and which ones can be added later as needed, you are ready to choose your service provider.
  • Compare VoIP Providers. - By comparing VoIP company service options, advanced features and equipment along with user and industry reviews, you can best make a wise decision, selecting the ideal VoIP provider for your enterprise.

Important Information to Request from Any Potential VoIP Provider

Before signing a service contract with any business VoIP provider, be sure to request basic service information and practices in writing. You need to be certain of such details as startup costs and monthly fees, any limitations and costs on portable phone numbers and exactly which features are included in the service package you select. You also need to know if international calling is included, charges for adding extra features and the extent of customer care and technical services provided. Also important are such issues as whether your provider offers a money back guarantee and if there are any cancellation fees. It is also helpful to determine prior to signing up for VoIP services if there are any hidden fees assessed by your chosen provider.

Take Full Control and Advantage of Your VoIP System

Once your new business VoIP system and service are in place, you and your staff members will have full-control capabilities for use of your business communications system. Your service provider will ensure connection with your online portal for customizing your telecomm options. These modern digital portals are user-friendly, enabling feature changes and additions to be made for immediate availability. You and your staff can make decisions and changes in real-time that work for you right in the moment.

You can manage your call settings remotely, directing calls to voicemail or having them transferred to another number or extension. You can also make exceptions to any chosen setting in your phone system. For example, if you are expecting an important business call and want to take that call, but hold all other calls for a few hours, you can set your phone to direct only the designated call to ring on your extension. This system allows and encourages you to take complete control of your telecommunications systems and settings so that the service works for your best interests and immediate needs at all times.

Major Business Benefits and Advantages of Installing VoIP

With an excellent quality VoIP system installed and running well in your company offices to provide remote access for you and your employees, you can work much more efficiently, achieving more in less time. You will enjoy the many benefits of knowing that you can leave the responsibility of your advanced office telecommunications system operations to your VoIP provider while you handle other important business matters. Other major benefits and advantages of your new business VoIP system enable you to accomplish the following:

  • Schedule Your Own Business Hours. ...

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/idspecial.php NEW SPECIAL PRICING; 0.00247 per transaction.

  • $0.00247 per lookup - that's less than 1/4 penny each!
  • Futher discounts for larger volumes
  • Easy-to-use RESTful API
  • SIP / HTTPS support
  • Asterisk integration tools available.
  • Free trial available
  • Real-time (never cached) data
  • NO contracts necessary
  • NO minimum volume commitments necessary
  • Includes access to our other services such as carrier lookup, phone append, etc.
  • First-rate customer support!

For more information, call us at 877-805-3282, or http://www.data24-7.com/idspecial.php. ...

Asterisk Canadian Users

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Asterisk Links for Canadian Users


Hardware Resellers


Digium Hardware


Phone Numbers

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You can call the following VoIP numbers for free. From a softphone, just copy&paste the full SIP address (sip:user@domain.com).
From a hardware SIP adapter, copy & paste the full SIP address into your speed dial web page. All of these numbers worked in July 2011.

Dialing through PSTN


  • TheTestCall +1 631-791-8378 / +1 416-342-9562 record/playback, dtmf tests, realtime echo, MOH, audio bridge, caller-id readback and more

Dialing through SIP URI (all destinations listed working as of Jan 1, 2013)


  • Free 411 sip:411@ideasip.com +1-800-FREE411 US free directory assistance

  • Mouselike.org (UK) Echo and audio quality tests sip:904@mouselike.org or PSTN:+441483604781 - Allows test / connection from anywhere.

  • Telephreak sip:telephreak@voip.telephreak.org - The Telephreak's Free Voicemail/Conference system


  • UCLA sip:13108254321@ucla.edu (no G.729)

  • QXIP Music on Hold sip:9999@qxip.net (Curtesy of QUASARMUSIQ)

  • QXIP Streaming sip:9901@qxip.net (Hypemachine RSS Feeds)

  • QXIP Streaming sip:9902@qxip.net (Misc. Web Radio)

  • QXIP Streaming sip:pirateradio@qxip.net (Pirate Party C.C. Radio Stream

Internet Connection Speed Test

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For VoIP to work correctly, you must have a strong and consistent Internet connection.

The quality of VoIP calls depends on the speed of your internet connection. The faster your Internet connection is, the better your calls will sound.

Our internet speed test is specially designed to test your system for VoIP phone calls. It checks your Internet connection's quality and speed and provides you with an easy-to-understand report afterward. You can also submit your test to a testing company to further diagnose any problems you may detect with your Internet connection.

Our speed test is currently undergoing maintenance. In the mean time, please use the SourceForge HTML5 speed test.

To start the test, you must click on the button on the test that says "Click to start test."

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See also

Transcoding Services

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Transcoding services

Companies dedicated to provide transcoding to those who don't have the hardware or licenses for transcondig.

R&R Managed Telecom Services

  • R&R - provides transcoding services using dedicated DSPs that are known to have higher reliability and speech quality compare to implementations using generic CPUs. R&R transcoding service is fully licensed and R&R indemnifies it's customers from license violation litigation. (Most codecs like G.729 and G.723 are protected by copyrights and must be licensed).

R&R transcoding service can be delivered as stand-alone application, or can be bundled with a completed hosted SBC. We have resources in US and Europe to transcode over 100,000 simultaneous sessions.

Supported codecs with any-to-any transcoding/transrating.
- G.729
- G.711
- G.723
- G.726
- G.722
- ILBC
- GSM
- Speex
- OPUS

Easy to start with R&R. Monthly terms.
www.r-rtele.com
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