snom released a new version of its award-winning snom ONE IP PBX, adding advanced mobility capabilities that extend standard IP PBX calling features to employee mobile devices. The new release, introduced at ITEXPO East (Booth #512) in Miami, January 31 - February 3, also features new management and security features to more easily provision and integrate a businesses cell phone cell fleet with the snom ONE IP PBX.

snom ONE IP PBX release highlights include:

• "Cell phone as an extension": The snom ONE allows cell phones to act as truly integrated extensions, incorporating call transfers, conferencing, internal extension dialling and other features.
• Broad SIP client support: The snom ONE now supports mobile SIP clients, running on popular platforms such as Android and the iPhone.
• Enhanced management features: The web-based interface has been enhanced to make administering the system even easier and more productive.
• Enhanced remote phone security: snom ONE provides WAN-based authentication for plug and play with snom 7xx and snom 8xx series phones, alleviating the need for users to enter password information and increasing simplicity and security for remote workers.
• Plug and play deployment: The snom ONE is optimized for all snom phones, enabling plug-and-play deployment and provisioning for the snom 3xx, snom 8xx and newly released snom 7xx series desktop phones, as well as other endpoints, including the snom M9 DECT phone, the snom PA1 public address system and the snom MeetingPoint conference phone.
• Virtual appliance: snom ONE is available as virtual appliances for VMware and Microsoft Hyper-V via a .vhd file. This addresses the demand for hardware failover without dropping calls.
• Enhanced Software Update Mechanism: The system can be updated easily via the web interface, alleviating the need to connect to the operating system to perform upgrades.

The snom ONE IP PBX is a SIP-based, full featured IP PBX optimized to work with snom phones, offering advanced features including agent groups, conference rooms, call recording, and many other features typical in a mature SIP based communications system.

The snom ONE is compatible with Windows, Linux and Mac environments and is equipped with robust web security through HTTPS and call security through TLS and SRTP. snom ONE supports mixed IPv4/IPv6 LAN and WAN environments and comes with an automatic blacklisting feature that makes it possible to expose public IP addresses. The snom ONE is also available as a turnkey system in snom ONE Plus appliances.

The snom ONE IP PBX is available in three versions: snom ONE free (downloadable), for up to 10 extensions, snom ONE yellow (for up to 20 extensions) and snom ONE blue (unlimited number of extensions and multi-tenant capabilities up to five companies). All versions offer the full feature set, including hunt and ACD groups, mailbox, auto attendant, conference rooms and paging, and are designed to take full advantage of the hardware features of snom's suite of desktop phones and endpoints. snom ONE blue also allows up to five separate corporate tenants, supporting multiple organizations to operate using a single IP PBX.

snom introduced snom Active, a new secure and hosted provisioning solution for snom IP phones being deployed as part of an integrated business IP telephony or unified communication system. snom Active provides snom distributors and certified value-added resellers with a free advanced provisioning and management tool for snom phones and endpoints that boosts the speed and efficiency of the deployment process for VARs’ implementing VoIP or UC in the enterprise.

snom Active provides snom partners with the remote capability for bulk configuration, provisioning and ongoing administration of snom’s SIP endpoints, including live updates of snom software and firmware. As a hosted, web-based service, snom Active offers VARs a highly reliable platform to manage and provision hundreds to thousands of snom phones across multiple distributed sites.

snom Active features include:

• Auto provisioning
• Managed snom subdomain
• Guaranteed 99.999% system uptime
• Bulk phone configuration
• Fleet and system-wide management and administration
• “White label” hosting opportunities
• Custom setup (office and end user level profiles)
• Full remote deployability
• Unlimited re-registerability on handsets
• Full HTTPS config traffic
• Managed password protection
• Military-grade security

Seamless UC support

The snom Active hosted provisioning service provides full support of the entire portfolio of snom phones in unified communications environments, including the snom 3xx and snom 8xx series’ of desktop phones, the snom m9 cordless DECT phone, and additional snom endpoints, including the snom Meetingpoint conference phone and the snom PA1 public address system.

Fast and Efficient snom provisioning with Microsoft Lync 2010

snom Active also supports the snom UC line of Microsoft Lync 2010-qualifed snom phones, including the snom 300 UC edition and the snom 821 UC edition, bringing the speed and simplicity of configuration and provisioning to large enterprises deploying snom UC phones with Microsoft Lync. snom Active with Lync also allows for managed password protection, meaning system and IT management do not need to know end-user passwords, which complies with access security protocols in many organizations.

Broad deployment with IP PBX, hosted and managed VoIP business services

For VARs offering hosted, managed or IP PBX communications services with snom endpoints, snom Active offers an easy-to-use management platform with guaranteed uptime and a new level of scaled network visibility and manageability.

By streamlining most provisioning and management functions of a snom IP telephony system, snom Active helps maximize cost efficiencies by eliminating costs associated with support, administration and maintenance. Official snom distributors and certified snom VARs can also qualify to offer hosted “white label” snom Active provisioning service, a customized and branded solution that snom channel partners can offer to their end customers.

The snom Active service is now available free of charge to certified snom resellers and official distributors worldwide.

Digium releases Asterisk 10. Asterisk is a communications platform that allows developers to create powerful business phone systems and unified communications solutions. Since its introduction 12 years ago, Asterisk has been used, free of charge, in nearly every country of the world to power telephone and other communications systems. It has been downloaded millions of times, including two million last year alone, establishing Asterisk as the most popular open source telephony engine.

The most important new feature in Asterisk 10 is its wide-band media engine. Digium has replaced Asterisk’s telephony-grade media engine with a more advanced one, providing support for studio-quality audio and a nearly unlimited number of codecs. By supporting high and ultra high-definition voice, Asterisk can now be used to power communications applications that would have otherwise required specialized or expensive equipment and service in order to convey nuances in speech or emotion. Digium has also updated Asterisk’s media support for Asterisk 10, with several new codecs, including Skype’s SILK codec, 32kHz Speex support and pass-through support for CELT.

Built with open source community support

Digium is advancing Asterisk with version 10, while simultaneously leading work on the Asterisk Scalable Communications Framework. Asterisk SCF will allow developers to create real-time communications applications that include voice, video and text that meet the demands of a full range of uses, from embedded applications to enterprise and carrier solutions.

Asterisk 10 makes its debut at AstriCon, the Asterisk User Conference & Expo, in Denver. Hundreds of attendees, including software and PBX developers, enterprise IT pros, systems integrators and call center and CRM developers, welcomed the announcement. In its eighth year, AstriCon is offering conference tracks focusing on technical information, carriers and call centers, cloud computing, commerce, government, enterprise and the Asterisk ecosystem. Developer conferences geared toward contributors to the Asterisk and Asterisk SCF projects are also taking place during this year’s AstriCon.

Asterisk 10 is available for free download and is licensed under the GNU General Public License v2.

New features in Asterisk 10

Asterisk 10 offers developers, integrators, resellers and telephony pros a range of new capabilities. A few include:

• New media engine—Asterisk 10 supports more media types and virtually any type of audio. The overhaul to the media engine allows Asterisk to support a nearly unlimited number of codecs.
• More codecs—The platform includes new codecs, including the wideband version of Speex, Skype’s super-wideband SILK and pass-through support for several CELT variants.
• Additional sampling rates—Asterisk previously operated on 8 and 16 kHz sampled audio, but now supports super- and ultra-wideband sampling rates as file format types for file playback or recording. Asterisk now supports 8, 12, 16, 24, 32, 44.1, 48, 96 and 192 kHz rates for superb audio quality.
• New conferencing application—Digium replaced the MeetMe conferencing bridge with an HD-capable intelligent bridge application called ConfBridge. It supports all codecs and conference rates and works on any Asterisk 10 system, regardless of operating system or architecture. Intelligent mixing algorithms provide each participant with the optimal audio quality for their connection. Also, ConfBridge is fully customizable, so systems administrators and integrators can configure call-in menus on a caller-by-caller basis.
• Support for videoconferencing—ConfBridge relays video of a designated speaker or the current speaker to other participants in the conference. Video-capable SIP devices that use the same codec are required.
• Significant new fax capabilities—Asterisk 10 includes T.38 gateway capabilities that allow outgoing fax calls from analog fax machines to be connected to T.38 fax endpoints over SIP and incoming T.38 fax calls to be delivered directly to fax machines. This allows for more straightforward integration of fax capabilities into an Asterisk system and allows users to get delivery confirmation from other fax machines.
• Text message routing—Asterisk has long been able to send and receive text messages, but can now route messages as well. Asterisk 10 supports the SIP MESSAGE and XMPP protocols, allowing it to act as a text messaging server and bridge between different messaging protocols.

VoxOx and VoxOx In Business announce the addition of Data-Max Wireless to its white label VoIP customer roster. Data-Max Wireless is leveraging the company’s managed wholesale VoIP enablement platform and robust back–end infrastructure to provide a wide range of services to customers nationwide. Telcentris offers white labeling of all Internet telephony services that are sold through its VoxOx In Business division, including: SIP Trunking, Hosted PBX, Hosted Contact Center, as well as carrier services, wholesale SMS, data services, and more.

Operated by 50-year Internet access veteran, Wecom Inc., Data-Max Wireless provides residential and business services throughout Mohave County in Arizona. Data-Max Wireless initially shied away from offering voice services; however, over the past few years, the company observed strong demand for VoIP services from their customers.

VoxOx In Business division's VoIP solutions are all powered by the company’s award-winning unified communications service delivery platform, which has been recognized for product excellence by INTERNET TELEPHONY Magazine for the past four years, consecutively.

VoxOx In Business’ white label VoIP portfolio is available both directly and through its acclaimed Channel Partner Program.

IPsmarx has announced a referral arrangement with 360networks, a full service wholesale provider of Private Line Transport, VoIP, IP, and Local Access (T1/DS3) services on its wholly-owned fiber optic backbone. Through this arrangement IPsmarx will recommend 360networks’ IP and Toll Free Origination services to complement IPsmarx Class 5 and IP-PBX Solutions.

8x8 announces the release of "8x8 Virtual Office Pro 2.0," an upgraded edition of its award-winning hosted unified communications offering with improvements designed to further enhance the productivity and mobility advantages of this powerful web-based solution.

Originally introduced in January 2010, 8x8 Virtual Office Pro delivers a complete suite of web-based communications and collaboration services - such as phone, fax, web conferencing, call recording, chat, mobile application and more - via a single online dashboard, enabling access to these tools remotely from any location using just a PC and web browser. In addition, 8x8 Virtual Office Pro reduces a business' communications costs by combining the services (including unlimited local and long distance VoIP calling) in a bundled offering priced at under $50 per user, significantly less than the cost of acquiring the same services separately from individual providers.

Enhancements included in 8x8 Virtual Office Pro 2.0 are as follows:

• Improved softphone functionality -- with advanced and instant call forwarding, call transfer, call waiting, 3-way calling, multiple call appearance lines, blind, warm and voicemail transfers, one click calling from Outlook, Google and social media networks; complete PBX functionality without the need for a desktop IP phone
• Added fax capabilities -- users can now send multiple documents in the same fax and create a customized fax cover sheet; service allows for sending and receiving unlimited faxes
• iPhone iOS4 multitasking support -- Virtual Office Mobile application now supports iPhone iOS4 multitasking and background operation; provides one-button access to voicemail, conference bridge and auto-attendant
• Social Media Integration -- easily post, tweet, chat and send status to your Twitter and Facebook contacts directly from the Virtual Office Online dashboard
• Enhanced Call Queuing -- subscribers with call queuing capability can now have agents log in and out of their call queues from within the online dashboard
• Video Chat -- users can now add live video to chats and calls with co-workers

These feature upgrades add to 8x8 Virtual Office Pro's existing core capabilities which include:

• 8x8 Virtual Office Online - a flexible, online dashboard that lets subscribers manage the key features and functions of their Virtual Office business phone service remotely, through an easy-to-use web portal
• Designate any convenient handset or computer headset to place and receive calls from
• Set call handling rules, listen to voicemail online and view a comprehensive listing of voicemails, phone calls and chats.
• Reduce costly roaming and international calling charges by using an 8x8 VoIP extension instead of a cell phone, hotel phone or landline desk phone, to place outbound calls
• 8x8 Virtual Office hosted PBX phone service - reliable, high quality business VoIP phone service with advanced features and unlimited local and long distance calling
• 8x8 Virtual Meeting -- flash-based web conferencing solution that allows users to create, join and invite participants to web, audio and video meetings
• Virtual Office Mobile -- place and receive (VoIP) calls and access common Virtual Office services and functions from an iPhone, iPod Touch or iPad
• Internet fax - send and receive unlimited online faxes from any computer; includes free local U.S. fax number
• Call recording and archiving - enables any inbound or outbound call to be recorded and later reviewed, downloaded, deleted and archived
• Presence management - tells other co-workers whether you are logged in, logged off, on the phone, off the phone or currently unavailable

8x8 Virtual Office Pro 2.0 is available to existing 8x8 Virtual Office Unlimited, Metered or Global Extension subscribers as an optional service for an additional $20 per extension per month. New 8x8 Virtual Office subscribers can receive the entire bundle (Unlimited Extension plus Virtual Office Pro) for a promotional price of $49.99 per month.

TalkFree has added new features to its popular 1LegCall wireless VoIP application and Zonefone mobile VoIP dialers. Both solutions now support the S60 5th Edition, Symbian OS version 9.4 Nokia mobile phone models and Short Message Service.

1LegCall and Zonefone now can be used with nearly the entire Nokia S60 phone series, including popular models X6, N97 and 5800 XpressMusic. These new edition phones are selling fast in the Middle East and Africa, which are strong markets for TalkFree. Indeed, the carrier uses its mobile VoIP platform to complete more than 500,000 calls originating in these markets daily.

Additionally, TalkFree has integrated SMS functionality into its existing mobile VoIP platform. With 1LegCall, the carrier’s resellers now can give their customers the ability to send international SMS messages at a fraction of the rate charged by traditional wireless carriers. “The same application that empowers our customers to make affordable calls using their mobile phone and VoIP now allows them to send affordable international SMS,” says Jean Templin, director, product marketing for TalkFree.

TalkFree’s 1LegCall application uses existing mobile handsets to deliver an inexpensive call solution using VoIP technology. Customers simply load the dialer application onto their existing phone using their 3G or WIFI connections and they can instantly bypass expensive fixed-line infrastructure and phone rates. The1LegCall application currently runs over 3 million minutes of traffic daily and has a growth trajectory of nearly 300 percent annually, demonstrating the strong demand for mobile VoIP technologies in emerging markets and Third World countries.

TalkFree operates primarily in the Middle East and Africa with targeted growth coming from Southeast Asia and rural areas of South America. The company specializes in assisting local in-country communications providers by delivering a turnkey, plug-and-play VoIP “business in a box” with no contractual obligations and inexpensive start-up costs. TalkFree’s multiple access techniques ensure high-quality voice connections regardless of Internet traffic volumes. All applications are managed in real time, in local language and local currency, and are supported by customer service experts who are available 24/7.

Goober Networks, a Unified Communications provider, announced goober 3.0 as a public beta, a new collaboration solution that includes multi-protocol instant messaging, HD Voice to landlines and cell phones and 6-way video conferencing calls to PCs including Windows and Macintosh.

Global IP Solutions has been selected by LG Electronics Mobile Communications Company to provide HD voice for its innovative Fixed Mobile Convergence solution. FMC allows telecom operators to combine fixed telephony, cellular telephony and Broadband into unified end-user services. LG Electronics added cutting-edge HD Voice capabilities to its FMC application by integrating GIPS VoiceEngine™ Mobile, which includes advanced acoustic echo cancellation technology. Good AEC implementation is critical to improve quality of voice calls on mobile devices.

Implementing GIPS HD voice technology provides users a great communication experience and offers Service Providers assurances that network, bandwidth and delay issues will be overcome.

Global IP Solutions announced that it has been selected by LG Mobile to provide HD voice for its Fixed Mobile Convergence solution. FMC allows telecom operators to combine fixed telephony, cellular telephony and Broadband into unified end-user services.

Global IP Solutions announced the availability of high-definition and Super-wideband Voice as well as one-way Video Conferencing/Chat capabilities for iPad developers, powered by GIPS VideoEngine Mobile.

Global IP Solutions, a provider of HD voice and video processing solutions, announced yesterday another first-to-market with GIPS VideoEngine for Android, which provides mobile developers the vital building blocks for integrating video conferencing/video chat into applications running Google’s Android mobile operating system.

Global IP Solutions is a voice and video processing software, enabling the development of video solution for video and voice over IP. They recently announced their new software available for android and iPhone platforms which will enable people to develop applications for video conferences.

ASC will demonstrate its new VoIP recording solution, EVOip 10.0 together with its new quality monitoring solution, INSPIRATIONpro 10.0, at UC Expo, London, England on March 10-11, at the Olympia Exhibition Center.

UC Expo is widely considered to be Great Britain’s leading business communications event focusing on collaboration, mobile, visual and voice solutions. More than 100 exhibitions and seminars will be presented for a target audience of IT and telecommunications professionals, system integrators, infrastructure providers and end users.

ASC’s new version of its award-winning VoIP recording software, EVOip 10.0, offers the strictest adherence to security requirements, meeting the payment card industry’s PCI DSS standards. Other new features include one-click backup and dynamic enterprise licensing. New and enhanced integrations with Aastra MX ONE and Avaya ACM complement the existing range of integrations with major IPT vendors including Alcatel, Mitel, Siemens, Cisco, 3Com.

EVOip captures telephone calls from the network and enables storage, playback and archiving of the entire interaction. The software can record encrypted calls, and is entirely software based. Many functions may be accessed directly through the IP phone, such as record-on-demand, start/stop, keep/delete and search-and-replay for recorded calls.

INSPIRATIONpro 10.0, ASC’s new quality monitoring solution, now facilitates agent evaluations through the recording of coaching sessions. It also allows complex searches of audio analytics, particularly useful for high-volume call centers with an otherwise unmanageable number of conversations.

INSPIRATIONpro helps call center managers learn about their agents’ service level through analysis and evaluation of recorded call data and screen activities. Other features include real-time contact between agents and supervisors without the customer’s knowledge, and a user-friendly browser-based interface for access from any location.

HelloSoft announces the extension of its carrier proven VoIP and Voice Call Continuity solution suite to future 4G-LTEdevices. HelloSoft's VoIP and VCC solution allows seamless handoff of voice calls between VoIP on the 4G LTE network and circuit switched voice systems that run on cellular 2G and 3G networks. Supporting VoIP over LTE (VoLTE) extends HelloSoft's recently released, award winning VoIP and VCC Convergence Client Software platform for Android mobile devices, which also supports handsets running Windows Mobile, Symbian and Linux operating systems.

HelloSoft is extending its partnership with Beceem to provide a complete platform for VoIP and VCC on Beceem's new 4G multi-com chip that supports both LTE and WiMAX. This extends the collaboration between both companies which have already enabled VoIP and VCC solutions to run on 4G-WiMAX devices. Beceem's 4G-LTE/WiMAX chip is designed to support session hand-off between WiMAX and LTE, which is a key enabler for HelloSoft to show its VCC and convergence software client providing unified communications across multiple 4G networks.

Adaption Technologies announces the release of ExamiNet, a remote LAN assessment tool designed specifically to meet the site prequalification and technical support needs of VoIP providers. ExamiNet provides a quick and convenient method of assessing a client's LAN to determine its ability to support VoIP services. The ExamiNet platform also provides a secure and efficient means of collecting the assessment data and allowing access to the data via a web portal. This assessment process is achieved with use of an application that was designed with the chief principle of simplicity for the client end-user. This simplicity removes the need for any technical understanding of networks, PCs or VoIP telephony by the client to collect this information and making it available to the technical support staff responsible for interpreting the data.

ExamiNet executes a suite of low level network tests from a clients host PC that is directly connected to a candidate LAN for the purposes of determining that network's current ability to support VoIP services. The ExamiNet test suite includes procedures that determine the status and support characteristics for key network protocols, services and settings necessary for hosting an acceptable level of quality VoIP service. These tests include: DHCP, DNS, firewall, routing, bandwidth, and QoS.

snom has teamed with Jazinga for a technology partnership that continues snom’s leadership in interoperating with open source Asterisk-based VoIP platforms. Through the pairing, businesses can now use snom’s portfolio of VoIP phones in concert with Jazinga’s all-in-one MGA120 appliance to ease deployment of VoIP systems and reduce IP telephony costs in the SMB. The interoperability between snom’s suite of SIP based business phones and Jazinga’s MGA120 IP PBX appliance creates a fully integrated end-to-end IP-based phone system for small businesses that blends robust enterprise-class IP PBX and networking features at a dramatically lower price point than alternative solutions.

Jazinga’s MGA120 IP PBX router is an Asterisk-based VoIP phone system that allows small businesses with up to 20 employees to take advantage of features like wireless networking, WAN connectivity, LAN services, dynamic DNS, e-mail and fax. The Jazinga appliance costs a fraction of other enterprise-grade systems and can be set up in less than 10 minutes via a do-it-yourself set of setup wizards. Additional telephone features offered by Jazinga include an auto-attendant, voicemail, conferencing, call forwarding, unified voicemail-to-email, on-hold music and more. Jazinga also incorporates QoS and firewall capabilities.

The new partnership enables Jazinga users to take advantage of snom’s portfolio of SIP-based commercial-grade VoIP phones, including the snom 320, snom 360 and snom 380, and the new snom 820 and the snom 870 touch-screen phone, all of which integrate with an Asterisk platform. Additional snom products include the snom MeetingPoint conferencing phone and the snom m3 IP DECT phone. Along with interoperating with Jazinga’s IP PBX, snom’s intelligent SIP endpoints also support a variety of unified communications applications including Microsoft Office Communications Server 2007 R2 via its special snom OCS firmware.

Siemens Enterprise and Sybase announced a partnership to develop jointly new fixed-mobile convergence solutions that will fully integrate in-office data and voice features with workers' mobile devices, according to the companies. Sybase's mobile device management technology will be coupled with Siemens Enterprise's fixed-mobile convergence solution to enable support for multiple mobile devices.

The companies said this partnership will yield a product that simplifies inventory and logistics for the organizations that deploy it. According to Siemens Enterprise, it chose Sybase because of its "rich expertise and proven leadership" in enterprise mobility. Sybase's Afaria simplifies the management of the various mobile endpoints that employees may have, and it will be integrated into the Siemens Enterprise MobileConnect v2.3 product.

For more:
- see the FierceMobileIT article here
- see the companies' press release here 

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Intel is targeting next-gen VoIP applications with its new Xeon processor tentatively called Jasper Forest, which the company said will be launched in the first half of 2010. The new processor will support more dense communications solutions while also lowering system power consumption from previous models.

The new chip also comes with "non-transparent bridging functionality," which will allow multiple endpoints to connect internally without the need for an external PCIe switch, according to an xchange article. Intel said the new design would let telecommunications companies move away from needing multiple architectures and platforms in their network core, by bringing all system designs into a holistic offering. 

For more:
- see the xchange article here 

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C2Call launches with FriendCaller 3 its iPhone OS 3.0 version of FriendCaller, the only iPhone app on iTunes that can establish a free Internet call with anyone on the Internet. This breaks with the convention of allowing free calls only between registered users of a service who had to have installed identical software first.

FriendCaller is an easy-to-use, browser based VoIP solution allowing users to make free calls from the website to anyone on the Internet, with no software installation for the “callee” or person being called. Additionally, the invited call participant is not even required to register. The peer-to-peer calls work with any browser on Windows, OS X and Linux, and now with FriendCaller 3 as an OS 3.0 app for the original iPhone, iPhone 3G, 3GS and the iPod touch (2nd Generation).

FriendCaller 3 defines a new, much more user-friendly and intuitive standard for VoIP apps on an iPhone. “The days when users first had to sign up, then download and install software and exchange user IDs prior to talking with each other are gone,” said Martin Feuerhahn, CEO of FriendCaller developer C2Call GmbH.

FriendCaller 3 users simply send the patent pending Callme links directly from their iPhone via email to anyone on the Internet, or place their personal Callme links on their social media profiles, i.e., FaceBook, Twitter, etc. A web-phone starts once the link is opened on any browser with no installation or registration requirements for the person receiving the call invitation. That person simply clicks and connects!

FriendCaller 3 makes great use out of a new iPhone OS 3.0 feature called “Push Notification.” With Push Notification, a FriendCaller user whose app is not open receives call notifications and instant messages (IM) on his or her iPhone screen which appear as a text message. By opening the notification, the app starts, and the user can call immediately if in WiFi, or text via IM in Edge/3G.

With OS 3.0, Apple also has also opened “In App Purchasing,” which makes paying for FriendCaller Call Credit even more user-friendly. FriendCaller calls are free, and only calls to phone numbers are charged by the minute. FriendCaller provides global coverage, with calls to most countries in North America and Europe cost only 2 cents per minute.

In App Purchasing will be available with the soon-to-be released advertisement-free FriendCaller 3 Pro version. Devices on OS 2.0 are still supported by the FriendCaller Instant VoIP app available on the iTunes App Store. Users can continue to keep and use their user ID and password for all FriendCaller app versions and the FriendCaller Website.

Business Process Outsourcing provider Scicom Americas announced that Cross Telecom and Sipera will provide Unified Communications security appliances and SIP trunk termination for its distributed call center services. Offering outsourced, distributed call center services to Fortune 500 and other large enterprise clients, Scicom has deployed Avaya's UC and VoIP technologies. Working with telecom integrator Cross Telecom, Scicom will leverage Sipera's UC-Sec appliances to provide security and SIP trunk termination for its distributed call centers. 

Sipera's UC-Sec security appliance provides VoIP and UC security on multiple levels, including privacy (encryption), access control, threat mitigation, and security policy enforcement. By leveraging the added security features of Sipera's UC security appliance, Scicom's clients will be able to meet specialized security requirements, such as PCI DSS (credit cards), HIPAA (healthcare privacy), GLBA (consumer data protection), and FERPA (student record privacy).

"Our clients can rest assured that their mission-critical call center communications handled by Scicom are secure," said Ramesh Ariyanayakam, Chief Operating Officer, Scicom Americas in a press release. "We chose to work with Sipera because its innovative VoIP and Unified Communications security solutions enable our clients to rely on us to safeguard their communications."  

For more:
- see the press release here

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IPsmarx Technology and DIDX have agreed a partnership with the aim of allowing VoIP service providers to expand their global reach.
The VoIP solutions provider delivers Softswitch and Calling Card Solutions to operators in 63 different countries.

Sipera Systems now employs Empirix’s Hammer VoIP testing solution, enabling them to develop superior VoIP and UC security solutions. With the Hammer solution, Sipera is able to discover, prioritize and correct potential issues during the pre-deployment phase, reduce the cost of bringing its products to market and ensure that its UC-Sec security appliances can simplify and protect VoIP and UC implementations with comprehensive security.

During its evaluation, Sipera confirmed that the Hammer solution provided unrivaled performance testing for high capacity throughput, platform stability and reliability. To ensure the quality and reliability of their products, Sipera’s development and testing team now relies on Hammer’s extensive media support, security test functionality, high load handling capabilities and ability to provide detailed QoS metrics. Hammer also provides broad signaling and security protocol test coverage (including SIP, SCCP, TLS and SRTP).

Hammer testing solutions emulate the realistic behavior of users and devices interacting with voice-centric applications. It makes calls, verifies features, measures voice quality and determines performance, as well as proving system and multi-vendor interoperability for VoIP, NGN and IMS architectures.

VoIP solutions are likely to benefit from more people around the world seeking to access the internet using satellite technology, according to research from Global Industry Analysts.
The technology has been tipped as a possible way to provide broadband services in more remote, rural communities where it will be much more difficult to deploy conventional broadband infrastructure.

VoIP solutions provider MetaSwitch has announced a strategic alliance with one of Argentina's leading systems integrators as part of its on-going expansion into the South American market.
The link-up with Technology Bureau will enable it to offer advanced IP voice services to residential and business subscribers.