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New Software Releases

August 25, 2012, 4:07 am
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This page is to inform on various VoIP related software releases.

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

Page Contents


August 2012


July 2012

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VOIP Service Providers

August 25, 2012, 6:48 am
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For a list of VOIP to PSTN service providers, indexed by country, please see:


PLEASE DO NOT ADD NEW CATEGORIES HERE.

VOIP provider services, exchanges and other business deals belong under VOIP Service Providers B2B

Please keep your entry in ALPHABETICAL ORDER in relation to the other entries in your section.
If you add a new entry, including an 'added on dd/mmm/yy' would make it easier to notice.

Miscellaneous VOIP related services, including peer-to-peer services, are listed below.

Peer to Peer Service


  • SWISS VOIP Service Provider - HI Ring Someone - The worlds’ first virtually free VOIP platform to offer free and ultra-low cost international phone services with no contract and no monthly fee.Whether you are sitting down for a morning espresso in Hong Kong, a lunch date in Zurich or even dinner in Los Angeles, Hi Ring Someone offers free calling; anytime, anywhere.
  • VOIP Service Provider in India - Spectranet - leading Internet Service Provider Company in India offering VOIP Services in India, International Calling Cards, Prepaid VOIP Services, Business VOIP Services and all other affordable VOIP solutions in India
  • iVOIPE VOIP Service Provider in USA iVoipe is the leading next generation Internet Telephony Service Provider (ITSP) providing mobile and fixed services that dramatically reduce the cost of roaming and international calling. ...
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VOIP Service Providers Residential

August 25, 2012, 6:54 am
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Sip Trunking Providers

August 25, 2012, 4:32 pm
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This page is a list of SIP trunking providers around the world. Please keep this list in alphabetical order. SIP providers looking to add their services can do so in the list below.

Country specifici pages:

http://www.1comms.co.uk VoIP provider for UK Businesses. Sip trunk services, UK DDI's and Asterisk Support for business users and resellers. White Label fully itemised per second billing.

1Pipe Telecom | OnePipe is a CLEC providing SIP Trunking, Hosted PBX, IP and other services, we provide services on our own network and gateways.

Alcazar Networks - Wholesale Services Over 3,100 DID rate centers. Per minute pricing as low as $0.0005/minute. Per channel pricing as low as $2.00/channel. DIDs as low as $0.10/each. A-Z termination. Over 1,200,000 numbers in DID inventory.

Amivox free your phone - Lower your communication cost VoIP provider for both consumers and businesses. Offer's free SIP account. Prepaid and very good rates for network termination with premium quality ( Amivox-Out) . Support for iPhone, Android and Blackberry. Shared balance for multiple users. Calling Amivox to Amivox is free - Sign up for free and try out the service.

Anveo offers phone numbers from over 48 countries with instant activation. Anveo's Voice 2.0 Communication and Collaboration Suite with powerful Visual Call Flow technology allows you to visually configure call handling and call termination options for your phone number. Anveo provides FREE SIP trunking and it is one of many termination options available.

BellVoz offers International and Domestic Long Distance Services with VoIP technology, helping business and consumers to reduce monthly telephony expenses.

Best VoIP USA BestVoIPUSA.com offers SIP trunking to private and commercial operators of Asterisk PBX switches. BestVoIPUSA.com also provides all of the well known features of SIP even if you are connecting directly to SIP compliant devices, handsets or servers.

Box Internet Services offers SIP trunking to private and commercial operators of Asterisk PBX switches. Boxis.net also provides all of the well known features of SIP even if you are connecting directly to SIP compliant devices or servers.

Brisnorth Communications Australia Brisnorth.com.au provides SIP trunks, VoIP and SIP Server hardware to Businesses Australia-wide. Carrier-Grade reliable SIP/VoIP services at very cost-effective rates. We can work with your current hardware/phones or upgrade you. We have Plans to suit all budgets and sizes of Business. Contracts and Bundles are optional (Customers are free to go un-contacted and un-bundled) email sales@brisnorth.com.au or call 07 3623 0800

BroadTone Networks
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VoIP Wholesale

August 25, 2012, 4:33 pm
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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 has done the work for you, and offers 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. ...
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SIP

August 25, 2012, 9:04 pm
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SIP, the session initiation protocol, is the IETF protocol for VOIP and other text and multimedia sessions, like instant messaging, video, online games and other services.

Abstract from the RFC 3261 (formatted_and_explained version) - SIP: Session Initiation Protocol

This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.

SIP invitations used to create sessions carry session descriptions that allow participants to agree on a set of compatible media types. SIP makes use of elements called proxy servers to help route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies, and provide features to users. SIP also provides a registration function that allows users to upload their current locations for use by proxy servers. SIP runs on top of several different transport protocols.


SIP is very much like HTTP, the Web protocol, or SMTP. Messages consist of headers and a message body. SIP message bodies for phone calls are defined in SDP -the session description protocol.

  • SIP is a text-based protocol that uses UTF-8 encoding
  • SIP uses port 5060 both for UDP and TCP. SIP may use other transports

SIP offers all potentialities of the common Internet Telephony features like:
  • call or media transfer
  • call conference
  • call hold

Since SIP is a flexible protocol, it is possible to add more features and keep downward interoperability.

SIP also does suffer from NAT or firewall restrictions. (Refer to NAT and VOIP)

SIP can be regarded as the enabler protocol for telephony and voice over IP (VoIP) services. The following features of SIP play a major role in the enablement of IP telephony and VoIP:

  • Name Translation and User Location: Ensuring that the call reaches the called party wherever they are located. Carrying out any mapping of descriptive information to location information. Ensuring that details of the nature of the call (Session) are supported.
  • Feature Negotiation: This allows the group involved in a call (this may be a multi-party call) to agree on the features supported � recognizing that not all the parties can support the same level of features. For example video may or may not be supported; as any form of MIME type is supported by SIP, there is plenty of scope for negotiation.
  • Call Participant Management: During a call a participant can bring other users onto the call or cancel connections to other users. In addition, users could be transferred or placed on hold.
  • Call feature changes: A user should be able to change the call characteristics during the course of the call. For example, a call may have been set up as 'voice-only', but in the course of the call, the users may need to enable a video function. A third party joining a call may require different features to be enabled in order to participate in the call
  • Media negotiation: The inherent SIP mechanisms that enable negotiation of the media used in a call, enable selection of the appropriate codec for establishing a call between the various devices. This way, less advanced devices can participate in the call, provided the appropriate codec is selected.

The SIP protocol

The SIP protocol defines several methods. ...
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voip-info.org

August 26, 2012, 12:55 am
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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 Guidelines before you post.

Update: We have added the Facebook Like and Google +1 button to the top right corner of all pages on Voip-Info.org. Please help recommend the wiki by clicking on them. We also now have Google+ page here and a Facebook page here. Visit them and add us. Thanks!


NEWS

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Web Conferencing With A Difference

August 26, 2012, 3:10 pm
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Professional Web Conferencing With A Difference

Smart Voice Network provides AVIDO online collaboration solutions and combines world-class web and video conferencing. Smart Voice uses innovative business social networking in a unified Cloud platform to drive real-time and social collaboration. With superior HD VoIP capabilities and multiple simultaneous video streams optimized for low bandwidth you can hold engaging live meetings, cost effective powerful webinars and interactive classrooms.


The Webconference is loaded with features here is a list of features

    • Live Video & Audio Streaming
Redesigned for optimized screen space utilization
Different sizes of video for different users
Full Screen Video
Start and Stop other broadcasts individually
Test your audio locally
Adjust individual user volume
Mute Incoming Audio when pushing PTT
HD and custom video settings for professional webcasters*
On-the-fly quality controls*
Integrated Audio/Video Wizard*
Dial in/out via PSTN / SIP*.

    • Messaging & Chat
Private Chat Messaging
Tabbed Chat
Moderator Chat
Chat colors
Choose text size
Show local times in status messages*
Emoticons
Question & Answer module with moderation*
Right-to-Left writing when required by language (Hebrew, Arabic, etc.)*
Allow editing of previous text chat messages*
Group chat*

    • Whiteboard & Markup
Redeveloped and redesigned
Scales and changes page with content
Initial Toolset & Functions: Shared pointer, Select Tool, Highlight pen / Pencil / Free draw / Magic Pen, Line/Arrow tool, Rectangle, Ellipse, Triangle, Polygon, Star, Text, Undo*, Redo*, Cut*, Copy*, Paste*,(coming soon) Delete, Erase page
Multi-layer support
Object manipulation via drag handles: Rotate, Resize, Move up a layer*, Move down a layer*, Bring to front*, Send to back*,
Object parameters (where applicable): Line color, Fill color, Transparency, Line weight, Shape specific parameters (coming soon)


    • Content Viewer, File Types, Document Converter
View multiple documents at the same time
Tabbed interface
Support for Additional document types: pptx, docx, xlsx, pdf, txt, video, audio
Improved converter architecture for higher stability and speed
Native support and shared player for MP3, FLV
Access shared content library
Cancel conversion already in progress
Improved feedback on duration of conversion and queue
Select multiple files and perform single action (delete, convert, open)
Improved controls: zoom, rotate, dragging*
Integrated Player for external video sources such as YouTube* (coming soon)
Integrated Player for external image sources such as Flicker* (coming soon)
Integrated Player for external document sources
Thumbnail navigation for multipage content*
Right-to-left paging when required by language (Hebrew, Arabic, etc)*


    • Server Administration
Improved interface
Search
Data sorting and filtering
Improved usage statistics
User tracking
Broadcast notifications to user Account Centers and into rooms* (coming soon)
Web-Based Installer, Upgrade Mechanism*
Set-Up Wizard*
Resource Monitor*


    • Real-time polling module
Record polls*
Pre-create polls*
Export polls*
Analyse poll results*
Assessment features*


    • Screensharing & Remote Control
Better codec using less bandwidth at equal quality
Share entire screen
Support for Windows, Mac OS X and Linux
Choose individual windows and applications to share (Windows and OS X only)
Remote Control other users’ applications
Better feedback when sharing is stopped
Revert to last A/V ...
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The best web conference

August 26, 2012, 3:17 pm
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Professional Web Conferencing With A Difference
AVIDO delivers a truly all-in-one web conferencing solution for all your organization-wide online communication needs.

Chat, Polls, Q & A
Web conferencing usage can vary widely, depending on business necessities, daily fluctuations in application requirements, and even personal preferences. Using AVIDO as your communication medium offers you easily accessible solutions to any communication challenge – whenever or wherever they arise.

Share & Present all your files as if you were in the same room.

Using AVIDO is easy, fast, and fun. plug-in and with no installation or download, you can be up-and-running in minutes; giving you and your team immediate access to your own online meeting room 24/7 That means you’ll have access to AVIDO features such as full-duplex video conferencing, screen and file sharing, chat, session recording and much more.
Presenter and participant(s) can choose chat font, size and even colors. They are also able to chat publicly with everyone or chat privately. You can even choose any language from AVIDO’s large selection which will be translated with a click of a button or you can select Auto Translate. This feature can enhance communications and make the chat more fun. AVIDO has emotion icons and much more..

With AVIDO Organizations are not confined to just voice and video conferencing, you may also need to share emails, files and documents – either in a structured environment or spontaneously.

Whatever you need, AVIDO makes it easy.

Simply upload your files to the online meeting room to ensure everyone is on the same page at the same time – literally. AVIDO supports Microsoft Office® documents, video files and even multimedia files such as Adobe Flash®.

With AVIDO’s screen sharing feature, you can show your computer screen at the click of a button and be sure that everyone is looking at the same thing. Intuitively and securely collaborate on live documents via remote control.

No downloads or Installation needed
The AVIDO client software is designed to work on any computer system that can run Adobe Flash Player®, meaning it will run on Microsoft® Windows®, Apple OS X®, various distributions of Linux® and Sun Solaris™. AVIDO will even run on Windows® PocketPCs for true on the road access.

The Most User Friendly Interface
AVIDO was designed with the user in mind. A typical conference organizer can learn the most important features within a few minutes, and 80% of the system within an hour. The key features are prominently displayed, making your first web conference an enjoyable and intuitive experience.

For a conference participant/invitee, it is even easier. Joining a meeting or
Please Visit our Website at:
Main Web Site
web conference
Or watch our You Tube Video
Video1
Video2
Video3
Video4

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Webinar at best

August 26, 2012, 3:26 pm
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How to Find the Best Free Webinar Services



The internet these days is a fascinating world yet at times it can be challenging to those who want to build an online business. Webinars, short for "web-based seminars," can be presentations, lectures, workshops or seminars that are transmitted over the web, enabling you to meet and market to a global marketplace.

There are many webinar services to pick from - Adobe Acrobat Connect, Citrix Online (GoToMeeting, GoToWebinar), Microsoft Office Live Meeting to name just a few - these services are very expensive and they do not give you a lot of options unless you pay a large amount of money but I would recommend using a similar webinar service as opposed to paying for a web room, especially when you're just starting out. I have personally used the Avidousa.com by smart voice network have also become interested in the services from smart voice network offer free options and are very easy for internet newbies.

Once you have decided on a webinar provider, the basic operation of your room will be based on these aspects: sharing desktop screens, talking via microphones or web cams and lastly the option for text chat. The most important will be the ability for you, the host of the webinar, to share your desktop screen as well as for your individuals in your audience to share their screen. This benefit allows you to literally show anything you like on your screen and broadcast your voice to your entire audience.

The potential activities you can use your new web room are endless from helping new associates with marketing, to showing how to setup blogs and websites to even helping your network marketing distributors place autoship orders and more. Having a global reach unleashes your potential to help and serve more individuals and eventually make more income online from home. So get started to day by doing some research and picking the best webinar service for your needs. And as mentioned, I have used avidousa.com and they offer reliable service.

While most computers should be able to run a webinar, you will want to use a computer with a fast processor. You will want to have a fast broadband internet connection (i.e. cable or DSL) and I would strongly suggest using a simple USB headset to transmit your voice. A web cam is not necessarily needed, but if you have one that can be useful for face-to-face meetings
Please Visit our Website at:
Main Website
Webinar Website
Or watch our You Tube Video
Video1
Video2
Video3
Video4

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Android VOIP

September 12, 2012, 7:44 am
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News


FAQ


VOIP Native Applications for Android

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DID Service Providers

September 12, 2012, 10:14 am
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A Direct Inward Dialing service provider delivers a telephone number over VoIP protocol including SIP, IAX2 or h323. This DID Phone Number will receive calls using a sip soft phone, hard phone, or an IP PBX. The charges are charged per month, and per min, and per channel based. You do not require a hardware card in case you have a DID service provider to receive call, ie a PRI Card or a Analog Card. The call travels to you all the way over the Internet

Cheapest DID Providers

see Cheapest ATAs and Service
see domestic USA DIDs for carriers

Free Service Providers Only (Free DIDs)


Algeria

  • Algerian DID numbers Currently only national Algerian numbers. Also 68 other countries available. Free forwarding to Voip/SIP, IAX,H323 Google Talk, etc many providers preconfigured. Forwarding to PSTN (landlines and mobiles) from 2cent/min, including mobiles. Mobiles Europe 6cent/min.

Argentina

  • Argentina Virtual Numbers $5.95 Per month / Free Setup| Virtual Numbers in Argentina with Free Forwarding Options(FlyNumber)
  • Argentina DID Numbers € 4.99/month| DomesticNumbers offers Argentina virtual phone numbers from 19 different cities in Argentina, including Buenos Aires. Website also in Spanish.

Australia


Austria

  • Sipgate.at Free personal 0720- national number in Austria.
  • TeleCallMart Unlimited Incoming Phone Numbers, Voip calls, SIP Phone, Auto Attendant, DTMF. No monthly fees. Low prices!
  • Austria Virtual Numbers $3.95 Per Month with Free Setup |Virtual Numbers in Austria with Free Forwarding Options(FlyNumber)

Bahrain

  • Bahraini DID numbers Currently only regional Manama numbers. Also 68 other countries available. Free forwarding to Voip/SIP, IAX,H323 Google Talk, etc many providers preconfigured. Forwarding to PSTN (landlines and mobiles) from 2cent/min, including mobiles. Mobiles Europe 6cent/min.

Brazil

  • Brazil Free DIDs Free Brazil DIDs. Limit 5 DIDs per IP. Free SIP Forwarding Only. Click on Free DID section on our site to request.
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VOIP Service Providers B2B

September 12, 2012, 1:57 pm
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Here is a list of VOIP Service Providers focusing on Business-To-Business services. This includes VoIP origination and VoIP termination, plans aimed at call centers, IVR providers and generic Asterisk users. See also:

Services which require the use of locked ATA devices should not be listed on this page. Nor should services which do not permit simultaneous calls — most services here support at least 4 simultaneous incoming calls. Please list only services which support Asterisk connections, via SIP or IAX2, to the PSTN.

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Hosted PBX

September 12, 2012, 2:01 pm
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Hosted PBX is a service where the call platform and PBX features are hosted at the service provider location. The business end users connect via IP to the provider for voice service.

"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 PBX. Hosted PBX is also sometimes referred to as Hosted VoIP.

Benefits of Hosted PBX


There are many benefits to using Hosted PBX rather than a traditional phone system, or an on-premise PBX. The main benefit is cost- a Hosted PBX system costs much less to set-up than an on-premise PBX. In many cases, there are no set-up fees for a hosted PBX system. Purchasing and setting up an on-premise PBX can cost tens of thousands of dollars. Hosted PBX phone systems fall under operational expenditure rather than capital expenditure, which also makes hosted PBX service attractive to businesses. With hosted PBX service, you pay a monthly fee, and the hosted PBX service provider takes care of the rest.

Another benefit to a hosted PBX system over an on-premise PBX is that hosted PBX service 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 a hosted PBX system. A downside to a hosted PBX is you may have a little less of an ability to customize your solution to your business, but many hosted PBX service providers can achieve a deep level of customization.

More Hosted PBX Service Providers



VoipTiger Offers a Free cloud pbx - Call center features - Numbers (DIDs) in 53 countries - Free Android/Iphone app - Grandstream IP-phones - Codecs: ILBC/G711/G729/G722


1Pipe Telecom | OnePipe is a CLEC providing SIP Trunking, Hosted PBX, IP and other services, we provide services on our own network and gateways.
  • Actual US CLEC
  • Branded customer portals
  • Multiple locations on one virtual PBX
  • Customized pricing.


.e4 Technologies - .e4 provides powerful, integrated IP Communications enterprise class services that provide the SMB customer to configure the exact voice, video, messaging and collaboration services they need in just minutes- all delivered on-demand.

  • 1 MONTH FREE TRIAL
  • VoIP and PSTN Service
  • PBX and ACD
  • Web, Audio & Video Conferencing
  • Skills Based Routing
  • Private/Encrypted and Public IM
  • Voice and Data Archiving
  • Reporting and Archiving
  • PSTN Gateway
  • CRM Integration
  • Web Contact Center


1aVOIP.com

1aVOIP.com delivers hosted PBX services with SIP & Mail feature sets. Set up your PBX environment with Voice, IM and Videos service, and connect to you customers with phone numbers in: ARGENTINA, BRAZIL, CANADA, CHILE, EL SALVADOR, MEXICO, PANAMA, PERU, PUERTO RICO, UNITED STATES, AUSTRIA, BAHRAIN, BELGIUM, BULGARIA, CROATIA, CYPRUS, CZECH REPUBLIC, DENMARK, ESTONIA, FINLAND, FRANCE, GEORGIA, GERMANY, GREECE, HUNGARY, IRELAND, ISRAEL, ITALY, LATVIA, LITHUANIA, LUXEMBOURG, MALTA, NETHERLANDS, NORWAY, POLAND, SLOVAKIA, SLOVENIA, SPAIN, SWEDEN, SWITZERLAND, UNITED KINGDOM, AUSTRALIA, HONG KONG, JAPAN & NEW ZEALAND


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SIP/IAX Services for Asterisk

September 12, 2012, 8:12 pm
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This page lists providers of SIP/IAX services for Asterisk.

Page Contents

Argentina

  • Tpad provides global SIP Trunking or IAX2 Trunking services to businesses around the world, along with high class wholesale VoIP / SIP termination, origination, toll-free, DDI, and DID numbers.
  • Supanet offers any business SIP Trunking or bespoke IAX2 trunks along with high class wholesale VoIP / SIP / IAX termination, origination, toll-free, DDI, DID, ISDN, SDSL, Fibre, MPLS, VPN, CPS, and Line Rental facilities.
  • Asterisk-voip.com.ar Ofrecemos tecnología , instalaciones completas y mantenimiento de Asterisk , instalaciones llave en mano, terminaciones SIP IAX for Asterisk y Trixbox. Pidanos un presupuesto Telefono +54.11.4136-3333
  • IP Communications offers SIP and IAX origination and termination services for Asterisk and Trixbox PBXs (DIDs, Toll Free, Long Distance, A-Z Calling).
  • Hablemos ITSP ofrece terminación SIP, soluciones varias de telefonía IP, numeración entrante, etc etc. Servidores en Argentina y Estados Unidos para mejor performance. Se puede generar una cuenta demo sin costo.
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Software Releases 2011

September 13, 2012, 1:23 am
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Dec 2011

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Software Releases 2010

September 13, 2012, 1:29 am
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December 2010

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Software Releases 2009

September 13, 2012, 1:33 am
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December 2009


November 2009

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Yeastar - NeoGate

September 20, 2012, 8:03 pm
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NeoGate TG200 - VoIP GSM Gateway(VoIP-GSM)

Designed for maximum cost reduction

NeoGate TG200 is a modular VoIP GSM/UMTS gateway with 1/2 channels providing GSM /UMTS network connectivity for soft switches, and IP-PBXs. It supports two-way communication: VoIP to GSM/UMTS and GSM/UMTS to VoIP. Thus the calls costs could be significantly reduced by VoIP or GSM network.

New_短_天线大号_正_NeoGate_TG200副本.jpg


Benefits

1) Cost Savings – Cost-Savings on phone calls between mobiles or to PSTN.
2) Back up – Should the landline network go down, GSM can be used as a cost-effective backup.
3) Easy to install – IP device with Web based management interface.
4) Easy to integrate.

NeoGate_VoIP_GSM.gif


Specification:

Number of GSM channels (Max): 2
Network type
UMTS: 900/2100MHz; 850/2100MHz; 850/1900MHz;
GSM: 850/900/1800/1900MHz
Protocol: SIP (RFC3261),IAX2
Transport Protocol: UDP,TCP,TLS,SRTP
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.726, G.729 A, GSM, Speex.

LAN: 1 (10/100Mbps)
Network: DHCP, Firewall, VLAN, DDNS, OpenVPN.

Size: 200x140x35mm
Power Supply: AC 100~240V/50~60Hz (DC 12V, 1A)

Features:

SIP proxy Registrar for IP phones included
Incoming call routing
Outgoing call routing
SMS sending and receiving (WEB interface)
Call Back
LCR (Least Cost Routing)
Top voice quality (EFR super sound)
Simple web based configuration
Easy to integrate / Easy to install


NeoGate TB400 - VoIP BRI Gateway(BRI-VoIP)

Smooth communication between BRI and VoIP network

NeoGate TB400 is a compact reliable standalone VoIP BRI gateway (BRI-VoIP/VoIP-BRI) offering the company using ISDN BRI lines an easy, cost-effective and flexible integration into any VoIP system or enabling any IP PBX to be connected to the public ISDN network at an affordable price. It could either provide VoIP access for your legacy PABX or extend an ISDN-BRI line of a PBX to a remote site over VoIP.

_NeoGate_TB400副本.jpg


Benefits

1) Access to VoIP network
2) Cost Saving - Cost-Saving on phone calls via VoIP.
3) Easy to install - IP device with Web based management interface.
4) Easy to integrate.

NeoGate_BRI_VoIP.gif


Specification:

BRI Port: 4 (RJ45)
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP,TCP,TLS,SRTP.
DTMF: RFC2833, SIP INFO, In-band
Codec: G.711 a/u-law, G.726, G.729 A, GSM, Speex.

Network: DHCP, Firewall, VLAN, DDNS, OpenVPN.

Size: 200x140x35mm
Power Supply: AC 100~240V/50~60Hz (DC 12V, 1A)

Features:

BRI ports can be used as TE/NT mode
SIP proxy Registrar for IP phones included
LCR (Least Cost Routing)
Simple web based configuration
Easy to integrate
Easy to install


SiSkyEE - Business Skype Solution

100% software based, Generate up to 30 Skype trunks!

Nowadays, Skype is very popular and you may found many customers are Skype users. ...
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VOIP Event Calendar

September 21, 2012, 1:11 am
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December 2012


October 2012


September 2012

  • 12-13 - Meet JeraSoft Team at Wholesale World Congress (WWC) in Madrid, Spain.

August 2012


May 2012

  • 14-16 JeraSoft Development, developer of billing software will attend ITW in Chicago, 14 - 16 May 2012. You can find us at the booth #612.

April 2012


March 2012

  • 6 - The SIP School is at UCexpo 2012 in London - What's up with SIP? Details
  • 26 - The SIP School is at Enterprise Connect 2012 in Orlando - SIP fundamentals and Interop Details


February 2012

  • 1-3 - Meet JeraSoft Team at ITExpo East in Miami, Florida, USA. Booth # 425.

January 2012

  • 15-18 - Meet JeraSoft Team at Pacific Telecommunication Conference’12 in Honolulu, Hawaii, USA. ...
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