VoIP Monitor - VoIP Events

The SIP Forum Announces Dates and Opens Registration for Third Annual SIPNOC

VoIP Monitor — Mon, 22 Oct 2012 21:05:54 GMT

The SIP Forum announces it will host the third annual SIP Network Operators Conference (SIPNOC 2013), an educational conference focused on the challenges and opportunities of deploying SIP-based carrier services worldwide. Building on the success of past conferences, this year the SIP Forum will add an extra day of special workshops and sessions, designed to educate service provider technical staff on best practices and strategies for the successful implementation of SIP-based services and applications.

SIPNOC 2013 will be held at the Hyatt Dulles Hotel in Herndon, Va., from April 22-25, 2013, and will continue its practice of providing leading technical and operations personnel from the global carrier community with educational, non-commercial and technical content focused on the real-world challenges of deploying SIP services in global IP networks.

Early bird registration is available at http://www.regonline.com/sipnoc2013. More details about SIPNOC 2013 are available on its conference website at http://www.sipnoc.org.

"The growth and acceleration of SIP services within the international service provider community - and the accompanying technological, logistical and businesses challenges - was the original impetus behind SIPNOC and remains our guiding motivation," SIP Forum Chairman Richard Shockey said. "We're pleased to bring this unique gathering for the network operator community back for a third year, once again providing a non-commercial, technically-oriented setting for discussing and sharing ideas and knowledge about SIP implementation."

SIPNOC 2013 will once again bring together communications industry leaders to learn, discuss and formulate new ideas and strategies to address the challenges and opportunities for SIP-based carrier services in fixed and mobile IP network environments. The conference, which is designed specifically for SIP network operations personnel and engineering staff, will feature well-known industry speakers and a number of highly technical educational and instructional panels and sessions. The SIP Forum will also host networking events at the conference and offer a series of informal "Birds of a Feather" sessions, which encourage discussion on a variety of topics in hallways, available meeting rooms and break-out areas.

"SIPNOC demonstrates the SIP Forum's ongoing commitment to serving as the industry's non-commercial think tank for international SIP interoperability and deployment," said Marc Robins, President and Managing Director of the SIP Forum. "Our third annual conference will expand upon key issues identified in previous discussions and through our SIP Forum Task Groups, with the aim of delivering high-level technical support and guidance to the broad service provider community worldwide."

SIPNOC 2013 will build on critical themes discussed at past events, including: application development and testing; SIP trunking interoperability and the SIP Forum's SIPconnect 1.1 technical specification; Fax over Internet Protocol (FoIP) interworking; implementing SIP over IPv6; user-agent configuration; emergency services; policy servers; security; operational issues; call routing and peering; troubleshooting and monitoring; SIP Interconnection; HD-Voice Deployment Challenges; and Video interop issues.

"SIPNOC is an international gathering where communications engineers and network professionals can discuss and troubleshoot the real-world intricacies of working with SIP every day," Robins added. "We expect this year's conference to build on our past successes, and explore new complexities as we strive together, as a community, to deploy SIP-based services in diverse network environments worldwide."

Attendees at SIPNOC 2013 will include telecommunications providers, major backbone operators, interconnect and wholesale solution providers, ISPs, ITSPs (Internet Telephony Service Providers), cable operators and wireless network operators, as well as large enterprises deploying major SIP initiatives. Industry stakeholders - such as network equipment vendors, government agency representatives and academic research organizations - are also encouraged to attend.

The SIP Forum enjoys an international reputation for developing key educational events around SIP deployment. The organization's SIPit series of interoperability testing events regularly provides a test bed for SIP-based applications and equipment and has been heralded as critical for the development of new products and services in the industry. The SIP Forum is also home to committees and task groups comprised of industry experts examining a myriad of SIP-related topics, including the use of SIP with FoIP, video, SIP over IPv6 and user-agent configuration. SIPNOC 2013 Corporate Sponsorship Opportunities

4PSA Showcases VoipNow 3 Service Provider Beta at ITEXPO West 2012

VoIP Monitor — Thu, 04 Oct 2012 21:05:40 GMT

4PSA announces its participation to ITEXPO West 2012 in Austin, Texas, October 2-5, where the company will be offering a unique preview of its latest generation Unified Communications solution, VoipNow 3 Service Provider Edition.

"ITEXPO is a playground for outstanding products, new technologies and ideas and 4PSA is bringing its contribution to innovation in the field of Unified Communications. Even from Beta stage, it is clear that VoipNow 3 Service Provider Edition represents a massive step forward for how providers will be doing business from now on," said Mike Ross, President of 4PSA.

Featuring a completely distributed architecture and enriched web interface, a new platform for Apps, support for Amazon S3 cloud storage, integration with Microsoft Lync, the solution can successfully handle even the most demanding service provider environments and end-user requirements. "VoipNow 3 Service Provider Edition will be the greatest release yet in 4PSA's history and I'm sure visitors will be excited to hear about the unique features we implemented" added Mike Ross.

Learn more about VoipNow 3 Service Provider Edition (Beta) on our website at http://www.4psa.com/products-voipnow-voipnow3beta.html or visit our newly introduced Get Satisfaction community at http://my.4psa.com.

AstriDevCon Details Announced; Registration Opens for October 22, 2012 Conference

VoIP Monitor — Fri, 31 Aug 2012 20:29:44 GMT

Digium announces details of its annual AstriDevCon conference, a day-long event scheduled for October 22, 2012 at the Sheraton Atlanta Hotel. Held alongside the annual AstriCon conference, AstriDevCon brings together the development community that continues to make Asterisk the most widely used open source software for creating business phone systems and other communications applications.

AstriDevCon provides an in-depth review of progress made in the past year and an open discussion about the future direction of the Asterisk project. The event offers an opportunity for attendees to meet the core development team in person, pitch ideas for new features and functions and coordinate efforts with others.

AstriDevCon is free for active Asterisk developers who complete a registration form and are accepted to attend. For details, and to access the registration form, visit: https://wiki.asterisk.org/wiki/display/AST/AstriDevCon. Capacity for this event is limited, so participants are encouraged to register early.

"When it comes to Asterisk development, AstriDevCon is one of the most productive events there is. Each year, the energy, passion and knowledge that is shared between Digium and community members sparks great new ideas and approaches," said David Duffett, director, worldwide Asterisk community. "We are eager to meet face-to-face with some of the 65,000 members who help make Asterisk the rock-solid foundation for the world's most advanced communications systems."

AstriDevCon is the Monday before the AstriCon User Conference & Expo, a three-day conference and exhibition designed to expand awareness and knowledge of Asterisk through education, exhibition and networking. Registration for AstriCon 2012 is open now on the official event site .

Voiceserve to Exhibit at ITW2012 and Present Its Latest Voipswitch Mobile Diallers

VoIP Monitor — Tue, 08 May 2012 20:53:12 GMT

Voiceserve announces that the Company's management team will attend the ITW telecom show in Chicago, from May 14-16, 2012.

Voiceserve representatives will be on hand (Exhibit Booth 705) to meet with VoIP service providers and explain the benefits of its latest Mobile Diallers known as Vippie 2, an application which can be installed on most smart phones and tablets. The latest dialler gives a new dimension to communication via 3g and wifi connectivity.

"We are looking forward to returning to the conference - and be able to tell the continuing story of Voipswitch's latest release. ITW is the perfect arena to enhance established relationships and generate new business especially with world renown carriers," said Voiceserve CEO Mr. Michael Bibelman.

AudioCodes Executives to Speak on Enterprise Connect SIP Trunking Tour

VoIP Monitor — Tue, 01 May 2012 22:19:41 GMT

AudioCodes announces that two of its executives plan to speak on the Enterprise Connect SIP Trunking Tour, a four-city tour promoting SIP Trunking services, education and technologies. As part of the event series, Alan D. Percy, Senior Director of Marketing for North America and Larry Clarkson, Chief Technology Office for North America will participate in panel discussions, offering real-world customer experiences, best practice advice, and tips when migrating from legacy TDM trunking to SIP Trunking services.

SIP Trunks are becoming more widely available and have tremendous potential—they can reduce enterprise costs for carrier services, as well as getting your enterprise one step closer to supporting true Next-Generation IP Communications. In response to the booming demand for SIP Trunks—and for information about SIP Trunks—Enterprise Connect is launching a four-city “road show” on this vital topic.

Enterprise Connect SIP Trunking Tour Locations and Dates:

May 9 :: Las Vegas :: Mandalay Bay
May 16 :: New York City :: American Conference Centers
May 22 :: San Francisco :: Grand Hyatt
June 6 :: Chicago :: Hyatt Regency

The program will feature an intensive day-long series of sessions and networking opportunities designed to help enterprise decision-makers understand the market for SIP Trunking and the technologies that drive it. That includes the following key topics:

SIP Trunking ROI – What We Really Know?
Technical Challenges – What’s Needed to Make SIP Trunks Work?
What’s SIP Trunking’s Role in Overall Architecture?
Do’s and Don’ts for SIP Trunking – Case Studies

In addition to these information-packed sessions, attendees be able to bolster their understanding of SIP Trunking by meeting informally with other colleagues at the tour event, during networking breaks, lunchtime, and an end-of-day cocktail reception. Attendees will also have the chance to meet with a number of AudioCodes executives and channel partners.

More information and registration for the event can be found here.

A special registration discount is available for pre-registration using the discount code “AUDIOCODES”

Worldwide Service Provider Technical Community to Converge at SIPNOC US 2012

VoIP Monitor — Mon, 30 Apr 2012 22:41:55 GMT

The SIP Forum announced a preliminary list of registered attendees and participants at the second SIP Network Operators Conference (SIPNOC US 2012), to be held June 25-27 at the Hyatt Dulles Hotel in Herndon, VA. The two-day conference, which focuses on the challenges and opportunities related to the deployment of SIP-based services in global service provider networks, is attracting technical leadership from MSOs and carriers from North America, South America and Europe including ADP, babyTel, Broadvox, Comcast, Cbeyond, COX Communications, Lumos Networks, iBasis, IntelliVerse, Socket Telecom, Sorenson Communications, Time Warner Cable, TDS Telecom, Telefonica International Wholesale Services, XO Communications, Uni-tel, Verizon and Vocalocity.

In addition to carrier participants, SIPNOC US 2012 has also attracted a myriad of SIP community stakeholders from vendors, governments and research organizations such as Acme Packet (which has also signed on as a SIPNOC US 2012 Platinum Sponsor), Avaya, the FCC, CableLabs, Commetrex, Dialogic Corporation, GENBAND, Georgetown University, Illinois Institute of Technology, ISOC, Polycom, Sangoma Technologies, Siemens, and Sonus Networks.

Registration for SIPNOC US 2012 remains open by visiting http://www.regonline.com/sipnocus2012.

"SIPNOC was conceived as a meeting place for network operators to come together to share their deployment experiences and ideas on how to make IP communications better, and provide tangible results that further accelerate the development and worldwide propagation of SIP and SIP network services," said SIP Forum President and Managing Director, Marc Robins. "We expect SIPNOC US 2012 to pick up where last year's inaugural event left off and be far more than just an educational conference, but a meeting of the brightest technical minds from across the global networking operator community."

The SIP Forum has also developed an impressive conference agenda and lineup of speakers at SIPNOC US 2012, to be announced in detail shortly.

SIPNOC US 2012 will focus on issues critical to the reliable and successful deployment and operation of SIP-based services in carrier networks. The agenda will feature special presentations, panel discussions and workshops covering key topics by network operators related to SIP-based services and infrastructure, including testing, application development, SIP trunking, FoIP, call routing and peering, IPV6, SIP security, troubleshooting and monitoring, emergency services and more.

Attendees at SIPNOC US 2012 will include telecommunications providers, major backbone operators, interconnect and wholesale solution providers, ISPs, cable operators, wireless network operators as well as large enterprises deploying major SIP initiatives. While the international carrier and service provider community is the primary focus at SIPNOC US 2012, industry stakeholders involved in major SIP initiatives such as network equipment vendors, government agency representatives, large enterprise network operators and academic research organizations are also encouraged to attend.

The SIP Forum has gained an international reputation for developing important, educational events surrounding SIP. The organization's SIPit series of interoperability testing events regularly provides a test bed for SIP-based applications and equipment that has been heralded as critical for the development of new products and services in the industry. The SIP Forum also has a number of committees and task groups made up of well-known industry experts examining a myriad of SIP-related topics, including the use of SIP in smart grid installations, FoIP, video and user-agent configuration.

SIPNOC US 2012 KEY INFORMATION
Registration

To register for SIPNOC US 2012, please visit http://www.regonline.com/sipnocus2012

Digium Announces Ninth Annual AstriCon to be Held October 23-25, 2012

VoIP Monitor — Tue, 17 Apr 2012 21:30:58 GMT

Digium has kicked off planning for AstriCon, the Asterisk Open Source Conference and Exhibition. The event, now in its ninth year, will be held in Atlanta, Georgia from October 23-25, 2012 at the Sheraton Atlanta Hotel. Digium is the creator and corporate sponsor of the Asterisk project, the most widely used open source platform for creating custom communication solutions. Speaker topic submissions are open, and the conference organizers are soliciting talk concepts for 2012. Digium invites those who would like to speak at AstriCon to submit information for consideration by May 1, 2012 at http://www.astricon.net/2012/speaking.aspx.

With nearly two million downloads per year, millions of deployments and a community of more than 65,000 members, the acceptance and growth of Asterisk has spawned an ecosystem spanning more than 170 countries. AstriCon gives all members of the Asterisk community – from telephony enthusiasts to businesses – a forum to learn about the technology. Asterisk integrators and business end-users can expect to hear the latest Asterisk news and project updates, gain access to in-depth technical sessions, participate in networking opportunities, meet potential collaborators and review and discuss detailed case studies of Asterisk projects. The exhibition space will consist of more than 40 exhibitors including Aastra, AudioCodes, Grandstream, Jenne, OpenVox, Orecx, Sangoma, Vitality and XORCOM.

Asterisk is the most widely used open source software for creating business phone systems and other communications applications. Using it, developers and other technical pros craft solutions such as IP PBXs, VoIP gateways, interactive voice response systems, conference bridges, voicemail servers and more. Asterisk also forms the basis for Digium’s award-winning Switchvox Unified Communications solution, which offers the most advanced business phone system features in a cost-effective, easy-to-use solution that scales as companies grow.

“As Asterisk continues its phenomenal growth, Digium and the Community continue working to improve and enhance capabilities, as it did most recently with the release of Asterisk 10,” said Bryan M. Johns, community director at Digium. “Digium also continues to devote substantial resources to the development of both the Asterisk project and hardware that more tightly integrates with Asterisk, such as the new VoIP gateways, R-series failover appliance and Digium Phones.”

Digium is once again pleased to partner with Technology Marketing Corporation to promote the event to a broader audience. TMC has helped support other Digium events, including Asterisk World, with training sessions, video production, attendee registration and exhibit management. Companies interested in sponsoring AstriCon and participating on the EXPO floor should contact Joe Fabiano at TMC: +1 (203) 852-6800, ext. 132.

Registration for AstriCon 2012 is open now on the official event site: http://www.astricon.net. The early bird rate of $495 is available until August 1, 2012.

4PSA Showcases VoipNow at WHD.Global 2012

VoIP Monitor — Wed, 21 Mar 2012 01:12:44 GMT

4PSA announces its participation at the largest event of the hosting and Internet industry, WHD.Global 2012, taking place at Europa-Park in Rust, Germany, between 20 and 23 March.

During WHD.Global 2012, 4PSA will showcase VoipNow Platform, the award-winning Unified Communications software for service providers and businesses, as well as the recent integration of VoipNow with Parallels Automation. "By integrating these solutions, we're enabling service providers to easily deliver Unified Communications with business grade PBX functionality", said Bogdan Carstoiu, 4PSA's CEO. "As VoipNow's powerful charging engine has been integrated with Parallels Automation billing, providers can create flexible service plans for their resellers and end-users," added Mr. Carstoiu. More details about VoipNow's integration with Parallels Automation are available at www.4psa.com/poa/.

On March 22nd at 12:05pm, 4PSA's CEO will hold an insightful presentation on social media and its impact on the way people and organizations communicate today. The presentation entitled "Social Cannibalization of Communication Channels" will discuss strategies that service providers must adopt to monetize on the increasing demand for social communications. 4PSA team will be available at the company's booth to answer questions and help visitors gain a deeper understanding of Unified Communications.

4PSA's Unified Communications Platform, VoipNow offers a large number of features and benefits for service providers and businesses alike, including an advanced provisioning system, integration with many cloud services, voice, video, instant messaging, faxing, presence, business and operations automation. VoipNow's scaling capabilities, high-performance and ability to address real-market needs have earned the product major awards from the industry, including INTERNET TELEPHONY Magazine's 2011 Product of the Year Award and the Unified Communications Magazine's 2010 Product of the Year Award.

4PSA Showcases Cloud Unified Communications at ITEXPO East 2012

VoIP Monitor — Thu, 02 Feb 2012 21:59:58 GMT

4PSA announces its participation at ITEXPO East 2012, the World's Communications Conference and Expo. The event takes place on February 1-3, 2012 at the Miami Beach Convention Center in Miami, Florida.

During this year's ITEXPO East, 4PSA's President Mike Ross will be presenting in two highly engaging panels, exploring the increase in the adoption rate for Unified Communications as a Service and the synergies between Unified Communications and Social Media. "We're always excited to attend ITEXPO. This event represents a wonderful opportunity to get together with top providers and industry experts, discuss the latest advancements in the field, and share insight", stated Mr. Ross.

4PSA will showcase its award-winning Unified Communications solution, VoipNow Cloud OnDemand. "VoipNow Cloud OnDemand has established itself as the fastest and most convenient solution for service providers who want to offer Unified Communications to a increasingly growing number of users and for companies that want to implement VoIP inside their business," Mr. Ross added.

VoipNow Cloud OnDemand is a fully-featured and flexible turn-key solution that bundles high-performance infrastructure with the company's award-winning VoipNow® Unified Communications Platform. VoipNow Cloud OnDemand instances are available in both US and Europe.

SIP Forum Announces Dates for Second Annual SIPNOC

VoIP Monitor — Wed, 09 Nov 2011 22:32:10 GMT

The SIP Forum announced today that it will host the second annual SIP Network Operators Conference, a two day educational conference focusing on the challenges and opportunities related to the deployment of SIP-based carrier services globally. SIPNOC US 2012 will be held in Herndon, VA on June 25-27, 2012 and will build on the success of the inaugural event last spring, which attracted leading technical and operations personnel from the global carrier community and earned high praise from attendees for its educational, non-commercial and technical content that focused on the real-world challenges operators face when deploying SIP services in global IP networks.

The SIP Forum will release further details about SIPNOC US 2012 including sign-up for early bird registration in mid-November at its conference website http://www.sipnoc.org.

SIPNOC 2012 will bring together leading technical minds from the telecommunications industry to learn, discuss and formulate new ideas and strategies concerning the challenges and opportunities for SIP-based carrier services in fixed and mobile IP network environments. The conference, which is designed specifically for SIP network operations personnel and engineering staff, will feature well-known industry speakers and a number of highly technical educational and instructional panels and sessions. The SIP Forum will also host networking events at the conference and offer a series of informal “Birds of a Feather” sessions, which encourage discussion of varying topics held in hallways, available meeting rooms and break-out areas.

SIPNOC 2012 will build on themes first discussed at last year’s inaugural event: addressing issues critical to the reliable and successful deployment and operation of SIP-based services in carrier networks and the opportunities that come with it. Among the topics expected to be on the agenda at this year’s conference are SIP trunking interoperability and the SIP Forum’s recently ratified SIPconnect 1.1 technical specification that provides a definitive and standardized set of guidelines for seamless, end-to-end interoperability between SIP-enabled IP-PBXs and service provider networks. Other themes to be discussed include Fax over Internet protocol interworking, implementing SIP with IPv6, and the sharing of best practices for the utilization of Wireshark for network diagnostics within IP network environments.

Building on the success of these US-based SIPNOC events, the SIP Forum is expected to announce details for a Europe-based event - SIPNOC EU 2012. The dates and location of the next event will be released later this year.

SIPNOC US 2012 will be held at the Hyatt Dulles Hotel in Herndon, VA June 25 -27, 2012.

Panasonic Showcases Award Winning SIP Telephony Solutions at AstriCon 2011

VoIP Monitor — Tue, 25 Oct 2011 21:25:13 GMT

Panasonic is showcasing an extensive range of IP telephony solutions at AstriCon in Panasonic booth #204 at the Westin Westminster in Denver, Colorado, October 25-27.

As an AstriCon 2011 Gold sponsor, Panasonic is demonstrating a broad range of SIP telephones including SIP Cordless Phone Systems and KX-UT series which are interoperable with Asterisk's open source PBX. Panasonic's SIP telephones work with the Asterisk platform which offers both classical PBX functionality and advanced UC features.

In its seventh year, AstriCon is the longest running conference devoted to the Digium Asterisk communications platform. AstriCon brings together open source enthusiasts, from coders and system integrators to service providers and enterprise IT professionals, who are looking for an in-depth understanding of Asterisk open source technology.

Panasonic's SIP Phone Systems:

The Panasonic SIP Cordless Phone System is a small business communication solution that offers the flexibility of convenient expansion as a company grows. The KX-TGP500 system features a wall-mountable base unit and one cordless handset. Expandable up to six DECT 6.0 cordless handsets, the model supports up to eight phone numbers and three simultaneous calls. It boasts Wide Band Audio (G.722) and five hours Talk Time, 10 days Standby. Its elegant design features a white backlit large LCD on the handset and a Handset locator button on the base unit. It also has a handset speakerphone, 2.5mm headset jack and belt clip.

The KX-TGP550 model has all the features and benefits of the KX-TGP500 and adds a corded base unit with a large white backlit LCD, plus a Hands-Free Speaker phone, Handset Call Button on the base unit, and one-touch call transfer with Busy Lamp Indication.

Also on display, the Panasonic KX-UT series offers a cost-effective communications solution for businesses of all sizes that leverages the latest developments in Hosted and Open Source PBX technologies and is designed to complement a company's existing communication infrastructure. Most models feature two data ports so users can connect a second network device without the time and expense of running an additional Ethernet cable. The KX-UT series models are Power over Ethernet ready which eliminates the need for additional electrical adaptors. Wide-band, high-definition audio (G.722 codec) coupled with echo cancellation and an expanded acoustic chamber allows the KX-UT series to offer crisp sound quality for crystal clear conversation.

Panasonic is also previewing the new KX-UT670, a highly expandable corded SIP phone with a seven-inch color LCD touch screen function that will help to transform business communication. Additional key features include HD Voice (G.722), two Ethernet ports, 3-way conference calling, IP camera integration, full duplex speakerphone, 100 entry phonebook and PoE ready.

Voiceserve to Showcase Its Leading Edge VoIP Software Platform at DEMO Fall 2011

VoIP Monitor — Mon, 12 Sep 2011 20:31:00 GMT

Voiceserve has been selected to present its VoIP Video-on-Demand software at DEMO Fall 2011. The Company's VoIP software is capable of streaming VOD on any smart-device, tablet, cell phone and computer from anywhere in the world. Companies are hand-selected by the conference's executive producers to present alongside this year's most promising technological solutions.

The conference, which takes place September 12-14, 2011 at the Hyatt Regency Santa Clara in the heart of Silicon Valley, has a 20 year history of introducing the tech community to the people and companies who are paving the way to the future.

LiveVox to Discuss Contact Center ''Cloud Evolution'' at ITEXPO West

VoIP Monitor — Fri, 09 Sep 2011 21:22:08 GMT

LiveVox announces that Chief Executive Louis Summe will join a panel on “The Cloud Evolution in Contact Centers” at ITEXPO West next week in Austin.

As co-founder and CEO, Summe has led the development of a highly scalable cloud contact center infrastructure over the past decade. LiveVox integrates core contact center applications like dialer, ACD, PBX IVR, and call recording. The PCI-certified platform offers real-time scale and a simpler network topology and faster voice/data transfers.

Summe will discuss how to implement cloud contact centers in a secure and manageable way, while solving historic contact center problems like integration, deployment and seasonal variability.

What: “The Cloud Evolution in Contact Centers”
Who: Louis Summe, Chief Executive Officer, LiveVox
When: Noon, CDT, Thurs., Sept 15
Where: ITEXPO West, Austin Texas

LiveVox will also be exhibiting at ITEXPO West (BOOTH 303). Attendees can stop by for more information about leveraging the Cloud in their call center.

4PSA's Cloud Instance to Take Center Stage at ITEXPO

VoIP Monitor — Fri, 09 Sep 2011 20:50:19 GMT

4PSA will be presenting the latest enhancements to their VoipNow Unified Communications Platform at ITEXPO West 2011, held at the Austin Convention Center, September 12th-15th.

VoipNow Cloud Instance solution offers all of the necessary infrastructure and software for service providers or businesses aiming to instantly provide multi-tenant Unified Communications to any number of users. "We think that this is a game changing moment for our industry" said Mike Ross, President of 4PSA. "Without any contract commitment, on a pay-as-you-go basis, a company can now deliver any combination of UC features to an unlimited number of users on the very same day," Mr. Ross also stated.

"The benefits for a service provider or business are enormous because the VoipNow Cloud Instance eliminates one of the biggest headaches that Service Providers or Business clients are constantly dealing with: the need to select, plan, purchase, manage, and maintain the servers for their hosted UC software. Along with the VoipNow Cloud Instance come all the advanced features of VoipNow Professional and VoipNow Automation. UC features include a highly scalable carrier class multi-tenant IP-PBX with VoIP calling, fax, conferencing, Instant Messaging, ACD Queues, Auto-Attendant and IVR, Video calling, Presence Management, auto provisioning for phones, and over 100+ features to choose from. Integration with other cloud-based applications such as CRM is easily achieved. And that's not all. Fully supported APIs also give each customer the ability to shape VoipNow based on their users' requirements," Mr. Ross added.

The VoipNow Cloud Instance is available in 4PSA Clouds in North America and Europe and can be purchased directly from 4PSA's website. Free 30-day evaluation trials are available for qualified companies.

Fonality Announced as Newest Platinum Sponsor of ITEXPO

VoIP Monitor — Mon, 20 Jun 2011 21:41:40 GMT

TMC announced that Fonality, North America’s fastest growing business communications company, has signed on to become a platinum sponsor of their ITEXPO conference and trade show, September 13-15, at the Austin Convention Center in Austin, Texas.

ITEXPO is the world’s largest and best-attended communications and technology trade show. ITEXPO West is currently ranked as the No. 3 fastest-growing event on Trade Show Executive Magazine’s Fastest 50 List. TMC expects to accelerate ITEXPO West’s growth by bringing the show to Austin, one of the fast-growing technology hubs in the U.S.

“ITEXPO is the industry’s leading venue for cloud-based and open source telephony solutions for enterprises and SMBs,” said Fonality CTO Rick Bushell. “We’re looking forward to interacting with the thousands of attendees who come to ITEXPO to show how Fonality’s solutions can help small- to mid-size businesses communicate smarter, faster and more affordably.”

Designed for small and mid-size companies, Fonality offers VoIP, Unified Communications and contact center solutions that are easy to use, simple to manage, and affordable to deploy. Fonality’s cloud-based model provides advanced features and services without the costly hardware, infrastructure or lengthy implementation cycles offered by legacy providers.

Total cost of ownership is dramatically reduced by 50 percent, or more, while users enjoy access to powerful communications capabilities typically associated with Fortune 500 firms. Fonality’s Head’s Up Display UC dashboard offers presence based capabilities to seamlessly help manage voice, e-mail and chat dialogue and ensure users can reach the right person, with the right information, every time.

“Fonality serves a large and rapidly-growing community of users, who come to ITEXPO to learn more about the latest advancements in the industry,” said Rich Tehrani, CEO and conference chairman for TMC. “Fonality has made significant contributions to ITEXPO over the years, and we’re looking forward to working together once again at ITEXPO in Austin.”

Registration for ITEXPO is now open.

Voiceserve to Present at CommunicAsia2011 Conference

VoIP Monitor — Fri, 17 Jun 2011 16:37:04 GMT

Voiceserve will be presenting its industry leading, comprehensive VoIP software and solutions platform this month at the CommunicAsia2011 International Communications & Information Technology Exhibition and Conference from June 21-24 in Singapore.

CommunicAsia2011 is Asia's largest ICT based event. Last year's event attracted over 38,000 industry professionals from over 10 Asian countries such as India, China, Japan, Indonesia and Korea, and provided 37 buyer-groups with 1,300 exhibitors. CommunicAsia2011 will feature emerging technologies, such as the latest innovations in VoIP and the latest advancements in Video-on-Demand.

Digium Welcomes Open Source Software Pros to AstriCon Developer Days

VoIP Monitor — Wed, 01 Jun 2011 17:26:45 GMT

Digium invites software developers and others who have made contributions to Asterisk and Asterisk Scalable Communications Framework open source platforms to participate in AstriCon Developer Days. This series of developer conferences is planned for the days prior to and following the annual AstriCon event. The Asterisk conference will be held on Monday, October 24 and the Asterisk SCF meeting will take place on Friday, October 28. AstriCon, the Asterisk User Conference and Expo, will run Tuesday through Thursday of that week at the Westin Westminster in Denver, Colorado.

Developers who have contributed code to the two open source projects headed by Digium may attend these special events at no cost beyond the AstriCon registration fee. The individual conferences will give developers the opportunity to help define, in interactive forums, the direction of each project. Additional information about the events, including AstriCon registration and speaking opportunities, is available at http://www.AstriCon.net. Early bird rates for the conference are available through July 10, 2011.

“Asterisk has long drawn its strength from its energetic community, which includes hundreds of active developers,” said Bryan M. Johns, community director at Digium. “We receive continuous feedback from developers and other enthusiasts, but want to invite and encourage involvement in the events that we’re creating specifically for them. We think the open forum and ability to receive instant feedback and reaction to suggestions and comments will be invaluable for all participants.”

Asterisk is the most widely used open source software for creating business phone systems and other communications applications. IT pros, telephony aficionados and software developers have downloaded Asterisk millions of times since its creation in 1999 and are using it in nearly every country of the world. Digium announced Asterisk SCF at AstriCon 2010 to enable the creation of real-time communications applications that include voice, video and text and that meet the demands of embedded applications to enterprise and carrier solutions. Asterisk SCF is designed to provide the highest levels of availability, scalability, extensibility, fault-tolerance and performance.

NextAlarm Launches New VoIPAlarm Marketing Campaign

VoIP Monitor — Wed, 01 Jun 2011 15:51:01 GMT

NextAlarm announces the launch of an extensive marketing campaign targeted at the alarm industry. The company's VoIPAlarm brand, focused on sales to alarm companies, will receive a push from well-known industry veterans Tom Christ and Terry Gurley. A series of biweekly webinars has begun to educate dealers about the product.

Christ and Gurley, with over twenty collective years of experience in the alarm industry, have previously built dealer networks of over 800 companies within 18 months. They specialize in high-tech products in the security space and are expert in training alarm dealers to identify, sell to, and support customers of technologies like VoIPAlarm's alarm broadband adapter.

VoIPAlarm was designed as a solution for alarm dealers and monitoring services losing customers due to the advent of VoIP telephone services. The majority of alarm systems installed in homes and businesses today are unable to reliably communicate over VoIP, as they were designed with the specific tolerances and DTMF ranges of standard PSTN phone lines. VoIPAlarm allows these alarm systems to communicate over broadband with the addition of a single, inexpensive adapter, called the broadband IP adapter. The adapter also works to re-enable alarms in locations that have disconnected their phone lines entirely, such as homes whose residents have switched to using cell phones. Unlike other broadband alarm products, VoIPAlarm is compatible with most alarm systems built in the last 20 years, and functions over any broadband Internet connection. Also unique to VoIPAlarm is support for two-way voice communication between a central station operator and an alarm owner, a feature common in medical alert systems and higher-end security systems.

VoIPAlarm also offers the IP video product NextView, which currently includes a fixed camera and a movable camera with pan and tilt, with an outdoor camera available soon. NextView cameras feature customized firmware that allows the NextView service to react to any alarm event. When an alarm panel reports an event through the VoIPAlarm platform, the system provides the customer and authorized dealer with video both prior to and after the alarm event. Using a proprietary buffering technology, the subscriber or dealer can view the seconds leading up to an alarm, as well as video taken after the event occurred.

Webinars are held twice a week to show VoIPAlarm to alarm dealers and other interested parties. Special introductory pricing is available exclusively through these webinars. To register for a free session, visit http://www.voipalarm.com/webinar.

Tone Software Presents VoIP QoS Session at IAUG Global Conference to Help Companies Address Convergence Challenges

VoIP Monitor — Thu, 19 May 2011 19:13:21 GMT

Tone Software will be on hand at the upcoming IAUG Global Conference. Held from May 22-26 at Caesars Palace in Las Vegas, the IAUG Global Conference is one of the world's largest international gatherings for communications technology professionals.

Supporting a converged voice and data environment and working with VoIP quality metrics can be a daunting task. Understanding which metrics to collect and analyze, their interdependencies, and how they manifest into user experience is key to successfully delivering VoIP quality and converged voice services. To that end, in a session titled, "The Language of VoIP QoS - Demystified," Tone's director of strategic technology advancement, Amit Kapoor, will help take the guesswork and unknowns out of supporting VoIP. Taking place on Thursday, May 26th at 9:45am, the session focuses on the terminology, impact, and utilization of QoS metrics to successfully support a VoIP environment.

The IAUG Global event provides a valuable opportunity for both managed service providers and enterprises responsible for delivering VoIP QoS and voice service levels in Avaya voice environments to meet Avaya DevConnect Partners such as Tone Software, and explore solutions to address their convergence challenges.

Tone's ReliaTel VoIP QoS and Converged network management technology is used in many Avaya-centric environments to manage Avaya VoIP QoS through direct analysis of Avaya RTCP data, as well as managing the entire underlying physical communications infrastructure. ReliaTel is platform-agnostic and fully supports the full gamut of Nortel communications technologies, as well as many other IT technologies, to provide comprehensive management of the entire communications ecosystem in a unified solution -- regardless of the technology mix.

Tone Software will be providing live ReliaTel demonstrations in booth #716 on the IAUG show floor.

Digium Flips the Switch on AstriCon 2011, the Event Devoted to Open Source Asterisk

VoIP Monitor — Mon, 09 May 2011 17:45:41 GMT

Digium is making plans for the eighth annual AstriCon, the Asterisk User Conference and Expo and the event of choice for hundreds of technical pros working with the world’s leading open source communications platform. This year, from October 26-28, AstriCon will come to the Westin Westminster in Denver, Colorado. In addition to opening registration today, Digium is now accepting submissions for speaker opportunities at www.AstriCon.net. Topics are open, but past sessions have focused on basic and advanced Asterisk tutorials, IP telephony security, call center and enterprise case studies, and product training.

Asterisk is the most widely used open source software for creating business phone systems and other communications applications. Using it, developers and other technical pros craft solutions such as IP PBXs, VoIP gateways, interactive voice response systems, conference bridges, voicemail servers and more. Asterisk also forms the basis for Digium’s award-winning Switchvox Unified Communications solution, which offers the most advanced business phone system features in a cost-effective, easy-to-use solution that scales as companies grow. The Asterisk community includes more than 65,000 members worldwide, hundreds of whom have made substantial contributions to the software’s development since its release more than 10 years ago.

In addition to AstriCon, Digium will hold developer conferences on Monday, October 24 and Friday, October 28 for Asterisk and Asterisk SCF, respectively. These two special events are open, at no cost beyond the AstriCon registration fee, to any developer who has contributed code to the Asterisk or Asterisk SCF projects. The conferences give contributors an opportunity to help define, in an interactive group of their peers, the direction of each project.

Broadvox 2011 Partner Summit is Underway

VoIP Monitor — Wed, 04 May 2011 19:03:47 GMT

Broadvox opened the 2011 Partner Summit today in Dallas. Sales channel partners and OEMs attended to share information about new products and sales positioning and to learn about the new Broadvox cloud solutions.

The Summit event opened with an Executive Round Table where the CEO of Broadvox and Cypress Communications, Andre Temnorod, and EVP Sales and Marketing, David Byrd, welcomed over 150 participants and discussed the evolution of Broadvox, the SIP Trunking Industry and its role in delivering Cloud Communication solutions. The session was moderated by Rich Tehrani, the CEO of Telecom Marketing Corporation, the only publishing and tradeshow company focusing exclusively on the rapidly growing voice/data convergence and on computer telephony integration. Rich Tehrani is a VoIP industry expert, visionary, author and columnist.

Broadvox announced GO!VBX Managed, a fully managed hosted VoIP and PBX solution offering QoS from the handset to the cloud. The GO!VBX Managed product includes LAN Services and routers, as well as optional phone leasing, at one fixed monthly price.

The GO!VBX solutions delivers a new pricing paradigm to the hosted IP PBX marketplace. By separating the per seat cost of the solution from the communications usage, businesses save as much as 45% over other hosted offerings. GO!VBX is positioned for the Broadvox VARs to add to their product and services portfolios. GO!VBX Managed can be represented by both the Broadvox Partner Channel and the Cypress direct sales channel.

Partner presentations Wednesday featured speakers from Digium, BroadSoft and Cisco.

The day ended with a summit networking event headlined by special guest speaker Roger Staubach ending with a Casino night and silent auction of Staubach autographed memorabilia.

Top Industry Executives to Attend CommuniGate Systems’ Summit in Berlin

VoIP Monitor — Wed, 20 Apr 2011 19:57:29 GMT

CommuniGate Systems announces the line up for its Annual Executive Summit in Berlin.

The summit takes place in Berlin, May 25th-27th and focuses on bringing together influential telecoms leaders to share and exchange innovative strategies for the future success of the telecoms sector, including:

Robert Mane: Chief Executive Officer, Cablegroup
Paul Wade: Chief Executive Officer, SmarterMobile
Georges-Pierre Selmer: Multimedia and Content Services Manager, NRJ Mobile
Jos de Kruijf: Owner dK Consult and Executive Advisor to Miyowa and eBuddy
Dr Ulrich Hammerschmidt: Vice President Innovation Projects, Deutsche Telekom, International Carrier Sales & Solutions
E. Brent Kelly: Senior Analyst and Partner, Wainhouse Research

Building on the acclaimed success of last year’s summit, the event invites senior executives from the industry to talk about how brand, superior quality and mobile apps in the cloud will help operators avoid marginalization by the over-the-top providers.

Topics covered by the guest speakers include:

Silicon valley startups dominate. What can telcos do to get back in the game?
Controlling the end-to-end delivery chain. What we can learn from Apple?
Voice revenues under threat. HD Voice breathes new life into the voice market
Your branded button. Deliver your own over-the-top applications in the cloud
Mobile Cloud: Customers trust you with their data, not Google
Mobile Broadband benefits the internet companies, dilutes operator brand and value

The summit’s topical agenda and first-class speakers are drawing significant interest from across the whole industry, with attendees coming from mobile, cable, fixed-line and virtual network operators. Operators already confirmed include T-Mobile, NRJ Mobile, Mobily, TelcoMobile, TeleColumbus and others.

4PSA Welcomes Partners and Customers to WorldHostingDays 2011

VoIP Monitor — Thu, 17 Mar 2011 14:43:53 GMT

4PSA announce its participation as an event partner to WorldHostingDays 2011, the world's leading hosting gathering. WHD 2011 will take place between March 22-25, 2011 at Europa-Park, one of the largest theme parks in Europe, located in Rust, Germany.

Previously known as WebHostingDay, with over 2,800 visitors from 73 countries in 2010, the four-day conference and trade show will bring together companies involved in the hosting business from all around the world. "Our previous experience with WebHostingDay has been very good and we are eagerly waiting to make it a success this year as well. 4PSA will be exhibiting at booth #74 where we will be offering live demos for visitors interested in gaining deeper understanding of Unified Communications and the key benefits it brings to service provider offering," said Mike Ross, 4PSA's President.

In 2011, 4PSA will be launching paradigm shifting products and services designed to bring people closer to a total Unified Communications hub. "On March 24, 4PSA's CEO and Chief Architect, Dr. Bogdan Carstoiu, will hold a 45-minute presentation covering the opportunities service providers can address by expanding the way their services integrate with new communication flows such as social networks," also added Mike Ross.

At WorldHostingDays 2011, 4PSA will showcase a full range of products including VoipNow Platform, an award-winning solution designed to accelerate Unified Communications adoption, and DNS Manager, the multitenant, user-friendly automation solution that delivers advanced DNS hosting to service providers and businesses. The company will also offer exclusive promotions to WHD 2011 visitors.

Cebit 2011 – New VoIP and 3G Gateways

VoIP Monitor — Fri, 25 Feb 2011 20:25:17 GMT

German Distributors Allnet, Herweck, Komsa and Michael Telecom will be displaying information about TELES products at Cebit (Hanover, Germany 01.03.2011 till 05.03.2011). The spotlight is on products from the Access Gateways business division: VoIP-ISDN gateways in the VoIPBox product line, compact GSM and 3G gateways in the ECOTEL product line and 19 inch rack system mobile radio gateways in the iGATE product line. New for specialized dealers and system houses: the VoIPBox product line and the 3G gateways in the ECOTEL product line.

VoIP-ISDN gateways for any hybrid infrastructure
Companies of all sizes can use VoIP-ISDN gateways in the TELES VoIPBox product line to implement almost any hybrid infrastructure using ISDN and VoIP components. For example, they can connect already existing ISDN telecommunications systems to SIP trunks, or integrate ISDN-based fax servers into new VoIP telecommunications systems. With this in mind, VoIPBox gateways provide four to 180 VoIP channels, as well as two to eight BRI interfaces or one to six PRI interfaces. Thanks to the VoIPBox, the ISDN settings are left unchanged when an existing ISDN telecommunications system is connected to a VoIP outside line. In addition, VoIPBox's auto-configuration means gateways can be commissioned via "plug and play". No field technician or manual set-ups are needed.

Appealing to specialized dealers thanks to user friendliness and fair prices
"VoIP-ISDN gateways in the VoIPBox product line cover all possible combinations of VoIP and ISDN. They can be easily configured by technicians and are offered at attractive prices" explains Elke Kürschner, DACH Sales Manager at TELES. "They are particularly interesting to specialized dealers because of their high degree of user friendliness, and fair prices and margins." List prices for the entry-level model with two BRI and four VoIP ports start at 411 euros.

Mobile radio gateway portfolio with new 3G gateways
TELES partner distributors will also provide information on their stands about the new TELES ECOTEL 3G gateways which have two or four UMTS channels for speech and data transmission up to HSDPA speeds. TELES ECOTEL 3G gateways use mobile radio to connect equipment to the telecommunications networks where there is no fixed-network connection, for example on construction sites, in business vehicles and on inland waterway ships. Depending on the model, ECOTEL 3G gateways are equipped with all popular interfaces: FXO, FXS, BRI and VoIP. TELES also supplies a complete portfolio of compact ECOTEL GSM gateways with one to four channels, including ECOTEL GSM classic gateways, a product line purchased from Vierling in 2010, and the modular TELES iGATE mobile radio gateways with up to 32 channels in a 19 inch rack. TELES ECOTEL GSM gateways and TELES iGATE systems lower connection costs to mobile radio networks and offer fallback solutions when the network fails.

Contact persons at the trade fair
The stands from Allnet, Herweck, Komsa and Michael Telecom are managed by competent staff who will help specialized dealers and companies select the right TELES access gateways and answer basic questions about usage and installation. Specialized dealers are welcome to organize appointments at Cebit with the TELES sales team in advance.

NetIQ Puts Latest UC Deployment and Management Solutions Through Paces At UC Expo 2011

VoIP Monitor — Wed, 23 Feb 2011 16:05:39 GMT

NetIQ will be demonstrating its NetIQ VoIP solutions at Unified Communications Expo, 6th to 9th March, Olympia, London. Visitors to the NetIQ stand 322 will be updated on the latest developments in how NetIQ solutions assure the infrastructure is available and secure for successful IP telephony deployments, as well as monitoring the end user experience and call quality.

At UC Expo, NetIQ will focus on how IT telephony management needs to evolve to keep pace with the complex and challenging operational issues of unified communications. This includes how IT organisations responsible for unified communications can extend their current system management systems to automate more of the routine UC management tasks, freeing up valuable resources and ensuring problems are more efficiently resolved.

NetIQ Logo

On the stand, NetIQ will demonstrate how NetIQ Aegis can be utilised to control and automate unified communications operations. Alongside this solution, there will be demonstrations of other NetIQ UC solutions including:

NetIQ AppManager for VoIP for maximising the performance and availability of IP telephony systems and applications
Vivinet Assessor for determining quickly and easily how well VoIP will work on a network prior to deployment
Vivinet Diagnostics for pinpointing call quality problems in VoIP networks and helping to explain incidents of reduced call quality, and automatically collecting data needed to remedy the problem

ASC to Exhibit VoIP Recording and Quality Management Solutions at UC Expo 2011 in London

VoIP Monitor — Thu, 17 Feb 2011 15:53:12 GMT

ASC will demonstrate its VoIP recording solution, EVOip, and quality management solution, INSPIRATIONpro, at the Unified Communications Expo 2011 on March 8-9, at the Olympia Exhibition Centre, London.

Widely considered as the UK’s leading business communications event, with more than 4,000 visitors in 2010, this year’s exhibition is divided into six technology tracks: voice, cloud, mobile, visual, collaboration and customer. ASC will focus on its quality, process and campaign management capabilities, with particular reference to its speech analytics application to customer interactions.

ASC’s speech analytics application uses keyword spotting to help categorize calls for high-volume contact centres, with an otherwise unmanageable number of conversations. Other features of speech analytics such as emotion detection help to detect problem calls, which identify customer needs and improve agent training.

EVOip captures telephone calls from the network and enables storage, playback and archiving of the entire interaction. The software offers the strictest adherence to security requirements, meeting the payment card industry’s PCI DSS standards.

INSPIRATIONpro helps call centre managers learn about their agents’ service level through analysis and evaluation of recorded call data and screen activities. It facilitates agent evaluations through the recording of coaching sessions and allows complex searches of audio analytics.

ASC extends an invitation to any interested parties to visit its Exhibition Stand 304 at UC Expo’11, to discuss projects which may require the use of its technologies.

Media5 to Showcase Media5-fone at Mobile World Congress 2011

VoIP Monitor — Thu, 10 Feb 2011 18:51:17 GMT

Media5 will attend the Mobile World Congress 2011, a premier industry event held in Barcelona, Spain, on 14 – 17 February 2011.

The company will showcase its latest SIP-based mobile softclient, the Media5-fone, available for Apple iOS devices (iPhone, iPad and iPod Touch) and Nokia S60 3rd Edition smartphones. Media5's integrated technology enables service providers to leverage their mobile VoIP offering with new services or as a valued-added feature to their IP-Centrex, Hosted IP-PBX and SIP Trunk portfolio of solutions.

In addition, Media5 will outline its vision of mobile softclient evolution with a preview of the Media5-fone v2.7, including a demonstration of Video over IP calls, and a preview for Android devices.

4PSA Demonstrates Cloud Unified Communications at ITEXPO East 2011

VoIP Monitor — Thu, 03 Feb 2011 18:27:14 GMT

4PSA is exhibiting as well as participating on an educational panel at ITEXPO East held in Miami, Florida, Feb. 2-4, 2011.

The latest release of VoipNow Professional, version 2.5.1, extends 4PSA's UC suite with higher performance and more integration using the open API. Each release of VoipNow re-enforces its internationally recognized reputation as the premier software platform for CLOUD CALLING. Fixed-line or Mobile applications can be easily customized and managed using web services. Applications utilizing all forms of voice, video, fax, IM, conferencing, and collaboration are supported, fast and easily.

In addition to presenting the latest product release at the show, 4PSA will also be launching the new partner program for Certified SIP Trunk and Cloud Infrastructure providers.

VoipNow is a software suite designed to simplify the deployment of Unified Communications services in the cloud. The suite includes Professional, Automation, and Core - three separate components that allow service providers to automate the delivery of a wide range of IP communication services.

Patton and Partners will Promote VoIP Buyback Program at ITEXPO

VoIP Monitor — Tue, 01 Feb 2011 19:02:59 GMT

Patton joins key strategic partners at ITEXPO East this week to exhibit SmartNode VoIP equipment and promote Patton’s VoIP Buyback Program offering cash rebates for other manufacturer’s VoIP gateways, routers and Integrated Access Devices replaced with SmartNode.

Delegates can meet Patton representatives with partners ABP Technology (booth 322), VoipStore (booth 701), Elastix (booth 1009) and others. ABP will also be distributing coupons redeemable during 2011 for free technical training in Patton’s three-day VoIP certification course.

During recent months, Patton’s operations have steadily gained momentum in the VoIP and Unified Communications sectors as Patton has focused on strengthening relationships with key business and technology partners in the Americas and worldwide.

Most recently, Patton added EHCP certification for proven interoperability with Elastix’iPBX to the extensive listing of Patton-certified SmartNode™ technology partners.

Patton’s business partner, VoipStore, has produced a library of video tutorials featuring Patton’s SmartNode 4960 PRI IAD and SmartNode 4110 Analog FXS/FXO gateway in joint solutions with 3CX, another Patton partner.

Last year Patton joined ABP on the master distributor’s IPSizzles tour and multiple trade shows jointly marketing SIP trunking solutions that bundle SmartNode with such Patton-certified solutions as Broadvox, Asterisk, 3CX and others. ABP currently offers an enterprise-class SIP Trunking package--simple and quickly-deployed—that features SmartNode gateways integrated with traditional and/or IP PBXs.

Junction Networks’ CEO To Present at IT Expo East

VoIP Monitor — Wed, 26 Jan 2011 18:33:37 GMT

Junction Networks announces that Michael Oeth, the company’s CEO, will be presenting at this year’s IT Expo East, taking place in Miami, Florida, February 2 - 4, 2011. He will take part in a panel discussion on Hosted vs. On-premises VoIP, an ongoing debate within the industry as to which is the preferred choice for business customers.

IT Expo is a two-day event addressing all aspects of IP Communications. Michael Oeth’s session will take place on Friday, February 4th from 11:00 - 11:45 AM at the Miami Convention Center.

The SIP Forum Launches SIPNOC

VoIP Monitor — Tue, 11 Jan 2011 18:38:35 GMT

The SIP Forum announces the SIP Network Operators Conference (SIPNOC), a new annual conference for the international service provider community. The two-day conference will bring together the leading technical minds from the telecommunications industry to learn, discuss and formulate new ideas and strategies concerning the challenges and opportunities for SIP-based carrier services in fixed and mobile IP network environments. Unlike other events, this conference is for SIP network operations personnel, such as NOC engineers, instead of a high-level conference for executives. The first SIPNOC 2011 conference will be held at the Hyatt Dulles Hotel in Herndon, VA April 25 -27, 2011.

SIPNOC will focus on issues critical to the reliable and successful deployment and operation of SIP-based services in carrier networks. The agenda will feature special presentations, panel discussions and workshops covering key topics by network operators related to SIP-based services and infrastructure, including testing, application development, SIP trunking, FoIP, call routing and peering, troubleshooting and monitoring, emergency services and more.

Attendees at SIPNOC will include telecommunications providers, major backbone operators, interconnect and wholesale solution providers, ISPs, cable operators, wireless network operators as well as large enterprises deploying major SIP initiatives.

The SIP Forum has gained an international reputation for developing important, educational events surrounding SIP. The organization’s SIPit series of interoperability testing events regularly provides a test bed for SIP-based applications and equipment that has been heralded as critical for the development of new products and services in the industry. The SIP Forum also has a number of committees and task groups made up of well-known industry experts examining a myriad of SIP-related topics, including the use of SIP in smart grid installations, FoIP, video, and user-agent configuration.

For More Information and to RegisterFor more information about SIPNOC, please visit www.sipnoc.org.

A special early-bird registration rate of $595 – $200 off the regular rate of $795 -- is now in effect until February 11, 2011! To register for SIPNOC, please visit http://www.regonline.com/sipnoc_2011.

IMS Brings VoIP into Mobile Telephony - and with It All the Security Risks

VoIP Monitor — Thu, 18 Nov 2010 18:14:32 GMT

Codenomicon announces that Ari Takanen, CTO of Codenomicon, will give a presentation on threat assessment and proactive defense against VoIP vulnerabilities at the 3GPP ETSI Release 8 IMS Implementation Workshop. The event will take place 24-25 November 2010, at the Sophia Antipolis technology park, in southern France.

Ari Takanen will give his presentation "Recommendations for VoIP and IMS Security" on Thursday, November 25th. He will talk about VoIP security threats, focusing on the attack surface analysis and threat assessment of IMS Release 8 architecture. The presentation is based on research and numerous audits on live IMS deployments across Europe performed during early 2010. Presentation also reviews results of selected security tests using test automation technique called fuzzing, which finds and identifies both known and unknown vulnerabilities in communication technologies. The presentation is partially based on the Securing VoIP Networks book Ari Takanen wrote together with Peter Thermos.

Involvement with standards bodies is critical for Codenomicon, who works with 200+ communication technologies across industries. The ETSI 3GPP conference gathers leading software testing experts, and both vendor and operator executives together to share experiences and best practices around IMS implementations and deployments. The purpose for Codenomicon is to share its insight for future needs for IMS network testing, and to demonstrate the strength of its model-based testing and security validation techniques.

For more information on Codenomicon Defensics, please visit: http://www.codenomicon.com/defensics/

For more information on Codenomicon products for 3G/4G/LTE testing, please visit: http://www.codenomicon.com/defensics/3g-4g-lte/

For more information on the 3GPP ETSI Release 8 IMS Implementation Workshop, please visit: http://www.etsi.org/WebSite/NewsandEvents/3GPP_IMSWorkshop.aspx

ElastixWorld Will Gather IP Telephony Manufacturers Along with Developers and Users of Elastix in Ecuador this November

VoIP Monitor — Mon, 25 Oct 2010 17:17:08 GMT

PaloSanto Solutions announces the final list of conferences to be presented at ElastixWorld 2010. ElastixWorld 2010 is the first event to gather the Elastix community from around the world in a two-day conference and a further day of integration activities.

"We were surprised by the interest generated by the event from the time we shared the idea with the community. I am very pleased about the reception and support we have received from the hardware manufacturers worldwide" said Edgar Landivar, Elastix Project Leader.

This event has already achieved major interest from software developers and hardware manufacturers of open VoIP products, including Digium, the company behind Asterisk, the most popular open PBX software on today's market.

According to Landivar, this is the first event of its kind in the region and they expect a significant Latin American audience. He also said "During the first week of registrations we had over 200 registered attendants , this number certainly exceeded our initial expectations,"

The first ElastixWorld event will be held in Quito, Ecuador from November 17 to 19. However, due to the worldwide use of the product, the company has announced that the event will be held in a different location each year.

Voiceserve Scheduled to Present at GITEX Technology Week in Dubai and Futurecom in Sao Paulo This Month

VoIP Monitor — Tue, 12 Oct 2010 16:27:10 GMT

Voiceserve will be presenting its VoIP software and solutions this month at the GITEX Technology Week from October 17-21 in Dubai and the FutureCOM in Sao Paulo from October 25-28.

GITEX Technology is one of the most influential and high-profile events in the global information and communication technology sector. Now in its 30th year, GITEX connects over 133,000 professionals with more than 3,500 suppliers. The conference is made up of industry professionals primarily from the Middle East who have purchasing authority of generally up to $1 million, most of whom are seeking new products, services or suppliers.

FutureCOM is Latin America’s largest and most qualified event of the telecommunications and information technology sectors, taking place in the TransAmerica Expo Center in Sao Paulo, Brazil. Participants in the industry come from 40 countries with executive, senior and middle management representing over 70 percent of attendees. The 2009 exhibition was home to more than 240 exhibitors. This will be Voiceserve’s inaugural appearance at the FutureCOM conference and its first exhibition in Latin America.

Voiceserve also noted that it has released a new version of its VoIP dialer for Blackberry cell phones that supports Blackberry’s latest OS 6.0 operating system. The new VoIP dialer provides a more fluid VoIP user experience and improved voice quality. Voiceserve’s upgraded dialer currently works on Global System for Mobile Communications phones, a standard for mobile telephony used in over 200 countries, worldwide. The company is also working on a Code Division Multiple Access version, the dominant network standard for North America and parts of Asia, which it expects to release shortly. Voiceserve’s upgraded dialer is available through Blackberry’s app store and Voiceserve’s website

911 Enable Announces Comprehensive E911 Webinar Series

VoIP Monitor — Tue, 05 Oct 2010 18:22:51 GMT

911 Enable will present a webinar series entitled "E911 Solutions for VoIP and UC Deployments," during the months of October and November. This webinar series will provide a complete view of E911 for IP telephony, including how to meet state and local E911 legislation, manage IP phone locations, and ensure proper routing of emergency calls to Public Safety Answering Points.

The first webinar in the series, "E911 for IP Telephony: What You Should Know," will cover the fundamentals of E911 and IP telephony. This introductory-level presentation is designed for IP telephony administrators, resellers, integrators, and consultants beginning to investigate E911 solutions. Attendees will learn:

What E911 is and how it works
The impacts of VoIP and UC on E911
How to comply with E911 regulations
The different E911 solutions available today

Subsequent webinars will explain in detail 911 Enable's solutions for today's leading VoIP and UC vendors, including Avaya, Aastra, Cisco, ShoreTel, and Microsoft. Designed for those who already have an understanding of the fundamentals of E911, attendees will learn:

How IP telephony impacts E911
An overview of 911 Enable's products and services
How 911 Enable integrates with each vendor's specific platform

The webinar series will be presented by Lev Deich, Technical Director of 911 Enable. During the vendor-specific webinars, Lev will be joined by industry experts with E911 knowledge specific to each platform. These experts will share their E911 knowledge and experiences, and will also be available to answer attendee questions.

The webinars will run from October 19, to November 23, 2010.

Digium to Announce Future of Open Source Communications During AstriCon 2010

VoIP Monitor — Wed, 29 Sep 2010 15:13:31 GMT

Digium announces that Kevin Fleming, director of software technologies, will deliver a keynote presentation regarding the future of open source communications software at the seventh annual AstriCon Open Source Telephony Conference and Exhibition. Each year hundreds of software developers, telephony and unified communications experts, resellers and VARs, and Asterisk enthusiasts attend AstriCon, the longest running conference devoted to the Asterisk telephony platform. Fleming’s keynote takes place on Wednesday, October 27 at 9:00 a.m. ET.

AstriCon 2010 will take place from October 26-28, 2010, at the Gaylord National Resort and Convention Center in National Harbor, Maryland.

Illinois Institute of Technology Conference to Explore the World of VoIP

VoIP Monitor — Thu, 23 Sep 2010 17:56:11 GMT

The Illinois Institute of Technology announces that its Sixth annual VoIP Conference and Expo will take place October 12-14, 2010 in Chicago. This conference, where industry and academia meet, brings together technical professionals and executives from the data and telecommunications industry, standards bodies, government agencies, as well as the business community. The presentations, exhibits, social events, and session breaks provide opportunities to discover different aspects of this rapidly evolving field.

"The overall theme of the conference is 'Where Industry Meets Academia,'" said Carol Davids, Alva C. Todd Professor of Information Technology and Management, Director, IIT SAT Voice over IP Laboratory. "Many conferences strive to bring senior level technology and business leaders together in one place but fall short of that goal. The IIT VoIP conference does just that. A full 80% of the attendees are typically senior business and technical executives from industry, representing both large multi-national companies as well as small start-ups in the space."

Conference Highlights include:

Jeffrey Scott Cohen, Sr. Legal Counsel with the FCC's Public Safety and Homeland Security Bureau will present on "Policy and Technology: The FCC's goals for Broadband Technology, 911 Alerting and Cyber Security"
Christopher Mayer, Vice-President, Systems Integration and Testing, Verizon Communications will chair a panel session on "Carriers View of VoIP and the Future"
Eric Burger, Georgetown University will address "Interoperability - We Can Achieve It - IETF Can Help" in a Keynote discussion.
Henning Schulzrinne, Professor in the Dept. of Computer Science and the Dept. of Electrical Engineering at Columbia University and co-author of numerous IETF RFCs, including SIP will speak on the smart-grid theme, "Connecting VoIP and Location with the Physical World."
Richard Shockey, Chairman of the Board of the SIPForum, will chair a panel entitled, "The V in VoIP is Video," examining the implications of recent explosion of products and services that allow us to stream video as part of our daily lives. Visit the SIP Tutorial tab on the conference website.
Henry Sinnreich and Alan Johnston, Industry consultants and authors will lead a full-day optional SIP Tutorial on October 12, and will also discuss in the main Conference, "How will VoIP reincarnate in the Web and Cloud worlds?"

Planned topics for the sessions include:

Voice over IP over Anything
Unified Communications
Voice over HTTP over IP - Voice over Web
Voices from Beyond the Internet
Mobility and VoIP
Protocols, Architectures & Infrastructure
Operations Test & Measurement
Emergency Services
Cloud Telephony
Smart Grid
Voice on the Web
Security And VoIP Networks, and
Others

Phone.com to Participate at COMPTEL PLUS Fall 2010 Convention & EXPO

VoIP Monitor — Wed, 08 Sep 2010 17:01:32 GMT

Phone.com announces its participation at the COMPTEL PLUS Fall 2010 Convention & EXPO taking place September 12-15 at the Gaylord Texan Resort and Convention Center in Dallas.

COMPTEL is the leading trade association for the competitive telecommunications industry.

Phone.com’s CEO Ari Rabban will be speaking at the conference session titled “The Explosive and Disruptive Growth of SMS”. This session will be taking place on Tuesday, September 14 10:00-10:45AM and will be discussing the following:

Today SMS messaging is exploding in the United States. The number of SMS users has grown 300 percent – from 30 million to 100 million – since 2001, and the number of messages has grown from 33 million to 48 billion in the same period. Moreover, SMS is increasingly built into applications, from social networking to information services. This explosion is having a disruptive effect across industry segments. In this session, speakers will address how carriers frequently block “short codes” as a means of discriminating against specific providers or specific content/applications, the impact of unsolicited SMS messages on consumers, and how SMS is creating huge additional demand for backhaul for cellular carriers and resellers.

“COMPTEL has always been the leading event for the competitive telecom industry, and we look forward to taking part in this great conference and sharing our thoughts about the SMS market. Indeed this is an exploding market, and although some might feel it reached its peak, the truth is that it’s far from it,” stated Ari Rabban, CEO at Phone.com.

TM Forum’s Management World: Destination Dublin

VoIP Monitor — Fri, 03 Sep 2010 16:25:17 GMT

TM Forum announces that after eleven years in Nice, France, its flagship Management World conference will move to Dublin’s new, state-of-the-art Convention Center - the CCD - May 23-26, 2011. Dublin and the CCD will provide the perfect business environment for Management World attendees—a leading capital city that is easy to get to, easy to do business in and with a reputation for hospitality that is second to none.

As a vibrant European hub for global technology companies, including major communications service providers, IT organizations and large online businesses such as Google, Facebook and eBay, Dublin is a natural fit for the increasing scope of the industry’s premier conference.

The Convention Center Dublin is the world’s first carbon-neutral conference facility. This purpose-built venue allows for an exciting range of new sponsorship opportunities available to TM Forum members, with significant support already shown by long-standing sponsors of the event.

In addition to a new location, a number of innovative features are under development for the 2011 conference. “Our highly successful Executive Program will expand in 2011, providing a unique forum for senior executives from across the industry to network with their peers,” added Willetts. “We’re also creating new set interactive presentation formats and exclusive content. In short, Management World 2011 will be our best conference yet!”

Stay tuned to www.tmforum.org for more information on the 2011 conference.

Key Facts:

Now In its 12th year, TM Forum’s Management World 2011 will offer a unique blend of thought-leadership, real-world case studies and interactive debate to help companies focus on growing revenue through new business models, increasing operational efficiency and cost reduction, whilst addressing the challenges of revenue assurance, customer experience and retention
Management World is widely respected as the annual meeting place of the industry and provides a unique mix of networking, business and exclusive content to more than 3,000 senior executives from over 80 countries worldwide each year
Complementing the conference will be brand new TM Forum Catalyst innovations, more than 10 TM Forum training courses to help companies adopt TM Forum’s highly successful Frameworx standard
In 2010, Management World attracted:
- More than 3,000 attendees from the Technology, Media and Communications markets
- More than 50% of attendees held executive and senior management positions
- 668 companies attended from 81 countries worldwide
- The event featured more than 200 expert speakers.

AstriCon Speakers Line Up to Discuss Future of Asterisk, IPv6, SIP Security and More

VoIP Monitor — Thu, 29 Jul 2010 20:18:10 GMT

AstriCon is quickly taking form, with speakers on a range of technical and business topics confirmed. Organizations of all sizes use Asterisk to create customized, feature-rich and scalable phone unified communications systems, VoIP gateways, conference servers and more. Digium the owner and corporate sponsor of Asterisk, hosts AstriCon, which is now in its seventh year. A few of this year’s speakers will include:

“Just How Vulnerable is Your Phone System?”

Sandro Gauci—“Just How Vulnerable is Your Phone System?” Sandro is the author of “SIPVicious,” one of the most popular brute-force attack test kit systems for SIP platforms. He’ll talk about attack methods, vulnerabilities administrators should understand and how to mitigate these issues on Asterisk.
Jim Kerr—“Orbitz and Asterisk: How We Did It.” Orbitz has implemented Asterisk to handle customer calls. Learn how this huge travel management agency integrated the platform with its legacy system, deployed it across call centers and achieved PCI security compliance.
Simon Perreault—“IPv6 in Asterisk 1.8.” Asterisk will support IPv6 in 1.8, and Simon will talk about the changes, configuration requirements and fundamentals of IPv6 required to get Asterisk running. Then, as part of a double-long session, he will walk through configuration of an entire platform on a purely IPv6 network.
Russell Bryant—“Asterisk Development Update.” Russell is the lead for open source Asterisk development at Digium. Hear about what’s happened in the last year with Asterisk development and what features and functions are coming soon.
Jim Van Meggelen—“Connecting Systems Together.” Jim is one of the authors of the book Asterisk: The Future of Telephony. His latest area of focus involves connecting disparate Asterisk systems. How to manage endpoints? What routing method should be used to connect offices? How about connecting Asterisk to global routing networks of other organizations? Get your island of phones connected after listening to Jim’s talk.
“Ask the Experts.” A panel of Digium sales engineers, support staff and engineers will give some short examples of the toughest problems they’ve encountered and solved, and then the audience will be invited to throw out their most difficult problems.

AstriCon will be held just outside Washington, D.C., from October 26-28, 2010, at the Gaylord National Resort and Convention Center. Additional information and online registration are available at http://www.astricon.net.

Ingate Systems Presents the New SIP Trunk-Unified Communications Summit at ITEXPO West 2010

VoIP Monitor — Wed, 07 Jul 2010 16:00:14 GMT

Unified Communications is coming of age and it's being driven by the urgent demand for SIP trunking, which lowers operating costs and delivers rapid ROI. To address the need for information on the what, why and how of Unified Communications and SIP trunking, Ingate Systems is partnering with TMC, thought-leaders and vendors in the space to present the new SIP Trunk-Unified Communications Summit at ITEXPO West 2010.

Free to all ITEXPO attendees, the Summit is a three-day seminar series providing in-depth educational information on SIP trunking and Unified Communications. The Summit will include general information panels and technical insight sessions from both the service provider and enterprise perspectives, and will feature visionaries driving the industry.

To date, confirmed speakers include Dialogic, Iwatsu, Mitel NetSolutions, ShoreTel, The SIP School and VOIPSA. David Yedwab, a Founding Partner in Market Strategy and Analytics Partners LLC., contributor to Unified Communications Strategies and TMCnet columnist, will discuss the future for service providers in a UC-enabled world. Additionally, service provider Telia will present their SIP trunk implementation with Intertex Data AB as a case study. Joel Maloff of Maloff NetResults will moderate the panel discussions.

The SIP Trunk-UC Summit will be held October 4-6, 2010 at the Los Angeles Convention Center in Los Angeles, California.

"This season will feature a strong focus on ROI. The economy has forced decision-makers to take a hard look at new technologies, to invest in solutions that lower costs, improve productivity and are easy to implement. SIP trunking fits the bill," said Steven Johnson, President, Ingate Systems. "SIP trunking is a secure, cost-effective way for enterprises to adopt SIP, and it's the first step toward UC adoption."

Sessions this season will include:

SIP Trunking Professional Development Program

Building for ROI
Case Studies
Service Provider Perspective - US and European Views
How to Sell SIP
Town Hall Meeting: SIP, UC and Security
Unified Communications Day

Fax-over-IP

Legacy PBX/PSTN and SIP Trunking
The Future for Service Providers in a UC-Enabled World
The ROI of SIP Trunking and UC
SIP Trunk Boot Camp

Deploying SIP Trunks - Getting it Right the First Time

SIP Trunk "Basic Training" with Ingate
Live demonstrations, during which participants will set up a SIP trunk in 20 minutes on-site, will also be part of the program.

Attendees can earn a SIP Trunking Professional Certificate by participating in the Professional Development Program on the first day of the Show.

"Ingate's SIP Trunk-UC Summit series is a unique opportunity for ITEXPO attendees to get a crash course on SIP trunking from the experts," said Rich Tehrani, CEO and group editor-in-chief for TMC, hosts of ITEXPO. "We are proud to once again work with Ingate Systems in providing these educational seminars at ITEXPO."

D2 Technologies Showcases Broad Range of VoIP Platforms and Applications

VoIP Monitor — Tue, 01 Jun 2010 17:01:23 GMT

D2 Technologies is showcasing the industry’s broadest range of VoIP platform offerings for OEMs/ODMs through on- and off-floor demonstrations with the world’s leading semiconductor vendors during the COMPUTEX 30 Taipei tradeshow. D2’s vPort Gateway embedded VoIP software allows manufacturers to use their IC platforms of choice to quickly deliver advanced Integrated Access Devices, residential gateways and other CPE with optimized, carrier-grade VoIP functionality over wireline, WiFi and WiMAX connections.

During COMPUTEX 30 Taipei, being held June 1-5, 2010 at the Taipei International Convention Center, interested parties can contact D2 at the show to schedule application and platform demonstrations with the following vendors:

Cavium Networks – vPort-GW live VoIP triple-play demo on the Cavium ECONA CN3420 platform. D2 Taiwan will have technical experts available by request or appointment.

Fujitsu – vPort-GW live VoIP over WiMAX demo on the Ralink RT3052 + Fujitsu WiMAX dongle platform. Personnel from D2 Taiwan will be presenting vPort solutions in the Fujitsu booth (TWTC Exhibition Hall 1 A0430a).

Marvell – vPort-GW live VoIP demo on the Marvell 88F6281 platform. D2 Taiwan will have technical expert available at the Marvell booth (TICC T0101C).

Ralink Technology – vPort-GW live VoIP demo on Ralink RT3052 platform at the Ralink demo suite (Grand Hyatt Taipei Room #1005/1006).

Sequans Communications – vPort-GW VoIP over WiMAX reference design, showing vPort running on the Ralink RT3052 SoC with a Sequans SQN1210 WiMAX modem (TWTC Exhibition Hall Booth #1 A0418a).

Ubicom – vPort-GW live VoIP demo on the Ubicom IP7K platform. D2 Taiwan will have technical experts available at the Ubicom booth (TICC Room #103).

XO, TMC Net and Broadsoft to Host Webinar: ''Beyond SIP Trunking - Unify the Enterprise''

VoIP Monitor — Wed, 05 May 2010 17:12:35 GMT

XO Communications in collaboration with TMCNet and BroadSoft, will host an interactive webinar on May 12th entitled, "Beyond SIP Trunking - Unify the Enterprise." The event will bring together several SIP trunking experts to discuss the latest industry trends and insights on how distributed enterprises can streamline their managed voice services.

Specifically, the hour-long session will cover topics such as the benefits of SIP trunking services and future trends, how to evaluate the return-on-investment of SIP, as well as things to consider when evaluating SIP trunk service providers. It will also offer an inside glimpse of various enterprise networks before and after SIP trunking was implemented.

The webinar will feature the insights and perspective of leading SIP trunking experts, including:

Lisa Pierce, founder and president of Strategic Networks Group, a consultancy dedicated to improving the quality of emerging telecommunications and IT services, and the service experience that business customers receive from key suppliers. She brings a unique expertise in a wide range of network technology including SIP trunking, unified communications, broadband access, managed network services, among other things.
Steve Carter, senior product manager at XO Communications, led the launch of the company's SIP trunking services, including XO Enterprise SIP, which won the INTERNET TELEPHONY Product of the Year Award in 2009.
Alex Doyle, senior director of global solutions marketing at BroadSoft, has been intricately involved in the development of the BroadWorks application platform and has been responsible for product solution management of BroadSoft's business and consumer applications.
Erin Harrison, senior editor of TMCnet, is a seasoned reporter and editor and covers IP communications, information technology and other related topics for TMCnet.

Webinar Details: Beyond SIP Trunking - Unify the Enterprise When: May 12th at 2:00 p.m. EDT Who Should Attend: Both technical and non-technical audiences from U.S.-based businesses Registrations: Complimentary Webinar - Click here to register for the webinar.

XO Enterprise SIP XO Communications is an industry leader in SIP trunking solutions for distributed enterprises. Its newest solution, XO Enterprise SIP, enables customers to utilize a centralized IP-PBX architecture in key locations to deliver VoIP services to branch locations across an existing wide area network or using the XO MPLS IP-VPN solution. Moreover, utilizing XO Enterprise SIP customers can achieve a number of benefits including:

Lower Total Cost of Ownership by using a single or fewer IP-PBXs to support all locations;
Reduced Operating Costs by not having to maintain costly PRI facilities or local voice trunks at each location, and eliminating the operating expense of managing separate voice and data networks;
Greater Flexibility by allowing locations to burst above their normal call capacity and sharing idle voice trunk capacity from other locations across the enterprise;
Increased Efficiency in network management through simplified and converged network operations, significantly less effort to connect new locations to the public switched telephone network;
Business Continuity with redundant Enterprise SIP connections and the ability to automatically re-route calls to alternate locations;
Extensive Nationwide Availability of XO VoIP services in all 50 states and more than 2,700 cities.

Acme Packet Schedules Enterprise Seminar Series: Deploying SIP Services Successfully

VoIP Monitor — Wed, 28 Apr 2010 15:51:13 GMT

Acme Packet announces the launch of the Acme Packet enterprise seminar series, Deploying SIP Services Successfully, kicking off in San Francisco on May 20, continuing on to New York, May 25, Chicago, May 26 and ending in Washington, D.C., on June 2. The goal of this series is to educate enterprises, contact centers and government agencies about the significant challenges of integrating SIP trunking, video conferencing, unified communications, contact centers and hosted cloud communication services, and the role of session border control in ensuring interoperability, security, quality, availability, and regulatory compliance.

The half-day seminar will feature presentations on solutions for SIP trunking, contact centers, video conferencing, and unified communications, as well as SIP-based service provider offerings. The target audience is the broad range of IT professionals that need to understand the issues and pitfalls involved in implementing these solutions, including CTOs, IT directors, network engineers and telecom managers.

Registration for the enterprise seminar series is now open on a first-come, first-served basis.

SIP Forum’s SIPit 26 to Take Place in Kista, Sweden, May 17-21, 2010

VoIP Monitor — Tue, 13 Apr 2010 18:41:04 GMT

The SIP Forum has announced that its next SIP Interoperability Testing event, SIPit 26, will be held May 17-21, 2010 in Kista, Sweden. The event will be hosted by Edvina and TANDBERG, and sponsored by Intertex, Ingate, .se, and the IPv6 Forum as an association sponsor.

Conducted by the SIP Forum twice a year, SIPits are the world’s premier interoperability testing events for the SIP, bringing together the leading SIP application developers, service providers and IP communications equipment manufacturers to ensure their SIP implementations work seamlessly together in an IP network testing environment. An important goal of the SIPit events is to help refine both the SIP protocol and its implementations in order to further establish SIP as a global interoperable standard for real-time Internet communication services.

"SIPit 26 brings together equipment vendors and service providers across the global IP communications industry to test and validate their SIP implementations in a live, real-world IP network setting in Kista, Sweden, a center of telecom technology in the suburbs of Stockholm,” said Marc Robins, SIP Forum President and Managing Director. "In addition, this spring event will also give companies the chance to test specifications from standards bodies such as the IETF, as well as recommendations formulated by SIP Forum task groups. A goal is to reinforce the progress that these groups are making to further establish the SIP protocol within the service provider, consumer and enterprise network environments.”

SIPit is organized by the SIP Forum’s Test Event Working Group and serves as a “plugfest” for participating companies to perform SIP interoperability testing with other participants in a live network environment. Conducted twice a year, with events rotating in the United States, Europe, and Asia, the SIP Forum has hosted 25 plugfest events around the globe. The previous event, SIPit 25, was hosted by the University of New Hampshire Interoperability Lab in Durham, New Hampshire in September 2009.

“The SIPit events are extremely effective testbeds, both for implementations and for specifications,” said Robert Sparks, chair of the SIP Forum’s Test Event Working Group. “We frequently have participants indicate that a week spent at SIPit provides results that would have taken months to achieve with individual pair-wise testing. Information about the state of implementation and interoperability of the specifications has been very useful in informing ongoing standards work.”

"Interoperability is the foundation for the TCP/IP protocol suite. As we move to realtime communications, it's important to test interoperability in all the platforms that are developed. For us in the Asterisk.org Open Source development group, it's been really important to participate in SIPit, where we can test, learn and improve our solution in a very open and friendly setting. SIPit is an important part of the success of the IETF Session Initiation Protocol," says Olle E. Johansson, founder of Edvina and developer of Asterisk. "We're proud to host this event in partnership with TANDBERG and thus become part of the success story of SIPit."

ABP's IP Sizzles 2010 Roadshow Coming Near You?

VoIP Monitor — Tue, 06 Apr 2010 18:15:42 GMT

ABP announces the seventh annual IP Sizzles Conference for technology resellers and integrators in telephony, data networking and IP video surveillance. The show will highlight innovative and profitable solutions resellers are selling in the IP technology and IP communications space. Leading themes are Unified Communications, IP Surveillance and IP Video applications for SMB. IP Sizzles will focus on key business aspects and sales process improvements for resellers to be more successful in today's challenging economic and competitive environment.

The fastest growing killer application is Mega-pixel Video Surveillance providing HDTV image quality and a wide variety of options for monitoring, recording and video analytics as well as IP Digital Signage. Industry consensus is that the PBX is the Unified Communications Platform and will incorporate Voice, Video, Fax, Presence, and IM messaging. Small and medium businesses expect to be using IP Video calls and Video Conferencing to improve their processes. IP Sizzles will provide the best in both areas with real world applications plus a bonus session on IP Digital Signage offering resellers all of the leading, market-driven solutions.

IP Sizzles provides commercial highlights, an expo to mingle with vendors and peers, and offers training sessions including hands-on workshop for the newest solutions. IP Sizzles attendees will also be among the first to see new products from Aastra, Broadvox, Digium, Epygi, Grandstream, ISS, Mobotix, Nuuo, Patton, snom technology, Veracity and others. IP Sizzles features speakers from different areas of the IP industry who will describe how ABP Resellers can reach a higher level in their core competency and ideas for expanding their sales and marketing efforts.

For the first time ABP will host a combined US tour and an International IP Sizzles. The ABP team and sponsors will cover eight major US cities and also travel to Central and South America. The tour will kick off in Santo Domingo, Dominican Republic on May 5th-6th for the International leg of tour and in Los Angeles on May 18th-19th for the USA start.

IP Sizzles 2010 is the "must attend" event of the year for Data & Networking, Telecom and Security resellers and integrators.

Alteva To Participate In VoIP Related Webinar Series

VoIP Monitor — Wed, 31 Mar 2010 19:49:14 GMT

Alteva will participate in Chorus Communications' 2010 webinar series, to run every Tuesday, starting March 23rd. The webinar series is designed to provide sub-agents with the knowledge base needed to sell the hottest products in the marketplace today, including hosted VoIP and unified communications solutions.

As a highly regarded thought leader in this industry, Alteva has been asked to participate in the following webinars:

Tuesday, April 6th and Tuesday, April 13th at 10:00 AM eastern time
Topic: International VoIP Connectivity, SIP Trunking & Session Border Controls
Presenters: InPhonex with Alteva

International porting capabilities offer the ability for phone service providers to seamlessly connect their customers with other locations abroad without the common constraints of cost normally associated with this type of connection. International companies will be able to have an appearance in the US and the ability to retain a local presence using a local number. Alteva and InPhonex will discuss how these capabilities create a standardization and consolidation of its billing across all of the user's offices. Specifically, Alteva will talk about the enterprise VoIP component, while InPhonex will concentrate on the residential and ultra small business capabilities. InPhonex is one of the many carriers that Alteva utilizes, as well companies such as Level 3 Communications and Global Crossing.

Seminar 4: Tuesday, May 4th and Tuesday, May 11th 2010 at 10:00 AM eastern time
Topic: Hosted vs. Premise based VoIP Solutions
Presenters: Alteva and Quick Connect

It is becoming quite apparent that in the near future, everyone will transition their phone services over to VoIP. Depending on the circumstance, hosted VoIP might be better suited for some companies more than on-premise solutions and vice versa. Alteva will collaborate with Quick Connect to provide an educational discussion on the benefits of each service, and a real perspective on when to deploy each of these products.

How to Sign Up

Agents looking to get educated on how to sell your services are invited to participate in the webinar series. A limited number of seats are available. Additionally, Alteva is offering its own webinar series that will cover overarching topics, as well as more detailed topics such as unified communications and cloud services.

To learn more or sign up for this program, please contact Louis Hayner, Chief Sales Officer at Alteva, at 1-215-789-4028 or lhayner@altevatel.com.

VoiceCon - Live From Orlando, FL

VoIP Monitor — Fri, 19 Mar 2010 17:06:23 GMT

You can watch videos live from VoiceCon Orlando, the only event that covers the most important issues in enterprise communications. We're sorry you couldn't join us; however you can still be a part of the excitement surrounding the event right from your desktop with VoiceCon TV.

You can watch live keynotes and summits from the conference.

During live broadcasts, you'll see keynotes from VoiceCon Orlando as they happen. When no live sessions are taking place, a loop of previously recorded sessions will be shown. You can also access a menu of previously recorded content by clicking on the video player's “Menu” button.

Below is the VoiceCon TV live broadcast schedule. We hope you can tune in!

Digium Announces Seventh Annual AstriCon to be held October 26-28, 2010

VoIP Monitor — Tue, 23 Feb 2010 17:52:01 GMT

VoiceCon Orlando 2010 Announces Best of VoiceCon Finalists

VoIP Monitor — Mon, 15 Feb 2010 18:46:18 GMT

VoiceCon announces the seven finalists of the Best of VoiceCon program, which recognizes exhibitors who have made significant technological advancements within the enterprise communications industry. A panel of industry leading judges will select the winner, which will be announced at VoiceCon Orlando 2010, happening March 22-25 at the Gaylord Palms Convention Center.

The seven finalists selected from over 40 submissions include:

Acme Packet-- Acme Packet Net-Net Session Border Controller in Skype for SIP: Skype has deployed Acme Packet Net-Net session border controllers to simplify the interoperability and feature compatibility of the Skype for SIP beta offering with enterprise IP-PBX equipment.
Avaya-- Avaya Aura Solution for Midsize Enterprises: A single-server, unified communications solution for businesses as small as 100 employees and scalable up to 2400 users and 250 locations.
Cisco-- Cisco Intercompany Media Engine: A new product that establishes a foundation for cross-organization rich media communications.
NET-- VX1200 with Extend features: Designed to support the adoption of enterprise voice for Microsoft Office Communication Server 2007 R2 customers, providing branch offices with several key features.
Psytechnics-- Experience Manager 5 with Service Desk Manager: Delivers per session QoE visibility to enable user- experience based IT operations and service management.
Siemens Enterprise Communications-- OpenScape UC Server 2010: The new iteration of Siemens’ next-generation, software-based enterprise communications platform for Voice, UC, Contact Center, Video and Mobility.
Sipera-- Sipera Secure Live Communications: A security solution enabling employees to use VoIP, UC, cloud telephony, and other low-cost and feature-rich communications applications on mobile devices.

Evaluating each company’s technological innovation and value to market, judges selected seven finalists to compete for the Best of VoiceCon award; the field of finalists was expanded from the initially-announced six to seven when the judges’ voting produced a tie. The winner will be announced on Wednesday, March 24 at VoiceCon Orlando. The Best of VoiceCon’s judging panel of industry experts includes:

Eric Krapf, VoiceCon Program Co-chair
Fred Knight, VoiceCon GM and Co-chair
Michael Finneran, President, dBrn Associates
Marty Parker, Principal, UniComm Consulting
Sorell Slaymaker, Communications Architect, Unified IT Systems
David Stein, Principal, PlanNet Consulting
Allan Sulkin, President, TEQConsult Group