VoIP Monitor - VoIP Solutions

Vertex Telecom Selects RedShift Networks as the VOIP/SIP Security Solution for Enterprise and Wholesale Customers

VoIP Monitor — Tue, 02 Oct 2012 21:20:15 GMT

Vertex Telecom has selected RedShift Networks to provide their Unified Communications, Collaborations, VoIP and Video UCTM Security technology; which is currently being switched on Vertex Telecom VOIP/SIP network. Additionally, Vertex Telecom has selected RedShift Networks to develop three custom program models for wholesale into three sectors: Financial, Healthcare and Government. The three additional models will work in conjunction with the “core” and “edge” products installed at Vertex Telecom.

RedShift Networks has deployed its UCTM appliances as a first step to further protect and lock down the “core” and “edge” of Vertex networks. According to Amitava Mukherjee, president, CEO and founder of Redshift Networks, “Our UCTM appliance security solution includes a patented behavioral learning engine that automatically gathers network intelligence identifying potentially dangerous vulnerabilities. Coupled with Redshift’s global threat signature database and blacklisting service, Vertex now has a comprehensive UC/VOIP threat management solution built into its core network.”

“We are very excited with our security solutions being implemented into award winning facilities like Vertex Telecom’s. We expect to grow our enterprise business with Vertex into the Philippines and Taiwan telecom markets with the rest of this year and in 2013.” RedShift Networks, Robert Barker, Regional Sales Director, RedShift Networks.

Eyeball Networks Brings VoIP Connectivity and Interconnection Solutions to Network Service Providers at ITExpo West

VoIP Monitor — Tue, 02 Oct 2012 21:14:04 GMT

Eyeball Networks will be at ITExpo West in Austin, TX. Oct. 2-5, 2012, showcasing mobile VoIP connectivity and interconnection solutions to conference attendees.

Reliable VoIP connectivity for mobile enterprise users is an important issue to service providers. Solving connectivity challenges is only the first step, as interconnection is the next big subscriber need - interconnection between devices, platforms, and communities.

Eyeball Networks enables network service providers to develop fully customizable VoIP communication applications and services using Eyeball’s Messenger SDK and server products that guarantee 100% connectivity. Interconnection is provided via Eyeball’s AnyConnect Gateway which delivers unsurpassed ability to bridge networks, platforms, and standards.

Eyeball has a suite of solutions available to help service providers, all based on Eyeball’s patented technologies - AnyFirewall Technology, AnyBandwidth Technology, and AntiSPIT Technology.

Visitors at ITExpo West 2012 are invited to come by booth 407, across from the Lync Pavillion, to meet the Eyeball team and see the Eyeball technologies in action.

Edgewater Networks and Metaswitch Announce Joint Solution for Service Providers Delivering VoIP Services to Enterprises

VoIP Monitor — Mon, 01 Oct 2012 20:55:24 GMT

Edgewater and Metaswitch announcee the availability of a joint offering aimed at service providers delivering VoIP services to enterprises. The solution combines Edgewater’s VoIP monitoring and installation solution, EdgeView, with Metaswitch’s MetaView solution. It enables service providers to easily provision and support enterprise environments where a heterogeneous mix of IP phone devices from different manufacturers are in use. These environments are costly to install and maintain as there is no standard interface across all of the manufacturers that can be used to automate provisioning or perform advanced diagnostics. The two products, EdgeView and MetaView, deliver simplified, scalable provisioning and enhanced visibility into the performance of today’s VoIP networks.

The combined solution provides the following benefits:

Plug & play IP phone installation: Edgewater’s EdgeView solution complements Metaswitch’s Session Initiation Protocol (SIP) solution by enabling the deployment of SIP-based IP phones from a wide variety of phone vendors such as Cisco, LG, and Polycom. The solution eases installation by allowing the end-user to simply plug a phone into a network powered by Edgewater’s EdgeMarc ESBCs and then follow a series of intuitive voice prompts. EdgeView then authenticates the user via Metaswitch’s Multimedia Telephony Application Server and automatically updates the phone’s software and the subscriber’s phone configuration. The customer is instantly on-line and ready to make calls.
Improved out-of-box experience for end-users: The combined solution uses a series of easy to follow voice prompts that provide a substantial improvement over alternative methods of configuration.
Seamless product integration using advanced APIs: The joint solution uses an authenticated API to hide systems complexity from network operators.
Easy to add new phones: As new IP phone models are approved for the VoIP service, network operators use a template driven interface to provision qualifying phones.
Real-time on-premise diagnostics: By deploying the joint management solution with Edgewater ESBCs at the customer premise, the service provider has remote access to a wealth of VoIP call quality performance statistics. These can be used to quickly identify and resolve problems the customer may encounter with their hosted VoIP service.

With today’s announcement Metaswitch becomes a member of Edgewater’s Plug & Dial Alliance program that enables service providers to significantly shorten hosted VoIP installation times and simplify ongoing moves, adds and changes. In addition to offering interoperability between, and automated setup of, many leading brands of SIP-based IP phones, the Edgewater Plug & Dial Alliance program continuously tests new devices and adds them to the program.

SolarWinds Introduces Free VoIP Management Tool - VoIP Call Detail Record Tracker

VoIP Monitor — Wed, 29 Aug 2012 21:17:05 GMT

SolarWinds announces the release of SolarWinds VoIP Call Detail Record Tracker, a robust free tool to help IT pros manage VoIP with Call Detail Record tracking, and the only free tool of its kind in the VoIP management market that allows users to search and display CDRs.

SolarWinds determined the need among IT pros for a VoIP CDR Tracker free tool based on demand and discussion on thwack®, the SolarWinds community of over 100,000 IT pros that offers a space to request new products and features, as well as share insight and address IT management challenges.

SolarWinds VoIP CDR Tracker provides IT pros with a desktop utility to track the performance of VoIP calls by searching, filtering and sorting Cisco CallManager call detail records. IT pros can quickly view key details about the call, including originating and destination number, originating and destination IP address, date, time, status, termination causes, and MOS. When an end user’s call is dropped, the IT pros can use SolarWinds VoIP CDR Tracker to assist in determining the root of the problem and find its solution.

SolarWinds VoIP CDR Tracker Highlights:

Search, retrieve, and view CDRs
Load up to 48 hours of CDR data
Support for Cisco CallManager CDR files

For a comprehensive VoIP management solution, SolarWinds VoIP & Network Quality Manager (formerly IP SLA Manager) monitors the performance of individual VoIP calls by analyzing call quality metrics available within the call detail record and provides real-time alerts when critical thresholds are exceeded. Coupled with its proactive WAN performance analysis capability, VoIP & Network Quality Manager will allow IT pros to troubleshoot and solve VoIP QoS problems faster and more effectively.

Pricing and Availability

SolarWinds VoIP CDR Tracker is free. Pricing for SolarWinds VoIP & Network Quality Manager starts at $1,495.

SolarWinds Strengthens VoIP Performance Monitoring and Proactive WAN Performance Analysis

VoIP Monitor — Wed, 01 Aug 2012 20:23:55 GMT

SolarWinds announces the upcoming release of SolarWinds VoIP & Network Quality Manager, formerly known as SolarWinds IP SLA Manager, to help IT professionals maintain the highest level of VoIP network performance and voice quality.

SolarWinds VoIP & Network Quality Manager monitors the performance of individual VoIP calls by analyzing call quality metrics available within the call detail record and provides real-time alerts when critical thresholds are exceeded. Coupled with its proactive WAN performance analysis capability, VoIP & Network Quality Manager will allow IT pros to troubleshoot and solve VoIP problems faster and more effectively.

SolarWinds VoIP & Network Quality Manager can be deployed as a standalone product or fully integrated with SolarWinds Network Performance Monitor, offering IT pros the flexibility to target VoIP and WAN monitoring and troubleshooting or gain a unified view of network performance.

SolarWinds VoIP Network Quality Manager (formerly IP SLA Manager) Highlights:

Monitor VoIP Call Performance - Monitor key VoIP metrics including jitter, latency, packet loss, and MOS by analyzing call detail records generated by Cisco CallManager. With SolarWinds VoIP & Network Quality Manager, users can configure real-time VoIP network alert notifications when specific voice quality thresholds are violated and then search for potential patterns within the same region, timeframe or reason code to troubleshoot the issue.
Troubleshoot VoIP Call Performance - Correlate individual call performance with corresponding network performance through the creation and association of IP SLA operations and CDRs.
Search and Filter Call Detail Records - Search and filter on the data found in every call detail or call management record. Using the pertinent details behind the call, users can use VoIP & Network Quality Manager's advanced troubleshooting capabilities to determine just what caused that poor quality. VoIP & Network Quality Manager retains CDR information up to one month, enabling users to search and view historical VoIP call details.
WAN Performance Monitoring - Monitor WAN performance by tracking key edge-to-edge router performance statistics using Cisco IP SLA technology. In addition, VoIP & Network Quality Manager keeps an eye on key applications by analyzing the performance of the underlying network protocols, including DNS lookups, FTP, HTTP, TCP connect, and UDP jitter.

The complete SolarWinds network management product portfolio includes SolarWinds NPM, SolarWinds Network Configuration Manager, SolarWinds NetFlow Traffic Analyzer, SolarWinds IP Address Manager, SolarWinds VoIP & Network Quality Manager, SolarWinds User Device Tracker, SolarWinds Log & Event Manager, Engineer's Toolset and LANsurveyor.

Vera Networks Launches Cloud-Based VoIP Call Flow Diagnostics Tool vCapture

VoIP Monitor — Mon, 30 Jul 2012 21:07:36 GMT

Vera Networks announces the availability of its vCapture service that enables operators of VoIP networks and platforms to quickly locate and help resolve call issues.

The vCapture service provides technical teams with the ability to target call issue-related SIP signaling data with laser-point accuracy without being in critically sensitive areas of the network infrastructure. The non-invasive, cloud-based, SIP signal capture technology records all SIP messaging within a switching network. This allows for in-depth analysis of session initiation protocol SIP signaling data, including call flow, signaling, and equipment-related problems.

The cloud-based solution, offered on a monthly subscription basis, reduces the time required to determine the root cause of most VoIP call and interop-related issues, improving customer response times, resource utilization, and reducing related operational expenses. It boasts an impressive array of features including an intuitive interface and protocol-aware searching and filtering capabilities.

Vera's vCapture service is the first cloud-based SIP capture solution and is capable of storing massive amounts of SIP signaling data, which then allows for retrieval of actual call information long after an event occurred.

The service was introduced in Chicago at International Telecoms Week earlier this year, and received enthusiastic reviews from the carrier community.

Global Mobile VoIP Solutions Market to Grow at a CAGR Of 64.6 Percent Over The Next 3 Years

VoIP Monitor — Tue, 12 Jun 2012 21:30:24 GMT

The Global Mobile VoIP Solutions Market is expected to grow at a CAGR of 64.6 percent over the period 2011-2015. One of the key factors contributing to this market growth is the increasing global network and wireless bandwidth capabilities. The report is based on an in-depth study covering the Americas, and the EMEA and APAC regions. The report aims to aid decision makers' understanding of the significant trends impacting this market.

Commenting on the report, an analyst from TechNavio, the author of this report said; "The Global Mobile Voice Over Internet Protocol Solutions market is witnessing the emergence of many new players because of the presence of several factors such as low entry costs and huge business opportunities. These new companies are trying to penetrate the market by offering low-cost communication services. This has resulted in a price war among vendors in the market and the established companies are losing their market share to the new entrants. This trend is expected to grow in the next few years with the market expected to witness the emergence of new global and regional players."

According to the report, since communication services are constantly in use, there is high demand among users for low call tariffs. Since mobile VoIP services are offered over IP-based communication services and require comparatively low investment as compared to legacy networks, over the top providers are able to offer Mobile VoIP solutions at a much lower cost. This is one of the main drivers in the Global Mobile VoIP Solutions market.

Further, the report also discusses how the unavailability of high-speed data connections can adversely impact the growth of the Global Mobile VoIP Solutions market.

Nymgo Positioned as the Most Competitive VoIP Solution Available

VoIP Monitor — Mon, 04 Jun 2012 20:39:58 GMT

Nymgo announces the official launch of an enhanced PC application and new mobile apps that offer the lowest rates and best quality VoIP calls from iPhone, Android phones and desktop computers to cellphone numbers and landlines anywhere in the world. Nymgo users will now be able to make the clearest VoIP calls possible on their preferred mobile or computing device, with the lowest rates available anywhere.

After four years of operation, Nymgo has proven to be one of the most reliable and inexpensive VoIP service providers on the market - with the cheapest calls to countries like Canada, the United States, China, India and more - and with recent funding from renowned equity firms Intel Capital and Abraaj Capital, Nymgo has been able to enhance its VoIP network while upgrading the apps that deliver that service to make the Nymgo experience even better. Nymgo will also use the new funding to continue growing as a VoIP platform.

Current iPhone, Android and PC users can download the new Nymgo app from the App Store, Google Play Store, or the Nymgo website, respectively.

WUN Systems Announces Extended VoIP Solution for iPhone, Android, and Mobile

VoIP Monitor — Thu, 31 May 2012 21:07:39 GMT

WUN Systems announces the launch of four new features to their VOIP telephone system for executive suites and workspaces. These new features were designed to extend mobile capabilities for workspaces and business centers. Executive suites will now have extended hosted VOIP capabilities directly through WUN Voice Softphone, Android, iPad and iPhone Edition.

WUN Voice Softphone is a carrier-grade, next generation Softphone application that enables the user to manage their communications easily and efficiently; all from a computer desktop. “Built on the WUN Voice platform, WUN Voice Softphone is specifically designed to meet the demands of business centers and workspaces. WUN Systems also included exclusive features designed for business and enterprise users. These new features can be deployed within an enterprise environment either by manual configuration via the Softphone Graphical User Interface or by using the WUN Voice positioning server” explains Dale Hersowitz from WUN Systems. The latest in Softphone technology, WUN Voice Softphone enables a dynamic interface with new task flows and customization options for an unprecedented user experience.

WUN Voice Android Edition is a highly secure, standards based mobile VOIP Softphone that works over both 3G and Wi-Fi networks. WUN Voice Android Edition facilitates easy and effective communication management with an intuitive interface. Deskphone-class functionality includes the ability to switch between 2 calls, merge calls and perform attended and unattended transfers. WUN Systems incorporates advanced security settings which allow for secure call signaling and works seamlessly with other WUN Voice desktop and convergence solutions.

WUN Voice iPad Edition is a VOIP Softphone that was designed to be specifically compatible with Apple iPad and iPad2 that uses a Wi-Fi or 3G connection to make and receive calls. “This new addition facilitates easy communication with an intuitive interface that includes all of the standard phone features such as call display, call history, voicemail indicator, and multiple call support” says Dale Hersowitz from WUN Systems. WUN Systems also integrated a multi-tasking functionality that allows WUN Voice iPad Edition to run in the background so users can access other applications while on a call. Advanced security settings are also integrated which allow for secure call signaling and audio encryption. Premium feature add-ons are available including a Presence and Messaging add-on as well as a low bandwidth audio code for bandwidth management.

WUN Voice iPhone Edition is a VOIP Softphone for Apple iPhone and iPod touch that uses a 3G/4G or Wi-Fi connection to make and receive calls. Using the iPhone or iPod's existing contact list, WUN Voice iPhone Edition facilitates easy and effective communication management with an intuitive interface that accommodates multiple calls. “Call functionality includes the ability to swap between two calls, merge and split calls and perform attended and unattended transfers” continues Hersowitz. WUN Systems also incorporates advanced security settings for secure call signaling and audio encryption. Premium feature add-ons for this hosted VOIP Softphone include Video Calling, Presence and Messaging and a low bandwidth codec for bandwidth management. These features can be purchased within the application under the Premium Feature Settings tab to enhance the user’s communications experience.

OAISYS Advanced Call Recording Joins SterlingVOICE Hosted VoIP Platform from Sterling Communications

VoIP Monitor — Wed, 09 May 2012 20:38:51 GMT

OAISYS announces a new strategic relationship with Sterling Communications. This relationship is designed to bring advanced call recording functionality to customers of its SterlingVOICE VoIP business communications solution. The solution integration provides numerous benefits to customers in critical areas of business concern, including quality assurance and personnel development, risk management and regulatory compliance, dispute resolution and communication archiving, and work productivity enhancement.

Sterling Communications' SterlingVOICE hosted VoIP solution enables companies of all sizes to enjoy easy and affordable access to advanced communication features and functionalities without the worries and expense of owning, administering and maintaining their own equipment, all for a set, per-user subscription cost that makes budget planning simple. By partnering with OAISYS for advanced call recording capabilities, including the most advanced set of regulatory compliance features in the industry, Sterling Communications provides its users with a complete, cost-effective solution for all their voice, documentation and business process assurance needs.

Sangoma Releases Transcoding Gateway Appliance

VoIP Monitor — Wed, 02 May 2012 21:12:07 GMT

Sangoma announces the release of its NetBorder Transcoding Gateway, the latest product in the growing range of gateway appliances to serve enterprises worldwide. This new gateway facilitates the interconnection of VoIP devices and networks with support for transcoding between a wide range of codecs-from low-bandwidth to HD-that are employed throughout the rapidly growing enterprise communications landscape.

The new NetBorder Transcoding Gateway provides bandwidth savings in interoffice VoIP and SIP trunking deployments with support for narrowband G.723.1 and G.729 codecs. Additionally, the appliance enables seamless connections between various SIP implementations such as those using different RTP packet sizes.

Beyond delivering bandwidth efficiency, the Transcoding Gateway offers many options for consistently high voice quality. In addition to the wide variety of narrowband codecs, support for HD voice codecs such as G.722 and G.722.1 makes it possible for HD and non-HD compliant devices to interoperate. On networks that are susceptible to packet loss, and the attending degradation of voice quality, more resilient codecs such as iLBC can be used to improve the performance and deliver solid call quality.

Available as a 1U appliance, the NetBorder Transcoding Gateway allows transcoding between any supported codec from a wide range including, iLBC, GSM-FR and as well as all common SIP and HD voice formats. These can be selected on a per call basis allowing for complete flexibility and the gateway can support up to 4000 simultaneous SIP sessions. The appliance provides very low latency processing to ensure low-jitter and excellent call quality.

ASC to Demonstrate Solutions for Financial Institutions at MEFTEC in Dubai

VoIP Monitor — Tue, 17 Apr 2012 00:31:59 GMT

ASC will demonstrate its quality management solution INSPIRATIONpro and VoIP recording solution EVOip, at the MEFTEC financial technology conference, April 25-26, 2012, at the International Convention and Exhibition Center in downtown Dubai. ASC will exhibit at booth E2.

In its eighth year, MEFTEC is considered the world's premier financial technology conference for emerging markets, with last year's event hosting 300 delegates from 27 countries. ASC will show financial service providers how to use INSPIRATIONpro and EVOip to analyze the level of service, optimize workflows and protect themselves from liability.

Marco Mueller, Executive Vice President of ASC, said, "Financial institutions represent one of our key partner areas, and our communications recording and quality management systems are designed to help them meet compliance requirements, provide fail-safe verification of transactions and ensure protection from liability. Our portfolio works for nearly any infrastructure and has been certified for interoperability by major telecommunications providers."

Mr. Mueller went on to describe the solutions in more detail. INSPIRATIONpro helps call center 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. It also allows complex searches of audio analytics, particularly useful for high-volume call centers with an otherwise unmanageable number of calls.

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

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.

snom Introduces ONE IP PBX with Advanced Mobility Features

VoIP Monitor — Wed, 01 Feb 2012 22:48:47 GMT

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.

Aria Technologies and Engin Partner to Deliver Business-grade VoIP Solutions

VoIP Monitor — Mon, 21 Nov 2011 23:15:21 GMT

Aria Technologies announces a partnership with engin. This partnership culminates in the availability of business-grade VoIP solutions that combine advanced IP telephony equipment from LG-Ericsson, with cutting-edge IP network infrastructure from engin, Australia’s first VoIP service provider.

Completely NBN-ready, the new packages will allow businesses to take advantage of VoIP technology immediately, providing savings on their communications costs with aggressive call pricing as well as providing a broad and comprehensive feature set.

Available from early December, customers can take up the new packages from Aria and engin that incorporate a monthly line service fee as low as $7.50 per trunk, with untimed 10 cent calls to local and national numbers. To promote the launch of this partnership engin is offering all new customers free access for the first 2 months.

snom Unveils New Provisioning Solution to Accelerate Enterprise VoIP and Unified Communications Deployments

VoIP Monitor — Wed, 02 Nov 2011 02:37:20 GMT

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 and Open Source Community Release Asterisk 10 at AstriCon

VoIP Monitor — Fri, 28 Oct 2011 18:02:19 GMT

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.

Voxbone Creates VoxTRUNK

VoIP Monitor — Tue, 18 Oct 2011 23:29:41 GMT

Voxbone announces the introduction of VoxTRUNK, in a move that helps enterprises increase efficiency, lower costs and simplify management of their inbound voice services. The new offering enables multinational companies to purchase global voice channels that can be used simultaneously to receive calls on both geographical and toll-free numbers.

Until today, customers had to buy separate channels for geographical and toll-free services. VoxTRUNK is the first global solution that enables a customer to purchase a single channel for both geographical and toll-free services, and it is available across Voxbone’s entire footprint, spanning more than 50 countries and 4,000 cities.

“Responding to customer requests, we are launching VoxTRUNK to make buying capacity more efficient and flexible,” said Voxbone CEO Rod Ullens. “Enabling geographical and toll-free services to share channels allows our customers to save money and to streamline service setup and management.”

Ullens said VoxTRUNK is particularly beneficial for customers with large calling requirements, such as call centers, conferencing-service providers and calling-card providers, because they typically offer both geographical and toll-free access numbers.

Voxbone Extends Service to Vietnam

To further benefit its growing customer base, Voxbone also announced the extension of its coverage to include Vietnam.

“Vietnam has one of the fastest-growing economies in the Asia Pacific market, and we’re pleased to make it the 51st country in our expanding global footprint,” Ullens said.

Global IP Telecommunications Releases Free Plug & Play Solution for VoIP Services that Enables VoIP Even in Restricted Networks

VoIP Monitor — Thu, 13 Oct 2011 23:33:28 GMT

Global IP Telecommunications releases a free plug & play software to supplement SIP VoIP services in order to make unobstructed telephony available at WIFI hotspots, in hotels and other restricted networks. The product SSC, "Simple SIP Channel," provides for unprecedented freedom in telephony. The actual VoIP service provider can be chosen freely.

The development took two years from concept to realization. The quality and speed of the data transport mechanism have been the focus of the tap-proof point-to-point encryption of voice data. SSC is freely available to every interested person from today. A Linux and Windows variant of the software can be downloaded from www.globaliptel.com. There is no limitation of use. SSC can be used with all Ninja Software Telephones from Global IP Telecommunications. Implementation of the relevant functionalities in third-party hardware and software is stipulated as well.

Data-Max Wireless Selects VoxOx VoIP Platform

VoIP Monitor — Tue, 20 Sep 2011 21:21:55 GMT

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.

NextiraOne Drives New Communications Solution for Austrian Engine Manufacturer

VoIP Monitor — Tue, 20 Sep 2011 21:14:26 GMT

ELIN Motoren has chosen NextiraOne to implement an upgraded communications network throughout the company's Austrian sites, as part of the company's project to construct its new headquarters in Preding/ Weiz.

At the heart of the new communications infrastructure is an IP-based converged network based on Alcatel-Lucent technology. The Gigabit Ethernet network for voice and data communications provides Power over Ethernet delivering high performance and reliability to the desktop.

A wireless LAN provides mobile voice and data communications in the company's warehouse, which houses some 3,500 pallets in 12 metre high racking, and in parts of the office complex in the company's new headquarters. This network, using Alcatel-Lucent's latest wireless LAN technology, has proved to be very successful. In spite of the potential for breaks in transmission due to the steel construction of the building and the racking, the transmission of verification data via hand and barcode scanners using the 65 wireless access points throughout the building has proved to be entirely reliable.

As part of the upgrade of the telephony system, the staff of ELIN Motoren GmbH have been equipped with IP telephones. Over 120 IP telephones and 100 DECT phones now provide reliable communications throughout the entire network in the headquarters Preding/ Weiz and at the company's Vienna and Salzburg sites. GSM integration also enables mobile telephony off-site as well as in the offices.

The central management system handles the administration of the network and all devices. New software is installed and updates and new configurations are carried out remotely without disrupting ongoing operations.

NextiraOne has also implemented a comprehensive security solution with Unified Threat Management. The Juniper Enterprise Firewall protects the whole network from malicious attack. Tools such as anti-virus, intrusion prevention, web filtering and spam filtering ensure the security of the applications and protection of devices and terminals. The Vienna and Salzburg sites and remote users are all connected to the headquarters Preding/ Weiz via a virtual private network.

Dominik Brunner, Managing Director of ELIN Motoren GmbH, said: "The substantial requirements we had set for the communications system and the short timeframe for the implementation of the project created a major challenge. We are therefore all the more pleased with the professional approach of the NextiraOne team and the benefits of the new solution, which covers our entire communications infrastructure and has simplified it significantly.

Long-term cost optimisation and the ease of use of the system were also deciding factors according to ELIN Motoren GmbH's Managing Director Gustav Hauschka.

Dr. Margarete Schramböck, Vice President of NextiraOne Germany and Austria said: "Our long-standing experience and comprehensive know-how in the integration of voice, data, security and mobility have helped make this project such a success. In particular the implementation of the wireless LAN is another milestone in the provision mobile network technology."

Trisys Introduces Replay SIP

VoIP Monitor — Fri, 09 Sep 2011 21:16:30 GMT

Trisys introduces Replay SIP, a scalable module of its popular Replay Call Recording solution that is easily added to IP-based telephony systems. As business adds IP phones to existing telephony systems in order to take advantage of cost efficiencies, access to phone application software is often lost or requires expensive upgrade. Now with Replay SIP business can freely add call recording functionality for less than $300 per phone. The small footprint, scalable product also moves Replay to the forefront of options for new, predominantly IP phone system sales.

Trisys’ Replay SIP is a 100% software based call recording solution. It is designed to record phone conversations taking place on SIP-based IP phone systems. Replay SIP runs unobtrusively on networks, monitoring VoIP traffic for desired calls, and converts them in to call recordings. With Replay SIP installed, authorized users can easily access call recordings for quality assurance, regulatory compliance, dispute resolution, and much more.

Replay SIP supports most SIP-based IP telephone systems, saves recordings as standard WAV files, which can be automatically archived or deleted, supports On-Demand and Pause/Resume recording providing that PBX and IP Phones support RTP “events” as per RFC 2833/4733. Replay SIP is available as a software "only" or as a complete turnkey solution including software and hardware.

ASC to Promote VoIP Recording and Quality Management Solutions at Gitex Technology Week

VoIP Monitor — Tue, 30 Aug 2011 21:58:47 GMT

ASC will be demonstrating its VoIP recording solution, EVOip, and quality management system, INSPIRATIONpro, at Gitex Technology Week, Dubai International Convention and Exhibition Center, Dubai, United Arab Emirates, on October 9-13, 2011, at Germany's national booth.

Gitex connects 3,500 international ICT vendors with 136,000 industry professionals, helping visitors to identify new products, innovative solutions and consumer trends; find ways to give their business an edge within the market; and discover new contacts and sales channels. ASC's solutions, EVOip for VoIP recording, and INSPIRATIONpro for quality management, provide powerful features for businesses with multiple locations and complex infrastructures.

INSPIRATIONpro, ASC's quality management system, helps call center managers learn about their agents' service level through analysis and evaluation of recorded call data and screen activities. The latest version facilitates agent evaluations through the recording of coaching sessions and allows complex searches of audio analytics.

EVOip is a software-only solution designed to capture, store, search, play back and archive telephone calls from VoIP networks. The latest version handles encrypted conversations and meets the payment card industry's strict PCI DSS security standards.

XConnect Calls on AudioCodes to Support High-Definition Voice Transcoding Feature

VoIP Monitor — Mon, 22 Aug 2011 22:53:18 GMT

XConnect announces that its new high-definition voice transcoding feature, which enables HD voice traffic to cross the fixed-mobile divide through an IP-peering federation, has been developed in conjunction with AudioCodes.

Now available for members of the XConnect HD Alliance, the new capability draws on the combined strengths of XConnect’s carrier-ENUM registry and multimedia interconnection services and AudioCodes’ Mediant 3000 high-definition transcoder.

In a tested, certified end-to-end solution, XConnect’s interconnection hub detects the need for transcoding, prompting the AudioCodes transcoder to translate between the different codecs used by fixed-line and mobile operators.

As part of the deployment, AudioCodes will also become an HD Alliance partner and join the XConnect Ready Partner Program to support cooperation and co-marketing efforts between the companies.

“With the recent increase in HD voice uptake on cellular networks, predominantly relying on wideband AMR, and the widespread use of the G.722 codec on wireline networks, high-definition interconnect becomes crucial for a ubiquitous quality of experience for both wireless and wireline users,” said Jeff Kahn, Chief Strategy Officer of AudioCodes. “We are delighted to be working with XConnect to advance what we at AudioCodes call HD VoIP.”

XConnect CEO Eli Katz said: “We are very pleased that AudioCodes has chosen to provide its expertise to help us enrich the benefits of our high-definition voice offering. AudioCodes’ transcoding technology complements XConnect’s ENUM registry and interconnection services, and we look forward to collaborating in other ways to bring innovation to market.”

XConnect Launches Feature to Deliver HD Voice Between Fixed and Mobile Networks

VoIP Monitor — Mon, 22 Aug 2011 22:51:58 GMT

XConnect announces a new transcoding technology designed to promote wider adoption and utilization of high-definition voice services. The new feature for members of XConnect’s HD Alliance delivers clearer, higher-quality and more effective voice communications by enabling fixed-line and mobile operators to seamlessly exchange HD voice traffic through an IP-peering federation for the first time.

“We are solving the key challenge to widespread adoption of high-definition voice services: the need to deliver traffic across disparate networks,” said XConnect CEO Eli Katz. “The enhancement to XConnect’s HD Alliance extends the benefits of cross-network HD voice service to mobile operators via cloud-based transcoding, thereby enabling fixed operators to connect HD voice calls with mobile endpoints and vice versa.”

Katz noted that the new transcoding feature will be increasingly useful as HD voice gains traction across all networks – fixed, mobile and over the top offerings – for consumer and enterprise markets.

To date, technology barriers have stood in the way of HD voice communication between fixed-line and mobile operators, which deploy different codecs. The new feature bridges the disparate HD “islands” by enabling transcoding in the XConnect multimedia interconnection hub. The resulting interconnection offers new revenue opportunities for fixed and mobile service providers, as well as improved call quality for more users.

The transcoding capability expands the benefits of XConnect’s HD Alliance – the world’s original peering federation dedicated to HD voice. Since its launch in April 2010, the HD Alliance has supported service providers using the G.722 wideband audio codec, the ITU standard for HD-enabled fixed communications networks.

Now, using the new transcoding technology, HD voice traffic can be delivered with tonal and audio quality preserved between fixed operators using the G.722 codec and mobile operators using the AMR-WB codec, which is most popular for wireless networks.

The HD Alliance, which operates within XConnect’s Global Alliance worldwide peering federation, currently has 11 participating service providers.

After launching the HD federation for U.S. operators, XConnect in September 2010 expanded membership to include U.K. participants as part of an initiative with Polycom, BroadSoft and Dialogic to promote HD voice uptake in the U.K.

VoIP Supply Expands VoIP Convergence with Valcom IP Paging Solutions

VoIP Monitor — Tue, 26 Jul 2011 02:37:28 GMT

VoIP Supply is happy to announce the addition of new Valcom IP Paging devices for mass notification and real time IP voice.

Paging system solutions for schools, manufacturers, hospitals, and businesses were traditionally a separate analog based system. Elimination of that separate analog system is possible with Valcom IP page controls that tie into an existing IP network that most institutions already employ for data and VoIP phones.

VoIP Supply now carries Valcom’s latest IP network-based Page Controls, Speakers, Horns, Intercoms, and Clocks. Before introducing the industry’s first IP and IP PoE loudspeakers, Valcom was already recognized as having a long history of providing high quality paging products used by major telephone carriers.

“VoIP Supply is excited to offer Valcom’s latest line of IP paging devices,” said Garrett Smith, Chief Marketing Officer at VoIP Supply. “Utilizing Valcom’s advanced communication solutions is yet another option to save time and resources that many organizations may not be aware of; IP paging systems can easily converge into existing IP networks.”

Click here for additional information about the Valcom IP Paging devices available through VoIP Supply.

BeroNet Launches Cloud Managed VoIP Card & Gateway SaaS Solution

VoIP Monitor — Thu, 23 Jun 2011 00:21:42 GMT

BeroNet launches 'BeroCloud', the SaaS Cloud based Solution that empowers its rapidly growing base of Solution Providers and Carriers to manage their installed base of BeroNet devices through any web browser.

As a Cloud SaaS solution, BeroCloud now solves a serious industry problem empowering VoIP Solution Providers and Internet Telephony Service Providers to remotely monitor their systems, batch update firmware, backup configurations, recover settings, create projects and schedule automatic tasks. BeroCloud is now available in Beta and exclusively by invitation for BeroNet Certified Solution Providers.

Digium Adds Fixed Mobile Convergence to Switchvox

VoIP Monitor — Mon, 06 Jun 2011 18:22:25 GMT

Digium introduces Switchvox 5.0, a new version that adds fixed mobile convergence and further integration with third-party business applications into its full-featured and cost-effective VoIP unified communications solution designed for small- to mid-sized businesses. The new release enhances Switchvox mobility to allow users to seamlessly integrate any type of phone with Switchvox. Users can select up to six phones of any type, including VoIP, digital, analog, smartphone or a soft phone, to converge with their Switchvox extension. The user can now route, record or transfer calls appropriately, at any location. Users of Switchvox SMB with active subscriptions can download version 5.0 to have access to these features at no cost.

Switchvox 5.0 is based on Asterisk, the world’s most widely adopted open source communications engine, and brings a new level of features and customization to IT administrators, users and resellers. Additionally, version 5.0 includes more APIs for greater customization and integration with third-party business applications, giving businesses the ability to leverage their other IT investments, such as CRM, customer support, accounting and ERP systems. Switchvox 5.0 also features an enhanced user interface, more detailed reporting for calls and call queues, and detailed online support. With all of these features included in each Switchvox SMB unified communications system, customers can realize an average cost savings of up to 60 – 80 percent over comparable VoIP business phone systems.

The key Switchvox 5.0 UC features include:

Fixed Mobile Convergence – Built-in integration for six phones, allowing for seamless transfers and recording from any phone. Find out more: http://www.digium.com/mobility.
Detailed Call Queue Reports and Logs – Granular call queue data for multiple queues and queue members.
Additional Application Programming Interfaces – Organizations can create custom integrations with third-party business applications for communications-enabled business processes. See a complete list: http://www.digium.com/switchvox/api.
Switchvox Graphical User Interface – A newly refreshed Switchvox GUI simplifies the configuration for groups of users for administrators and users. See these and more features: http://www.digium.com/switchvox-features.

Pricing and Availability

Current Switchvox SMB customers can access Switchvox 5.0 directly through their unit today at no additional charge. For new Switchvox SMB customers, pricing starts at $3,195 for up to 30 users. To find out how to buy Switchvox, visit: http://www.digium.com/how-to-buy.

Cytracom Plans for More Growth with CoreXchange

VoIP Monitor — Thu, 02 Jun 2011 18:44:53 GMT

CoreXchange has been chosen to provide additional connectivity and data center services to Cytracom. Cytracom provides high-availability VoIP solutions through an exclusive partner network and owns and operates its own equipment on a fault-tolerant network engineered for high demand VoIP traffic. The company anticipates strong growth this year as a result of increasing market reach and awareness and the expansion of its channel partner network. Cytracom continues to rely on CoreXchange as it grows its network and data center infrastructure necessary to support and provide superior VoIP solutions.

Through its leading mesh of Tier-1 carriers and latest generation routing and switching equipment, CoreXchange provides fault-tolerant performance to meet high-demand network applications such as VoIP, e-commerce, media streaming, and online gaming. CoreXchange’s SAS70 Type II audited Dallas co-location facilities provide redundant power via redundant UPS and diesel generator backup, environmental controls, security, and 24/7 physical access. CoreXchange’s comprehensive support includes a 24/7 on-site network operations center offering free remote hands and eyes services.

Fonality Announces Trade-In Program to Help Growing Businesses Transition to Cloud-based VoIP Solutions

VoIP Monitor — Wed, 25 May 2011 17:43:10 GMT

Fonality announces the launch of its “Turn it In/Turn it Up” trade-in program, providing small and mid-size businesses with a cost-effective solution to upgrade their business communications capabilities. Through this program, SMBs can trade their outdated, legacy phone systems for fair market value and receive credit toward Fonality’s cloud-based Unified Communications, VoIP and contact center solutions.

Specifically designed for SMBs, Fonality’s six-time award-winning communications solutions are simple to use, easy to manage and affordable to deploy. The company’s cloud-based solutions deliver Fortune 500 features without the costly hardware, infrastructure or lengthy implementation cycles associated with legacy on-premise IP systems. Productivity-enhancing features, such as unified messaging with email, secure chat and Microsoft Outlook contact integration, are combined with audio conferencing, photo caller ID, visual voicemail, email/text, ring-back and on-the-fly call recording. Fonality solutions start at a cost of $30 per user per month, which includes calls, and offer a total cost of ownership up to 50 percent less than legacy phone solutions.

New Fonality customers can experience substantial productivity gains, as a recent Webtorials “State-of-the-Market” report indicated that Unified Communications can help SMBs regain hours of lost employee productivity each week. Fonality Heads Up Display is an award-winning UC platform that connects phones, desktops and important business applications into a single, user-friendly interface.

The “Turn it In/Turn it Up” program is designed to valuate trade-in inventory according to brand, model, quantity and quality. To qualify for the program, all equipment must be in working order, without damage beyond normal wear and tear. Legacy solutions from Cisco, Avaya, Shoretel, Mitel, Nortel, NEC, Panasonic, Fujitsu and Siemens are eligible and additional competitive displacement offers may be available.

In partnership with a leading buyer for used telephony equipment, the seller will receive account credit to apply to a new Fonality solution within 30 days of the buyer’s point of receipt. The “Turn it In/Turn it Up” program also facilitates environmentally responsible recycling, refurbishment or disposal of used equipment.

Tone's ReliaTel VoIP Management Solution Rated Compliant With Avaya Aura Communication Manager 6.0

VoIP Monitor — Mon, 23 May 2011 19:26:45 GMT

Tone Software announced that its ReliaTel VoIP management solution is compliant with key IP telephony solutions from Avaya.

The ReliaTel solution provides comprehensive VoIP quality and service level management for converged environments, including real time VoIP QoS and QoE analysis and trunk performance reporting for Avaya Communications Manager through the SAT interface. The ReliaTel RTCP, SNMP and SAT functionality is now compliance-tested by Avaya for compatibility with Avaya Aura Communication Manager 6.0.

Tone's ReliaTel manages both VoIP call quality and the underlying converged network infrastructure in a holistic manner. The management solution provides network and IT support teams with a unified management portal to rapidly and accurately pinpoint issues affecting network quality performance occurring at every segment across the converged environment.

Tone Software is a member of the Avaya DevConnect program -- an initiative to develop, market and sell innovative third-party products that interoperate with Avaya technology and extend the value of a company's investment in its network.

As a Gold member of the program, Tone Software is eligible to submit products for compatibility testing by the Avaya Solution Interoperability and Test Lab. There, a team of Avaya engineers develops a comprehensive test plan for each application to verify whether it is Avaya compliant. Doing so ensures businesses can confidently add best-in-class capabilities to their network without having to replace their existing infrastructure -- speeding deployment of new applications and reducing both network complexity and implementation costs.

Axia Technology Partners Raises the Bar for Telephony Solutions Providers

VoIP Monitor — Tue, 17 May 2011 17:31:20 GMT

Axia Technology Partners announces its implementation and vendor partnership with Sonus Networks and the NBS 5200 platform, giving this leading national provider of converged IP solutions a more scalable and robust solution to improve growth and operating efficiencies for their national client base.

Axia TP is a leader in the technology services realm and has established peering access to 29 of the nation’s largest carriers. The Axia converged IP solution offerings include nation-wide VoIP services, Enterprise and Call Center PBX solutions, Wide Area Network connectivity, and Internet Access. Providing state of the art infrastructure exemplifies the company’s commitment to quality customer service and consistent care. For Axia, having upgraded infrastructure is a vital part of providing next generation services.

The Sonus VoIP switching architecture expands Axia’s carrier class services with the ability to integrate higher levels of security and encryption, network call control, quality of service, advanced media services and high volume switching.

“The introduction of this advanced carrier class VoIP switching platform into our network will help us to better identify and reduce toll fraud, improve VoIP security , and lay the foundation for exponential market expansion,” said Josh Ross, Managing Partner of Axia TP.

Axia TP switches millions of local, long distance, and toll free minutes monthly and the expansion will not only provide its existing clients with top tier service, but prepare the company for the business growth over the years ahead.

Tpad Launches State-of-the-Art VoIP SIP Trunking Solution

VoIP Monitor — Fri, 13 May 2011 17:32:16 GMT

This system is able to combine data, voice and high-speed Internet connectivity over one connection, thereby providing existing and potential customers significant cost-savings while enjoying the benefits of VoIP.

SIP is a service that is deemed more cost-effective than any other telephony solutions available today. It is a streamlined next generation IP telephony solution which allows exchange of voice traffic through Internet connectivity rather than traditional physical cables or PRIs, which translates to cost savings and enhanced efficiency.

Tpad Sales Director Simon Jones explains the benefits of SIP Trunking further by saying, "What Tpad offers with SIP Trunking is not only about saving, it is also about improving workforce mobility and increased productivity as we give each of them connectivity at a fraction of a price. Employees can make phone calls and can be reached through a single number that is accessible at multiple locations so that your customers, co-workers and vendors can get prompt responses."

Tpad's SIP Trunking is offered in various packages that meets the varying needs of small and medium sized enterprises as well as large corporations. Marketing Manager Steven Johns have more to say about their SIP packages, "We aim at providing customers a complete all-in-one solution for their every business telecom need. Our SIP Trunks are offered with synchronous high-speed Internet connection that assures lightning-fast services, not only in the upload and download of data, but in routing calls as well." He goes further explaining, "In fact, our customers will be pleased to know that SIP Trunking isn't only limited to voice communication; it also encompasses sophisticated Instant messaging, toll free number accessibility, media conferencing, and many more highly advanced features."

SIP Trunking by Tpad is backed by Internet connection services offered by its partner company, Supanet, a leading Internet service provider boasting over over a million subscribers to date. Tpad and Supanet has just recently sealed the deal with this partnership to offer customers a fully-equipped, excellent-quality and higly-efficient IP network that allows maximum productivity.

Tahir Mohsan, Tpad SEO says, "Tpad has been very consistent with the development of the most advanced and cost efficient data and communications solutions so that the business sector can operate more efficiently." He concludes, "With our new SIP Trunking service, businesses large and small can save big while enjoying the most technologically advanced services that can be offered by VoIP."

Tone's ReliaTel 3.1 VoIP QoS Management Solution Released

VoIP Monitor — Tue, 10 May 2011 16:21:21 GMT

Tone Software Corporation announces the immediate availability of its ReliaTel Management Solution 3.1 Release featuring full support for virtualized environments. ReliaTel now supports virtualized deployment of the ReliaTel application to monitor and analyze VoIP quality, manage network performance, and ensure communications service levels throughout converged environments -- regardless of their technology mix.

Fueled by the proliferation of cloud computing and the aggressive pursuit of cost efficiencies, demand for virtualization support in the industry's leading hardware and software technologies continues to grow. As an organization's VoIP traffic and converged communications expand, the volume of critical data and processes to monitor and analyze also grows exponentially. ReliaTel's virtualization support enables organizations to easily expand their ReliaTel image to manage their growing VoIP and converged network environment from end to end -- from SIP at the core, to softphones at the edge -- on demand, without re-configuration efforts. Highly scalable, users can host the ReliaTel VoIP QoS and Converged Network management solution in a virtualized instance sized to meet current voice workloads -- and easily scale the solution into a larger virtual instance that supports expanded voice traffic and network loads in the future.

The benefits provided by virtualization are many, including increased data center scalability, reduced power consumption, lowered hardware overhead costs, and cost effective disaster recovery. MPS can especially benefit from ReliaTel virtualization support within their NOC environment, where server consolidation, power utilization, operational efficiency, and centralized provisioning all reduce the bottom line overhead costs that directly erode profit margins of delivering managed services to clients.

ReliaTel 3.1 also provides dramatically increased value for Avaya users by fully integrating RTCP metric analysis with Avaya CDR data, enabling voice support teams to more rapidly troubleshoot voice quality issues throughout their Avaya ecosystem. When quality degrades, ReliaTel users now have immediate access to important metrics not available in standard RTCP based QoS metrics such as number dialed, caller ID, TAC, trunk group, and circuit number. Also, additional ReliaTel Avaya CDR reporting facilities provide granular per call details, trunk group level performance, hourly, daily, call duration, and call type trends for Avaya communications environments.

D150 Voice Transcoding Series Launched by Sangoma

VoIP Monitor — Thu, 05 May 2011 18:57:41 GMT

Sangoma launches the D150 voice transcoding series, the latest addition to its transcoding board offering, targeted for the embedded and stand-alone VoIP solutions markets.

In a constant battle to maximize capital investment and improve ROI, network operators need to push as much voice traffic through their existing infrastructure as possible. Operators may choose to encode (or compress) the voice signals with any one of a variety of VoIP codecs, such as G.723 or G.729. Moreover, if a call needs to traverse two different networks that each support different codecs, the voice signal must be transcoded in real time. The processes of encoding and transcoding are processor intense and can often cause load related issues with the server that is managing the process. The D150 boards are specifically engineered to perform the required transcoding without impacting the host performance, allowing the system to support a significantly increased number of calls.

The D150 series supports a wide range of industry standard codecs and is offered in 3 form factors for greater deployment possibilities and flexibility.

The D150-ETH board provides the ability to add transcoding capabilities for compact form factors where no PCI interfaces are available or for when the CPU does not have enough power to handle extra loads
The D150-BOX appliance provides the ability to easily set-up stand-alone transcoding media servers within a very small footprint
The D150-PMC board allows the addition of voice transcoding to be embedded in custom hardware designs using the PMC IEEE 1386 standard

The D150 board makes it possible to convert numerous simultaneous channels of voice communication from one type of codec (e.g. G.711) to another (e.g. G.729), without affecting latency or using up precious host CPU resources. Each D150 product can also run up to 400 channels of any-to-any voice codec conversion with unmatched quality.

The D150 software drivers also provide "plug-and-play" capabilities for both Asterisk (News - Alert) and FreeSWITCH – two leading open source telephony projects. With the compatible drivers, the open source telephony platforms can use the D150 boards as seamless voice transcoding resources. This, company officials have said, means that existing Asterisk and FreeSWITCH applications can readily start leveraging the D150 capabilities. Further, the open source telephony software can be used as a gateway or session border controller to provide network-based transcoding services.

snom Teams with Iristel to Deliver Hosted VoIP Solutions

VoIP Monitor — Thu, 28 Apr 2011 15:52:59 GMT

Iristel and snom announce the snom 3xx and snom 8xx series desktop phones, as well as the snom m9 DECT phone and MeetingPoint conference phone, have passed rigorous interoperability tests, qualifying snom endpoints as preferred devices for operation within the Iristel network.

The combination of snom endpoints, network services and infrastructure products and Iristel’s Hosted PBX (private branch exchange) Solution offers Iristel customers a fully integrated IP telephony solution with HD audio capabilities and VoIP functionality for both large and small business environments, and residential scenarios.

Iristel’s Hosted PBX Solution enables subscribers to customize their PBX according to their specific needs and offers enhanced features, like call forwarding and call transfer, that can be seamlessly managed through an intuitive user interface. By ensuring its network is compatible with key IP telephony solutions, Iristel delivers a standards-based platform that enables customers – especially SMBs – to economically and efficiently migrate to new technologies without immediately abandoning investments made in their existing IP infrastructure.

snom’s suite of VoIP phones includes advanced IP phones such as the snom 3xx series, full-color touchscreen desktop phones such as the snom 870, wireless DECT phones such as the m9 and related endpoints, such as the MeetingPoint conference phone. All built with open SIP firmware allowing for simple installation, industry-wide interoperability and crystal-clear sound quality. As unified communications systems have proliferated, the advanced technology and open standards in every snom product has proven to be an ideal combination for providing enterprises with cost-effective and feature-rich IP telephony.

3CX and OpenIP to Provide Low-Cost, Quality VoIP Solutions

VoIP Monitor — Thu, 21 Apr 2011 15:48:26 GMT

3CX announces that all customers will be able to benefit from quality, low cost, business calls after the interoperability testing with OpenIP had proven successful. This interoperability between 3CX and Paris based VoIP service provider, OpenIP, allows the customers who use the 3CX VoIP PBX for Windows to benefit from the OpenIP high quality internet telephony services for voice calls as an integrated solution.

3CX and VoIP service provider, OpenIP, will provide fully integrated VoIP solutions for SMB’s in Paris, France. The OpenIP SIP Trunk configuration has been fully integrated into the latest version of 3CX’s software, allowing the company to become a 3CX Supported SIP Trunk provider. As a multitude of businesses throughout France already take advantage of the 3CX VoIP PBX for Windows, they now have the added value of OpenIP being onside to provide high quality internet telephony services for voice calls as an integrated solution.

Sipera Systems UC Security Solution Undergoes Compatibility Testing With McAfee ePolicy Orchestrator Security Management Platform

VoIP Monitor — Tue, 19 Apr 2011 21:40:56 GMT

Sipera Systems announced that its award-winning UC-Sec enterprise communications security solution is undergoing compatibility testing with McAfee ePolicy Orchestrator security management platform.

When the testing is completed, security managers can gain visibility into the entire spectrum of IP-based business communications -- including voice, video, data and web applications -- from a single central management point. This enables security managers to have unprecedented control over the security of voice-over-IP, IP video conferencing, SIP trunks, collaboration applications and other UC applications.

The Sipera UC-Sec appliance is a plug-and-play network device that provides comprehensive security for real-time UC applications. As a complement to the existing security architecture, the UC-Sec provides application-layer firewalling, intrusion prevention, threat mitigation, access control and policy enforcement in real-time for delay sensitive applications.

The UC-Sec encrypts UC-traffic, terminates SIP trunks, forks media and signaling for compliance, and permits an enterprise to safely and securely extend VoIP and UC to any end point in any location. Sipera's customers use the appliance to conduct secure VoIP on smartphones, in remote offices, to distributed call center representatives and to enforce security policies on all UC traffic in real-time.

The Sipera security system is undergoing compatibility testing as part of its participation in the McAfee® Security Innovation Alliance (SIA) program, which helps accelerate the development of interoperable security products and simplifies the integration of these products within complex customer environments.

Sipera's groundbreaking "Borderless UC" architecture enables all communications to be encrypted and compliant on any UC device, at any internal and external location.

cAfee ePolicy Orchestrator platform is the first to let enterprises and governments centrally manage security and compliance products from multiple vendors, offering unprecedented cost savings and return on investment. With more than 40,000 customers and managing more than 60 million PCs and servers, this unique platform is helping McAfee SIA partners to extend their reach and create complementary functionality.

Counterpath Strengthens Enterprise Feature Suite with Latest Update for Bria

VoIP Monitor — Thu, 07 Apr 2011 01:19:43 GMT

CounterPath announces a new release of its Bria multimedia softphone will be available later this month. The Bria 3.2 update for Mac and Windows adds support for multiple Instant Message and Presence accounts and the introduction of a Ribbon for Microsoft Outlook 2010, among other features.

Bria is a highly secure, standards-based, multi-platform softphone that enables voice and high-definition video calls, making it ideal for enterprises, government agencies and other organizations that want to replace their desk phones or add unified communications functionality to their existing IP phone platform.

Bria integrates seamlessly with a wide variety of enterprise and carrier infrastructure equipment from major vendors, including Alcatel-Lucent, Avaya, BroadSoft, Cisco Systems, Genband, Metaswitch Networks, NEC and Nokia Siemens Networks, enabling fast, cost-effective implementations. Bria also supports Asterisk-based telephony systems.

The array of capabilities in Bria 3.2 includes:

Multiple Account Integration. Bria users can pull in and communicate with contacts from different sources and accounts, including local and company directories, Microsoft Outlook, XMPP, XCAP and WebDav servers. Contacts can be merged into a single view with all of their information from different sources in one place.
Enhanced Contact Management and Display. The latest updates enable Bria users to call or IM a contact with a single click, as well as expand or collapse their list to show more or less information about each contact.
Ribbon for Microsoft Outlook®. Bria 3.2 includes a Ribbon for Microsoft Outlook 2010, allowing users to place calls directly from their email accounts. The new integration features also include a contextual display of contacts in Outlook's To-Do bar, providing quick access to Bria communications options.
Company Chat Rooms. Bria now enables organizations to create their own chat rooms, providing employees and authorized external users, such as business partners, with a convenient new way to communicate, connect and collaborate.
Improved User Interface. Bria 3.2 puts volume/mute controls, frequently used features and key information, such as status, all in a toolbar for convenient access.
New Workgroup Management Options. Bria users now can manage workgroup functions and add or remove contacts to a workgroup directly from the contact list.

Current Bria 3.0 and 3.1 users will be automatically updated to Bria 3.2. Pricing for a single copy of Bria is $49.95, with additional volume pricing available.

4PSA's VoipNow Brings Unified Communications to YMCA Cincinnati

VoIP Monitor — Mon, 28 Mar 2011 17:38:50 GMT

4PSA announces the official sponsorship of the YMCA of Greater Cincinnati, one of the area's largest non-profit organizations focused on youth development, healthy living, and social responsibility.

4PSA's mission is to deliver solutions that help people communicate easily and collaborate on a whole new level. The company is committed to supporting all non-profit organizations that strive to make this world a better place. 4PSA provides free licensing for VoipNow Professional, its award-winning Unified Communications solution, to charities, non-profit, non-governmental, and non-commercial organizations, Open Source projects and communities, and last, but not least, to academic institutions for classroom learning purposes.

The YMCA of Greater Cincinnati chose 4PSA's VoipNow Professional after evaluating several different Unified Communications solutions. "We have looked at several different products that would offer Unified Communications and a non-proprietary SIP compliant PBX; 4PSA's VoipNow is the product with all the features required by our organization and the intuitive web management makes it easy to manage," stated IT Director of YMCA of Greater Cincinnati Hari Ambati.

4PSA's VoipNow Professional, the winner of the 2010 INTERNET TELEPHONY Product of the Year Award is a leading solution designed to deliver Unified Communications. Organizations interested in applying for a free Non-profit license can find more details here.

ASC Implements Communications Recording Solution at Harborough Lifeline Center in UK

VoIP Monitor — Mon, 28 Mar 2011 17:35:26 GMT

ASC announces the successful implementation of its VoIP communications recording solution, EVOip, supplemented with WEBplay and POWERplay, to ensure the safety of older and disabled citizens residing in the District of Harborough in the United Kingdom.

The solution was installed at the Harborough Lifeline Center, a contact center providing 24/7, 365-day service, integrated with community emergency organizations including police, ambulances and firemen. It allows immediate replay of the most recent call to assist operators in complex situations as well as remote access from any web browser to let managers working from home listen to a critical call and determine the best course of action.

The communications recording system is integrated with the contact center’s existing infrastructure, a Mitel 3300 v. 6.1 telephone system and Mitel 5220 IP handsets. The communications recorder handles both digital and analog lines and uses station-side passive recording of VoIP, tapping the LAN connection at each IP phone.

Harborough Lifeline Center offers its customers alarm equipment and a pendant with a button they can push in case of an emergency. A speech unit can pick up their voice from anywhere in the house even if doors are closed. Calls are answered in less than one minute, and nearby responders, known as “key holders,” are designated in advance and can provide support until help arrives.

Please read the entire case study at: http://www.asctelecom.com/brochures/en/CS_Harborough_Lifeline_UK.pdf

VoX Communications Develops Video Voice Mail for the Ojo Vision Digital Video Phone

VoIP Monitor — Wed, 23 Mar 2011 17:10:39 GMT

VoX Communications has developed video voice mail, adding a unique and enhanced feature to the Ojo Vision phone. A video message cannot normally be sent to the mailbox of customers that use the Ojo Vision phone without this development effort.

Pervasip's Chief Information Officer, Mark Richards, noted, "We continue to use our expertise in cloud-based telephony and VoIP to bring exciting and innovative features to the devices that run on our network. Our customers, including the customers sold to by our new marketing partner, Globalpreneurs, can now send and receive a video voice-mail message on the Ojo Vision phone. We expect that the extra features that we provide to our customers will give an advantage to organizations that market our product. Now grandchildren can leave their grandparents a special video message, or people can leave friends and family a surprise video message."

In addition to the video voicemail and true-to-life video calling experience, the plan includes a full suite of traditional calling features like caller ID, call waiting and 3-way calling. Innovative features like 3-way video conferencing, connecting to a large screen LCD or TV and unlimited video calling are also included.

Ojo Vision Digital Video Phone features include:

High Quality Video over a seven-inch, high-resolution LCD digital screen
Simple plug-and-play installation
Also works as a digital photo frame
Display to external TV or PC Monitor for enhanced calling experience
Enhanced contact list with snapshot pictures

Talari’s WAN Virtualization Solution Now Rated ''Avaya Compliant''

VoIP Monitor — Tue, 22 Mar 2011 17:37:35 GMT

Talari Networks announces that its appliance-based WAN Virtualization solutions are compliant with key IP telephony solutions from Avaya. Talari’s Adaptive Private Networking technology for WAN Virtualization enables enterprises to combine two or more sources of inexpensive and abundant public network connectivity to create a network with equivalent or better performance than expensive private WAN connectivity such as MPLS. The Talari APN appliances, running APN Software 2.1, now are compliance-tested by Avaya for compatibility with:

Avaya Aura Communication Manager - 6.0
Avaya Aura Session Manager - 6.0
Avaya Aura Communication Manager Messaging - 6.0
Avaya Modular Messaging - Messaging Application Server - 5.2
Avaya 9600 Series IP Telephones (H.323 3.1.1) and (SIP 2.6.4)
Avaya Modular Messaging Server (5.0).

One of the companies benefiting from the interoperability of Talari Networks and Avaya solutions is Sno-Isle, a group of public libraries serving communities in the North Puget Sound, Wash. region.

Talari Networks is a member of the Avaya DevConnect program—an initiative to develop, market and sell innovative third-party products that interoperate with Avaya technology and extend the value of a company’s investment in its network.

As a Gold member of the program, Talari Networks is eligible to submit products for compatibility testing by the Avaya Solution Interoperability and Test Lab. There, a team of Avaya engineers develops a comprehensive test plan for each application to verify whether it is Avaya compliant. Doing so ensures businesses can confidently add best-in-class capabilities to their network without having to replace their existing infrastructure—speeding deployment of new applications and reducing both network complexity and implementation costs.

Fonality Releases trixbox Pro 5.2 IP-PBX Solution

VoIP Monitor — Wed, 16 Mar 2011 15:07:28 GMT

Fonality announces the general availability of Fonality trixbox Pro 5.2, a hybrid-hosted IP-PBX software solution designed for small and mid-size businesses. A hybrid-hosted solution combines the reliability of a premise-based VoIP system with the flexibility and costs savings of a hosted-cloud model. Users will experience increased productivity and lower total cost of ownership while leveraging advanced Unified Communications, contact center and VoIP calling features.

Fonality trixbox Pro 5.2, available only through certified Fonality resellers, is an open standard, software-based IP-PBX solution that can be delivered with most commercial servers and SIP phones. By leveraging a cloud-based, hybrid-hosted model for remote management capabilities, Fonality trixbox Pro 5.2 simplifies the ability to add new features and users.

Fonality trixbox Pro 5.2 delivers access to Fonality Heads Up Display, the award-winning UC dashboard that includes features like click-to-call, visual call parking, secure chat and real-time employee presence. A recent Webtorials“State-of-the-Market”report indicated that UC can help SMBs regain hours of lost employee productivity each day. The capabilities of Fonality trixbox Pro 5.2 combine to offer SMBs Fortune 500-caliber communications capabilities at a total cost of ownership typically 40 to 60 percent less than legacy IP-PBX providers. Advantages of Fonality trixbox Pro 5.2 include:

User-Based System: A user’s physical location is irrelevant to the ability to have multiple numbers associated with a single user license and access Fonality’s “FindMe” features
Multi-Site Unified Communication Features: Multiple Fonality HUD servers can be connected company-wide, regardless of location, to enable presence, secure chat, virtual conferencing and a unified contact center experience
Advanced Call Processing: By leveraging multi-core and multi-threaded processing, users can experience even greater platform stability and a more responsive, faster operating system
Proactive Monitoring: Fonality provides a diagnostic view of communications systems and allows administrators to analyze any device for recent calls to obtain metrics such as latency, jitter and packet loss, ensuring enterprise-grade voice quality of service

ViBE IP Telephony Removes Obstacles to Widespread VoIP Adoption

VoIP Monitor — Tue, 15 Mar 2011 18:00:08 GMT

Powernet has launched ViBE, a ‘game-changing’ IP telephony solution, which removes critical Quality of Service obstacles that prevent widespread VoIP adoption. ViBE uses innovative bandwidth optimisation technology that enables at least 60 concurrent business quality calls over an average UK ADSL connection, with minimal impact on data traffic. Powernet is actively looking for resellers for the product.

This breakthrough technology removes the QoS problems associated with VoIP implementations and slashes connectivity costs to offer businesses a scalable and affordable IP solution. It also brings within reach the possibility of a wholly VoIP-powered call centre on a single ADSL connection for the first time.

Traditionally, competition between voice and data traffic has made it difficult to guarantee business-quality calls over an ADSL broadband connection. ViBE overcomes this problem by giving priority to voice packets over any kind of connection (ADSL, Ethernet, Satellite or even 3G) whilst maintaining optimum voice quality and supporting multiple concurrent calls without any compromise to data transfer rates.

This innovative technology offers three key benefits to business users:

Quality of Service – With traditional VoIP, it’s not uncommon to see jitter and latency approaching 100ms, causing delays and interruptions even with only one call. By contrast, ViBE sees the number reduced to around the 5ms mark, becoming effectively zero in good conditions.
Redundancy - ViBE can bond multiple lines together. The redundancy guarantees that if one line fails then the call may still continue uninterrupted using the other connection, almost entirely eliminates the possibility of dropped calls.
Analytics - ViBE monitors your connection without any interference or impact on traffic, giving you up-to-date information regarding performance and uptime.

Telcentris Augments White Label VoIP Portfolio with Two New Solutions to Address Customer Demand

VoIP Monitor — Mon, 14 Mar 2011 17:38:17 GMT

Telcentris announces the commercial availability of WL2 and WL3, two new offerings that expand the company’s white label portfolio from a single, entrance-level package to a three-solution suite. Telcentris’ white label portfolio leverages the company’s end-to-end managed wholesale VoIP enablement platform, and now, with the addition of these new solutions, provides customers unprecedented control over their private-labeled telecommunications offerings. Addressing a number of customer requests, the full suite caters to a wide variety of audiences, ranging from resellers, MLMs and entrepreneurs interested in entering the VoIP market; to interconnects, MSOs, MSPs, ITSPs and traditional phone companies (ILECs, CLECs, IXCs) seeking to add VoIP to their business services. In addition, Telcentris is providing its white label solutions through its Channel Partner Program, making the company among the first to offer a comprehensive white label VoIP portfolio to channel partners.

Telcentris now boasts three white label packages that enable customers to private-brand SIP trunking, hosted IP-PBX solutions, carrier services, SMS, callback services and residential landline replacement at wholesale prices. All three options can be set up in a matter of weeks. The solutions are as follows:

WL – Telcentris’ original turnkey, managed VoIP enablement solution, launched in December 2010. This offering, which includes a number of pre-packaged VoIP products and services ready for private labeling in a matter of days, is best suited for resellers and entrepreneurs looking to enter the VoIP market with minimal upfront investment. Package highlights include:

Carrier-grade billing platform
Administrator portal
Fully branded members portal
Customized invoice template
Simple pricing model
Transaction detail reporting
Ability to assign purchased numbers
Ability to create customers and accounts
Ability to modify offerings, tariffs, subscriptions and plans

WL2 – This new offering mirrors the original white label solution model but also grants the customer API access and more control over functionality, interfaces, troubleshooting and product pricing. This solution is designed for more technical audiences such as interconnects, MSPs, ITSPs and carriers, but also may be utilized by resellers and entrepreneurs with access to technical resources. WL2 offers all of the original white label solution’s features and services, and also adds highly sought-after features, such as:

Access to all of the original WL offering’s features and services
API accessibility
User access level management
Personal DID inventory
Troubleshooting tools (ability to access SIP logs, billing logs and call tracing)
Cost / revenue reporting
Ability to create customer plans, tariffs, and representatives

WL3 – This new solution in the suite takes white labeling to the next level of customization and scalability. Akin to a hosted or partitioned softswitch, this offering grants full access to Telcentris’ underlying technology platform, enabling customers to add direct carrier interconnections, leverage oversubscription, create products and resellers, and access carrier-grade Least Cost Routing. This solution is geared towards more technical audiences, including interconnects, ITSPs, MSOs, ILECs, CLECs, IXCs and other international telcos, who may have an interest in adding VoIP services to their portfolio but have existing carrier relationships and/or advanced customization needs. WL3 capabilities include access to all abovementioned features plus:

Oversubscription (concurrent call sessions)
Direct carrier interconnections
LCR platform
Vendor usage reporting
Ability to create resellers and sub-customers
Ability to create products and vendor tariffs
And more

All white label customers receive web-based interactive training sessions spanning three to five days.

4PSA Introduces Flexible Unified Communications with VoipNow Cloud Instance

VoIP Monitor — Thu, 10 Mar 2011 21:54:23 GMT

4PSA announces the release of VoipNow Cloud Instance, 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 Instance is intended for service providers that choose to deploy Unified Communications and want to base their offering on services highly flexible from a financial and operational perspective. Compared to other UC cloud offerings on the market, this solution comes with many advantages due to the tight integration between hardware and software resources. From a customer's perspective, the main benefit is that the instance is private; it is not shared with any other entity. This means that the service provider has full access to all system parameters, down to the low-level settings.

VoipNow Cloud Instance comes at a very low cost due to the built-in technologies that scale the cloud instance resources dynamically with the number of customers provisioned by the service provider. "Another major advantage of VoipNow Cloud Instance is the scaling path. A cloud instance can grow from 0 to 10,000 Unified Communications customers without any downtime, which is very important for service providers. They are in charge of their instance and can choose the voice carriers they prefer. Furthermore, they have full control over the system through a range of web-based management tools", added Mr. Carstoiu.

VoipNow Cloud Instance services are available both in US and Europe. The service is provisioned in a couple of minutes and comes with all the necessary tools to provide Unified Communications. Customers can choose between two functionality levels: both with full Unified Communications feature set and, when selected, with VoipNow Automation. Automation allows service providers to reduce costs by streamlining provisioning, invoicing, and other important business operations.

Pricing for VoipNow Cloud Instance service starts at US $159 per month and includes support for up to 100 UC customers. Each additional customer is charged with a maximum of US $2.1/month. The service is backed up by a 99.95% uptime guarantee.

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

Virtual PBX Announces No-compromise, End-to-end Hosted Phone System for Small and Mid-size Businesses

VoIP Monitor — Tue, 22 Feb 2011 19:26:55 GMT

For many small and mid-size businesses, deciding how to provide an affordable phone service that suits employee needs and allows for great customer service can be a struggle. Do I go wireline or wireless? Can I depend on VoIP? Is a hosted model right for me? How do I get all the best features, and how can my small business look like a big business to my customers?

Virtual PBX announces the perfect solution for this classic dilemma – Virtual PBX Complete. Virtual PBX Complete has already been recognized for its outstanding innovation by Internet Telephony Magazine.

Virtual PBX Complete delivers everything a business needs to provide professional phone support for customers, prospects and employees while eliminating the need to buy dial-tone service and other expensive options from traditional service providers. Businesses need only an Internet connection, and Virtual PBX Complete will do the rest. It can be used as an inbound call router to existing phones, or users can select a complete phone system with VoIP lines and phones. Pricing plans for the complete service start as low as $9.99/month.

Unlike other hosted IP-PBX solutions, Virtual PBX Complete does not require VoIP. For example, with Virtual PBX Complete, a small startup can set up the service so they can run their entire business using only cell phones. Others could implement a Virtual PBX Complete solution using VoIP desk phones with direct extension dialing and all the other features associated with a traditional working environment. Some companies will want to combine the use of VoIP phones where appropriate and cellular or land lines where needed. Virtual PBX Complete offers the flexibility to work where and how the client needs.

Virtual PBX has also dramatically reduced the difficulty of installing a new phone system. Virtual PBX Complete VoIP phones are true “plug-and-play” devices. Users simply connect the phones to their broadband Internet connection and the devices register and connect automatically with no user intervention.

DETAILS ON VIRTUAL PBX COMPLETE

Virtual PBX Complete includes access to all of the features developed by the company over its 15-year history. Pricing plans set a new bar for value, with unlimited packages available for just $19.99 per user per month and metered plans that start at under $10 per month. In addition, Virtual PBX Complete is built around the company’s award-winning Open VoIP Peering concept, allowing clients to use VoIP hard phones and soft phones registered to Virtual PBX or other providers. Unlimited minute plans are even available to users who have no VoIP phones at all – a breakthrough concept as other IP-PBX providers require a VoIP phone on every extension.

Key features of Virtual PBX Complete include:

Dial-tone service
Optional Polycom VoIP phones
Auto-attendant with professional greetings and greetings library
Customizable greetings and hold music for different departments
Complete voicemail with retrieval online, by phone or by e-mail
Inbound and outbound faxing with no fax machine
Automatic or manual call recording with optional admin and/or user control
Free integrated conferencing
Automatic call distribution queues
Skills-based routing
Departmental load balancing
Multiple-role administration and security
Follow-me/find-me capabilities
Call preview and smart caller ID to know who is calling and what they want
Real-time voicemail monitor and interrupt
Local, toll-free and international numbers
Multi-business support
Customizable music on hold
Easy web management and real-time monitoring

MiaRec Announces the Release of Call Recording in HD Quality

VoIP Monitor — Tue, 22 Feb 2011 18:09:25 GMT

MiaRec announces the release of HD call recording. Award-winning call recording and monitoring solution – MiaRec - supports now G.722 wideband codec. This assures better compatibility with Polycom’s high definition IP phones, Cisco, Avaya, Grandstream and other IP phones, which support wideband codecs.

There are a lot of benefits of VoIP telephony, the most obvious one is cost. But does low cost always mean low quality? With G.722 codec the answer is absolutely not. HD VoIP telephony is a completely new level not only of VoIP telephony but of the world of communication in general. G.722 codec now enables VoIP conversations to be even clearer than traditional digital line/PSTN calls.

Conference calls are a direct beneficiary of the enhancements offered by wideband audio. Participants often struggle to figure out who is talking, or to understand speakers with accent. Misunderstandings are commonplace due primarily to generally poor audio quality and an accumulation of background noise. In addition, the ubiquity of conference calls combined with the level of reverberation over speaker phones creates another set of audio problems. Wideband G. 722 codec provides better, more lifelike voice communications and markedly improved intelligibility because of the additional voice data included in the audio stream.

HD VoIP telephony provides all benefits of VoIP telephony, including convenience and price, but also higher quality of voice communication. A new version of MiaRec Call Recording was designed to allow businesses to get the quality enhancements of high definition VoIP telephony.

By successfully achieving interoperability with the G.722 codec MiaRec Call Recording and Monitoring Solution is able to provide high definition call recording with the new audio standard for VoIP PBX communication. This means interoperability with a range of Next Generation IP phones supporting G.722 codec. Among them are the new Cisco Unified Communications Manager, the next generation of Cisco phones, and the new Polycom and Avaya phones.

At the same time MiaRec HD call recording enables a superior audio quality without requiring any additional storage space. All HD call recordings are archived and compressed in MP3 with optimal bitrate settings for HD audio.