Open source VoIP/Telephony One of the first open source VoIP projects -and one of the earliest VoIP PBXes, period-is Digium (Profile, Products,
Articles) -sponsored Asterisk. A highly mature platform licensed under the GPL, Asterisk
supports almost everything that even larger enterprises would desire of a VoIP gateway solution, including voice mail, call forwarding, conferencing, and even IVR (Interactive Voice Response). It also has call-detail records
- the golden goose of VoIP-as well as advanced features suitable for use in virtual classroom or virtual conference room
applications. Its large developer community contributes still more add-ons for the platform, both commercial and open source.
Voice over IP Voice over Internet Protocol, also called VoIP IP Telephony, Internet telephony, and Broadband Phone is the routing of voice conversations over the Internet or through any other IP-based
network. Protocols used to carry voice signals over the IP network are commonly referred to as Voice over IP or VoIP protocols. They may be viewed as commercial realizations of the experimental Network Voice Protocol (1973) invented for the ARPANET.
Voice over IP traffic can be deployed on any IP network, including ones lacking a connection to the rest of the Internet, for instance on a local area network.
VoIP Telephony with Asterisk
Signate offers the second edition of the best-selling book about Asterisk, VoIP Telephony with Asterisk, a well as Telephony Stack software that turns your PC with an internet or PSTN telephone connection into a VoIP PBX ready for configuration.
The print version of VoIP Telephony with Asterisk is available from Signate and through selected resellers for $39.95 plus shipping in the United States. The Signate Telephony Stack Set is $59.95. You save $10 when you buy them both together.
Signate has partnered with OSoft to offer an e-book version of VoIP Telephony with Asterisk that uses the OSoft ThoutReader?. The
Thou Reader, which is distributed at no charge at www.OSoft.com, makes it easy for e-book readers to browse, search, bookmark, and append content in electronic form, and readers can eliminate shipping and customs costs as well as transit delays at the time of purchase.
VOIP Wiki - a reference guide to all things VOIP
This Wiki covers everything related to VOIP, software, hardware, service providers, reviews, configurations, standards, tips & tricks and everything else related to voice over IP networks, IP telephony and Internet Telephony.
If you need more space then create a Wiki page for your company and post only a brief synopsis here. Entries longer than 85 characters will be truncated without prejudice (either the author can trim it or someone else will do it for you...but it might not be exactly how you wanted it).
Open Source E-Book Reader Signate, a leading global provider of design, installation, configuration, training and management services for open source VoIP telephone systems, today announced the publication of an e-book version of VoIP Telephony with Asterisk, its best selling guide to the open source Linux PBX movement that is revolutionizing business telephony. The e-Book uses OSoft's ThoutReader?."The digital telephony market is moving fast, and Signate came to us looking for a way to offer books to their overseas customers without shipping costs and long transit delays," said Mark Carey, OSoft president.
VoIP Telephony with Asterisk 2nd Ed. Asterisk is the Open Source software PBX that's revolutionizing business telephony. Asterisk runs on Linux and provides a strategic, full-featured approach to voice and data transport over TDM, switched, and Ethernet architectures. Forward-thinking observers expect open source IP telephony products will replace proprietary hardware and software with standard Linux servers and open call control over the coming
decade. Once your open source PBX is installed, you will be able to place calls through the telephone company or over the Internet. Your PBX and your telephones can be anywhere. A phone in a home office or a phone in an overseas office works just like a phone in your server room. And the calls can be free or very low cost.
Open Source VoIP Branches Out Telephony is generally regarded as a mission critical business service, and any downtime can be costly in terms of lost productivity and lost business: Most companies want their phones to be a dial-tone service. And since the PBX is the heart of a corporate telephone system, you'd imagine it would not be something that the company would want to scrimp and save on. No one ever got fired, in other words, for buying a Cisco PBX.Which is why the Asterisk open source PBX is quite surprising. Running on Linux on standard PC hardware with suitable PCI interface cards, it works as a PBX with extra telephony features like voice mail and conferencing, working with analog phones and standards-based IP phones for VoIP telephony. And it's certainly no geek plaything. You could run your own home PBX using Asterisk, certainly, but a single machine can handle raw call volumes in the low thousands,or about 120 channels with echo cancellation and
transcoding. And by using built-in peering technology you can link up multiple Linux boxes to make a PBX serving a hundred thousand users.
GNU Develops Open Source VoIP
GNU has decided to compete with existing VoIP solutions by developing and releasing an open source alternative stack, that is said to provide a scalable environment for enterprise-level VoIP demands. The stack is designed to emphasize modularity and extensible functionality and be integrated with web servers and databases.
This development is being sponsored by Tycho Softworks, which helps to maintain the GNU Bayonne telecommunications server. In addition to Bayonne, the GNU telephony stack also includes the GNU RTP stack, the Open H.323 stack, and the GNU Gatekeeper H.323 call server.
Over IP Technology A telephone service that uses the Internet as a global telephone network. Many companies, including Vonage, 8x8 and AT&T (CallVantage), typically offer calling within the country for a fixed fee and a low per-minute charge for international. Broadband Internet access (cable or DSL) is required, and regular house phones plug into an analog telephone adapter (ATA) provided by the company or purchased from a third
party. Software-based phones require using the computer to make and receive calls. Usually a no-cost option if both sides are on the same service, softphones also mean you can call any phone in the world no matter where you are in the world from your laptop (with an Internet connection). Per-minute charges apply for calling regular phone numbers, but you may not be able to receive a call from a regular phone. In 1995,
Vocal Tec Communications introduced the first VoIP service in the U.S., which was softphone based.
Emergency response team picks open source telephony An emergency response centre in South Africa has implemented an open source VoIP telephony system to cut the cost of handling calls.
Asterisk, an open-source application that provides all the functionality of a standard private branch exchange (PBX) system, has been installed at the Provincial Emergency Management Centre, in Cape Town, South Africa.
The installation was conducted by Connection Telecom and completed at the end of January. The system will be used for the first time in March during one of the country's biggest sporting events, the Argus Cycle Tour.
Signate Announces Second Edition of VoIP Telephony Signate, the leading provider of VoIP telephony solutions that combine high performance hardware and open source software, today announced the second edition of VoIP Telephony with Asterisk, by Paul Mahler, Signate's Chief Technical Officer. Intended for those new to the Asterisk open source PBX software project, the second edition of Mahler's best selling guide has been revised, re-organized and expanded to reflect the current state of the project.
VoIP Telephony with Asterisk Second Edition shows readers how to combine Asterisk open source software with industry-standard Linux-based computer hardware to create reliable, highly capable VoIP telephony systems with a total cost of ownership that's typically less than half the cost of proprietary systems.
Source for corporate networks
The OSS InteropLabs Initiative explores this query. During InteropLabs' HotStage event in early April, the team assembled a group of components to show how a substantial enterprise could use open source software to run its IT infrastructure. The scenario is this: A large corporation with its purely open source network acquires a smaller company, which then has to migrate from being a pure Microsoft shop to interoperating with the open source network of its new parent. This fictitious enterprise requires the network services of any growing company: departmental communications, file storage, e-mail, VOIP telephony and a network infrastructure to support users.
Four racks of Hewlett-
Packard Proliant DL320 1U servers and assorted appliances were configured to represent a typical network topology. A server farm of Red Hat Fedora Core 4 servers and a mix of other Linux platforms, including several IBM/Lenovo laptop clients and network appliances based on open source software from Vyatta and Force10, supported the applications and services running across the network.
Integration There can be few organisations that have not heard of the potential benefits of migrating to voice over IP telephony ? free calls to other VoIP enabled partners and radically reduced costs for calls to conventional telephones.
Whilst there has been a great deal of publicity regarding consumer services such as Skype and Vonage, many companies are still unclear how to migrate to VoIP in the most cost-effective manner, preferably without having to throw away all their existing hardware. The latest proprietary VoIP-enabled PBX hardware can cost tens or even hundreds of thousands of pounds, an enormous upfront investment in what is still a rapidly developing technology.
Industry-Standard VOIP Phone Using All Free Software
"Voice over IP on a HardPhone running Linux and just using Open Source software became real. We have
successfully installed and tested (interoperability with Cisco 7960 as well as Pingtel xPressa in an environment with a partysip SIP registrar and proxy) the linphone SIP phone on a StrongARM based TuxScreen. Here is the link describing the steps for others to use the setup as well: TuxScreen running SIP. All the infos for setting up a comparable installation can be found on the URL, please also feel free to ask or drop opinions. Many thanks to the linphone developers as well as to my student Florian
Winter stein (for working on a console linphonec version). The setup (on a StrongARM system) is well suited for PDA (iPAQ) or wearable environments as well."
developer Channel Aculab offers solution providers and VoIP developers a wide range of hardware and software building blocks for integration into high performance, wired and wireless communications solutions. Products for use in telco or enterprise platforms incorporate digital network access and media processing resources, in both PSTN and IP environments.
To Aculab, the importance of not only developing high quality products but also working closely with our customers and partners to help them through each stage of their product's lifecycle is paramount. Aculab supports VoIP developers with pre-sales and post sales technical consultancy and support, training and co-marketing.
Open-source telephony server runs on Windows XP The second generation Open Source Bayonne telephony server is available for immediate testing and further development from
Telephony.org. GNU Bayonne 2 is the first release of the GPL-licensed telephony server available for installation and testing under Microsoft Windows, according to lead developer David Sugar.
Sugar says that Bayonne 2 is designed to run as a "headless system service" on any 32-bit Windows OS including Windows XP Embedded. However, only sound card and SIP drivers are currenty supported under Windows. Additional Windows support for Voicetronix, Intel-Dialogic, and openh323 is expected in the near future.
Digium announces low-cost business VoIP card
Digium's Wildcard TDM2400 enables as many as 24 phone ports, either all VoIP or a mix of traditional analog telephony and VoIP. The card also could hook up traditional fax and other equipment into a VoIP environment. Previously, Digium offered a 4-port analog card.The 32-bit 33MHz PCI 2.2-compliant card comes in two flavors: FXS, which connects the Asterix open-source PBX to analog or IP phones; and FXO, which connects to a phone line through a PC.
Digium also announced its new echo-cancellation module, which promises enterprise-grade VoIP quality. The module is optional with either card.
Professionals Internet Technology
The show is the only Linux and Open Source event in the UK and will appeal to a cross section of industries including automotive, health, banking, retail, education and local government. Participation has already been confirmed by some of the largest players in the Open Source world including Google, RedHat and Novell.
The two-day event will include a comprehensive conference programme, featuring real life case studies such as Chris Lewis, Group IT Manager, Malmaison Ltd, a major hotel-chain, which is in the process of moving to an Open Source environment. Laurent Lachal, a senior analyst, from research group Ovum will talk about Linux software and service revenue opportunities in enterprises and public sector organisations, as well as among individual consumers.
Single-handedly kill VOIP telephony Devised a way for telephone companies to detect data packets belonging to VoIP applications and block the calls. For example, now when someone in Riyadh clicks on Skype's "call" button, Narus's software, installed on the carrier's network, swoops into action. It analyzes the packets flowing across the network, notices what protocols they adhere to, and flags the call as VoIP. In most cases, it can even identify the specific software being used, such as
Skype's."If, like me, you're just getting into the entire world of Voice over IP or Internet Telephony, this story should be pretty disturbing.
The IEEE Spectrum writes that this solution from Narus isn't expected to affect within-VOIP-network calls (e.g., Skype to Skype) but rather VOIP calls that are redirected out onto the existing telephony infrastructure.
Peer-to-Peer VoIP telephony debuts Popular Telephony recently unveiled its technology concept for serverless and switchless telecommunications, which the company heralds as a "ground-breaking, patent-pending core technology for the implementation of peer-to-peer, server-free telecommunications systems." Peerio is described as a highly portable, operating system independent middleware for embedded devices that seamlessly integrates into VoIP ICs, IP phones, and other telephony-enabled systems and modules, and is said to be capable of being ported to any IP telephony device.
company says Peerio addresses the scalability, security, redundancy, and other system issues that are required in both enterprise and global telephony network implementations, while supporting all standard and advanced telephony features. Additionally, it is "protocol-agnostic," enabling switchless and serverless communications for VoIP systems and devices running on SIP, H.323 and all other standard or proprietary protocols.
Open Source Telephony Market
The Digium 24-port card offers the highest analog density available in a PCI card and can scale up to 48-ports with two cards and two slots,? said Mark Spencer, president of Digium. ?With its flexible scalability features, our 24-port card is the best hardware card available for small and medium businesses looking to build an inexpensive, sophisticated VoIP telephony solution without compromising the use of multiple PCs.?
The Wildcard TDM2400P replaces the requirement for a separate channel bank and T1 interface cards while offering superior echo cancellation on both FXO and FXS interfaces. The quad-FXO and quad-FXS modules are interchangeable allowing the combination of interfaces up to six slots for 4-port FXS or FXO modules. With this new card, small and medium businesses can benefit from features such as high density in fewer PCI slots and an industry standard 50-pin Amphenol connector for easy installation.
Open source phone system
When we started in 1997 and then opened our first real office in 1998, the first phone Summersault ever owned was a small, gray two-line office model with, I believe, five separate voice-mailboxes. It cost us around $200, after we spent a long time researching and discussing just the right one to get. It sat quietly on my desk, and when the occasional call did come in, everything worked just fine - we never had to open it up, reprogram it, reboot it, back it up, or monitor it. It?s not hard to long for those days, as Summersault?s growth has meant some costly and time consuming expansion in our phone infrastructure over the years. But our recent experience installing and configuring the Asterisk open source PBX phone system has given me some hope that we?re returning to an era where the phone is once again a useful tool that saves people time and makes communication more efficient, instead of less so.
is VOIP Telephony
Short for Voice over Internet Protocol, a category of hardware and software that enables people to use the Internet as the transmission medium for telephone calls by sending voice data in packets using IP rather than by traditional circuit transmissions of the PSTN. One advantage of VoIP is that the telephone calls over the Internet do not incur a surcharge beyond what the user is paying for Internet access, much in the same way that the user doesn't pay for sending individual e-mails over the Internet.
There are many Internet telephony applications available. Some, like Cool Talk
and NetMeeting, come bundled with popular Web browsers. Others are stand-alone products. VoIP also is referred to as Internet telephony, IP telephony, or Voice over the Internet (VOI)
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