Source: andyabramson.blogs.com

A really comprehensive list of what has been tabbed the Top 50 Open Source VoIP applications has been assembled and posted.

In reading through the list I’m amazed at the familiarity I have with many of the applications, including many of which some of my agency’s clients have worked with, or in the case of one, has actually been a client, VoIPster just before their acquisition.

Source: snapvoip.blogspot.com

The Philadelphia-based Ivy League university, University of Pennsylvania, currently has over 1,250 Session Initiation Protocol (SIP) IP phones on desktops, tied to a back end based on SIP Express Router — an open source VoIP call-control and routing stack, and Asterisk for voice mail messaging. Don’t go away! This is just the start. The University has extending the VoIP network in to a 15000 seats, in it’s plans!

Deke Kassabian, the university’s senior technology director for information systems and computing, plans to grow that installed base by a factor of more than 10 over the next five years. Driving the project is the desire to get off costly Centrex monthly fees and infrastructure, and the promise of an open source, standards-based VOIP infrastructure that provides superior integration and control.

"If we can run one modern IP network for voice, video and data …. there’s a clear win," Kassabian says. "If we provide business telephony internally, less money leaves the university."

The Linux-based SER call control and Asterisk messaging servers were a better fit with UPenn’s standard back ends for authentication (Kerberos and RADIUS), its OpenLDAP directory structure, and e mail. While commercial IP PBXs are adaptable to these platforms, "they don’t work that way out of the box" typically, he adds.

With open source running extensively throughout the university — from directories, to e-mail, DHCP and DNS — the level of expertise in open source troubleshooting and development was there to support the Asterisk plans, Kassabian says.

"For years UPenn has had a strong open source talent pool. As a result, we have the staff and expertise to develop and roll out open source VOIP."

UPenn’s work with the Asterisk community is also paying off by improving the product itself. University programmers have already contributed to two additions to the code base, which is now supported in the main release. One change integrates IMAP-based voice mail and messaging stores, and another involves improvements in SMDI signaling between IP phones and voice-mail system back end.

"Instead of going off and making changes ourselves," Kassabian says, "we get our changes built into [the code base] and don’t have to maintain them ourselves. They’re part of the next distribution."
The infrastructure Kassabian and his team built is designed for high-availability VOIP, with redundant connections to IP call and feature servers, PSTN and IP telephony service provider (ITSP) point-of-presence links. Two data centers on campus host redundant clusters of Asterisk boxes, SIP proxy servers, and media/messaging feature servers. Phones on the network can register to and access any set of servers. "In this way, there’s no single failure, and no single site failure that would take out the servers," Kassabian says.
For outbound calling, UPenn is using a mix of VoIP and PSTN services. For long distance service and other calls, UPenn is plugging its campus VoIP network directly into a SIP trunking service from Level 3. A pair of dedicated Cisco 3600 routers also support PSTN links for local calls, and as a back for ITSP service.

Kassabian and his staff, mostly with IP networking backgrounds, also had to get up to speed with voice system jargon and terminology before being able to understand user needs. "We had to learn about how people use their phones," Kassabian says. "I had to learn what a bridge line appearance actually was."

Like many large organizations converging voice and data networks, consolidating the school’s telecom and data network teams helped tremendously.

"Having people from our traditional telecom organization learning IP technology has been great," he says. "And our networking staff has been learning telephony. That’s all been part of it. As time goes on, more of us are more well cross-trained and the two technologies come together very well."

More information at UPenn voice

Source: voipcentral.org

The Linux users have now a reason to feel-good for Skype. After almost nine months, Skype has made a beta release of its VoIP software for them adding sophisticated features to open source platform for the first time.

Skype is always late while updating its Linux versions in comparison to Windows and Mac. Skype 3.2 version is available for Windows users and 2.6 beta for Mac users. Anyway, the VoIP giant has finally updated its Linux 1.3 this month and dubbed it Skype Linux 1.4.

The key features of latest version include call forwarding, enabling users to forward incoming calls to another phone line and support for gilbc 2.3 systems. Skype has also fixed the bugs found in its Alpha version.

You can download and test the Linux 1.4, for which you require 400 Mhz processor, 256 MB RAM,20 MB free disk space, Microphone and speakers or headset and obviously Internet connection.

Read

Source: voipcentral.org

The greatest advantage of open source VoIP application over proprietary is that the former offers the flexibility for organizations to modify the code as needed for specific purposes since the code is open.

Besides cost saving factor, the open source brings a strong security in the VoIP network than the closed source. Perhaps, it is the one of the reasons why the demand for open source VoIP will increase tremendously in coming days.

Here we have a familiar open source player Solegy, a provider of hosted VoIP software has announced the availability of its free Softphone that would operate on Windows.

Based on open source code, the Softphone supports SIP (Session Initiation Protocol) VoIP. It can be downloaded from Solegys website.

Compatible with hosted ServicePDQ platform, the SIP Softphone includes instant messaging, branded skins and account management features like available pre-paid balance, time left for call, real-time resource allocation, and delegated administration and one-click purchasing of additional credit.

The company claims the latest softphone integrates the iLBC, speex, GSM, and G.711 open-source codecs. It can also support proprietary codecs such as g.729 and g.723. The striking feature of Solegy softphone is that it has the ability to modify to bring IM client with present, customer branding and web page integration.

Source: snapvoip.blogspot.com

My previous article "SER vs OpenSER, There is no real Comparison" is little out of touch. Even though I wrote the article I still partial to SER. Look at the links on the right side, "My favorite SIP Router" it has been the same since I started this blog. What drove me to write the article was the same reason that OpenSER came to be. Corporate or political bickering’s, and nonoperational home site, (one day it was up and next day it was down) so much so I stopped even trying to visit iptel.org site. I did visit Berlios developer site. That is the time I discovered OpenSER. What do you expect? it was a source of encouragement for me. I started immediately to work on OpenSER. I will continue to do so until SER assures me that it will not play the old tricks. I still have both the tracks and have deployed SER and OpenSER for many of my clients. I am also a very good fan of Asterisk and TrixBox (formerly Asterisk@home) and there are instances that I needed Both SER and Asterisk to find a solution for certain clients.
VOIP IP Telephony is very much more than what you see in the press. I have Academic, and corporate deployments that you will not read about in the press. Some of the deployments I have hit the ceiling on most of the OSS software but was able to overcome by using other OSS solutions like clustering to solve my problems. Although I like to code, due to the nature of my work and academia, I spend more time in design and deployment. So I let the others do the coding. That is why I rely on SER, Asterisk and the likes so much. One thing is for sure. Unless client insists, all my designs are OSS based. And it stand close to 90% OSS now.
But as time went by, things started to change at SER. New and better management, a new and certainly better IPTEL.ORG site and has come a long way. I do still work on SER and I did write about New SER.

All this said, SER is still a better solution than OpenSER. But for that information, please wait for the Soon to be written "Technical Diffrences between SER and OpenSER"

Links; as they were published on my site;
VOIP IP Telephony: What’s new in SER 0.10.x ? A lot and…

VOIP IP Telephony: SER gets a new home, Migrating to Drupal.

VOIP IP Telephony: SIP Express Media Server, SEMS, design documentation released by IPTEL.ORG

The article relating to SER vs OpenSER
VOIP IP Telephony: Ser VS OpenSER, There is no real comparison!

Ser at IPTEL.ORG
OpenSER Site

Source: snapvoip.blogspot.com

Several open-source vendors made announcements at last week’s AstriCon, Asterisk’s annual conference. Probably the most significant was that Digium, which makes commercialized Asterisk VoIP software, is partnering with Polycom to co-develop a SIP-based suite of telephony products for small and midsize businesses. Polycom phones will be able to be bundled with Digium’s Asterisk Business Edition, and Polycom will be Digium’s preferred VoIP phone manufacturer. Other companies announced applications like click-to-call for Asterisk.

Conference organizer Steve Sokol says AstriCon has grown each year since the first event two years ago. This year, for the first time, there also were a significant number of enterprise users at the conference. Sokol suggests news stories, venture capital funding for Digium, and developments like Sam Houston University dumping Cisco VoIP for Asterisk have solidified open-source telephony as an option. The switch from Cisco to Asterisk, which I wrote about in this aricle;
VOIP IP Telephony: A Suprisingly Simple Switch from Cisco Call Manager to Asterisk

Other companies have also recently bought into open-source communications. Amazon said last month that it would be rolling out open-source telephony from Pingtel enterprise-wide after a pilot in its Seattle headquarters; and marketing group InterMedi@ Marketing Solutions’ 1,000- seat call center will soon be powered by Ranch Networks’ Asterisk switches.

Fonality’s Lyman says the examples of his company, Digium, and Pingtel show that there’s a support model out there that works for enterprises and raises examples like Apache Web servers to show where open source is being used today in other critical applications. Still, don’t expect an immediate groundswell of support for open-source IP PBXs just yet. The open-source telephony movement is in its early phases—Asterisk is a fly to Cisco’s 800-pound gorilla—and most large vendors and customers haven’t jumped on the bandwagon.

Conference proceedings could be found at astricon.net, but why PPT files, I read all PDFs but my system does not read ppts.

Links;
Astricon 2006 files
Asterisk
The Sokol site
Here are some people who were at Astricon;

Attractel

Attractel, developer of state-of-the-art VoIP solutions, exhibited a suite of Attractel family products at AstriCon. Zoiper is their SIP/IAX support softphone that is packed with a variety of conventional and advanced features for Windows/Linux/Mac OS systems.Comes in Free and Biz version. Their Predilux has more to offer than just a predictive dialer with different modes to best fit Call Centre needs, the Service Line Platform offers service line providers a versatile tool to easily manage their core business and their Rate Engine is a real-time application for ITSPs to flexibly delineate and maintain their rate and markdown plans.

AudioCodes

AudioCodes successfully completed certification testing with Digium’s Asterisk Business Edition Software for AudioCodes’ media gateway platforms. Some AudioCodes SIP Gateway products that completed certification testing are the TP260/SIP, Mediant(TM) 1000 and the MediaPack(TM) to ensure interoperability and compatibility when integrated with Digium’s Asterisk Business Edition software. AudioCodes and Digium’s collaboration will offer better design options for telephony developers and enable rapid deployment of high quality and scalable SIP-based solutions.

Digium

Digium partnered with Polycom and AudioCodes for added conveniences and benefits such as rapid feature development for SIP-based telephony solutions with Polycom and rapid deployment of high quality and scalable SIP-based solutions with AudioCodes. See Polycom and AudioCodes summaries for more information.

LumenVox

LumenVox provides speech recognition for Fonality’s trixbox Application Platform, through integration into the platform. The Lumenvox Speech Engine, which recognizes words and phrases, will be part of trixbox 2.0. The partnership brings point-and-click installation and management of the LumenVox speech recognition to the trixbox community.

Mexuar

Mexuar launched its Corraleta Technology SDK for seamless VoIP Click-to-talk applications. This new solution will enable rapid development and deployment of VoIP click-to-talk apps for online businesses so that any website visitor can use the company’s web browser and PC to make free calls and talk with sales or support in the company’s contact center within ten seconds. The Corraleta-developed application can be used to initiate calls to the contact center via a button on a web page, or can trigger a callback from the contact center to the user’s chosen telephone. In addition to enabling IP calls, the Corraleta SDK can also provide information about the user’s online experience to the merchant’s customer service agent when the call is initiated. The agent software can display this information directly or use it to trigger lookups in company databases to retrieve details such as customer records, purchase histories and so on.

MIX Networks

MIX Networks, a service provider, offering a full suite of high-value next-generation VoIP solutions to residential, business and reseller customers has recent implemented DUNDi (a peer to peer system for locating Internet gateways to telephony services created by Digium) to ease of network administration, add stability and flexibility to its already strong network foundation. In keeping with the goal of delivering quality VoIP services to end users, MIX Networks builds a strong, redundant end-to-end MPLS VoIP network.

NetLogic

NetLogic announced "PBX-Preferred" Certification program that is based on the rigorous testing of application IPBX software and systems which includes bench testing to ensure proper SIP trunk connectivity and quality, beta testing in the field with actual business users over NetLogic SIP trunks and implementing in a "general availability" environment with NetLogic SIP trunks.

PBXtra(TM) Standard and PBXtra Call Center from Fonality; Switchvox SOHO and Switchvox SMB; and Digium’s Asterisk Appliance have earned PBX-Preferred Certifications. IPBX software earning the NetLogic PBX-Preferred Certification include: Digium’s Asterisk Business Edition; Asterisk; and Free PBX.

NETXUSA

NETXUSA Inc. has established a West Coast Distribution center in Henderson Nevada. The new location will provide a larger window of time to have products shipped while realizing increased savings on freight charges. As in the South Carolina corporate office a full compliment of VoIP products will be available for immediate shipment.

Polycom

Polycom and Digium partner to offer integrated SIP-based telephony solution for SMB market. This multiyear partnership will cover development and marketing of the SIP-based telephony solution and will provide SMBs with a tightly integrated, standards-based solution with simplified provisioning, broad support for Asterisk telephony features on the Polycom phones and the delivery of new capabilities. Polycom’s SIP desktop and conference phones will be combined with Digium’s Asterisk Business Edition, at a lower price point than proprietary systems and with added control, rapid feature development and more. With this agreement, Polycom becomes Digium’s preferred VoIP phone provider.

Ranch Networks

Ranch Networks provides optimal VoIP solution to Intermedi@ Marketing Solutions for 1,000 seat call center, across five facilities. Intermedi@ Marketing Solutions will now gain access to Ranch Network’s unique features such as VoIP Matrix Technology(TM), 1+1 High Availability and Media Bridging through RN 40 appliances to ensure call quality and reliability. This unified VoIP PBX and predictive dialer based on the Asterisk and VICIDAL open source projects will also provide a more secure network across Asterisk server farms.

RedFone

RedFone announced their next generation foneBRIDGE (T1 or E1 PRI-to-Ethernet bridge) that provides Asterisk with high availability, fail-over capabilities and load balancing. Normally offered through partner dealers, the foneBRIDGE is an integrated black-box "appliance" designed to streamline installation and enable redundant design of Asterisk based VoIP systems.

SIPBox

SIPBox appoints Roger Gusloff as new chief financial officer. Roger brings over 18 years of experience in financial leadership as a CFO, consultant, and a senior accountant. Most recently, he was CFO of Acme Refining Company, where he was responsible for the complete management of the company’s financial position. During his tenure, Roger oversaw multiple strategic acquisitions and managed assets exceeding $40 million. Roger began his career at KPMG Peat Marwick, where he spent several years as a supervising senior accountant before starting his own consulting firm, Gusloff & Associates. Roger was responsible for all facets of the business’ operations including the financial management as well as sales development, and employee and contract management.

Star2Star

Star2Star gives TAPI dialer back to open source community. This TAPI (telephony application programming interface) is used for connecting telephone services to a computer running Windows and can integrate with any Windows-based product such as ACT and Outlook. Star2Star’s TAPI Dialer software allows users to "click to call" via the Internet and will work with general Asterisk installations.

T2 Supply

T2 Supply unveils new line of video conferencing products, the HDX 9000 series. New products include the HDX 9001(TM), HDX 9002(TM) and HDX 9004(TM). The HDX 9001(TM), HDX 9002(TM) both have four professional high definition video inputs and three video outputs with standard definition video at 30 fps; the 9002 has an added feature of 1280×720 (720p) resolution - the highest quality video resolution available in the market today. The HDX 9004(TM) has five professional high definition video inputs, four video outputs, standard definition video at 30 fps and 1280×720 (720p) resolution. Both the 9002 and 9003 can be used with the Polycom Eagle-Eye camera for the UltimateHD experience.

Trixbox

Fonality introduced trixbox 2.0, the newest version of its easy to use open source telephony and application platform. This new version provides the trixbox community with increased reliability and many new features including a point-and-click, web-based graphical user interface (GUI) and expanded LAMP (Linux, Apache, MySQL and PHP/PERL) stack to LAAMPS by providing Asterisk telephony and SugarCRM. Additionally, trixbox 2.0 includes FreePBX, FOP, HUDlite and web-based package manager which allow users to upgrade individual components of their deployment versus having to reinstall from scratch with each upgrade.

VoxBone (Read the article, http://snapvoip.blogspot.com/2006/11/voxbone-adds-support-for-asterisks-iax.html )

VoxBone announced support for the Inter-Asterisk exchange protocol (IAX) from Digium. In addition to IAX, VoxBone already supports SIP trunking services, providing customers with a choice of standards to meet their particular needs.

Xorcom

For the first time, Xorcom presented their full suite of all-FXO Astribanks. Xorcom, creator of Asterisk-aligned hardware, demonstrated two Astribank-32 FXS connected to a TS-1 unit and also presented the 16 port and 8 port versions with analog phones. Since all Astribank units are USB connected to the Asterisk server, implementation is very easy. There is no need to shut down the machine, open panels or run into Ethernet issues. The Astribank units are controlled by the Asterisk server and do not have their own identity like gateways do, which simplifies the maintenance significantly. The integrator deals with configuring only one unit - the Asterisk Server. The TS-1 is an embedded Asterisk server and has a web interface, one half-size PCI slot and 2 x USB 2.0 connections. Special features include USB, disk, complete backup and restore, updates from the internet and more.