VoIP Monitor - VoIP Software

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.

Ingate Debuts Full Lineup of Software SIParator and Firewall E-SBCs

VoIP Monitor — Tue, 05 Jun 2012 21:47:40 GMT

Ingate announces the Ingate Software SIParator and Ingate Software Firewall, new software-only versions of the company's Ingate SIParator and Ingate Firewall E-SBCs.

The Ingate Software SIParator/Firewall offers the same security and SIP-enabling functionality found in Ingate's hardware-based E-SBCs. The software can be installed on customers' own servers (or integrated with IP-PBXs or media gateways) or used as a virtualized application and all of Ingate's usual advanced modules can be added as required.

Available now, the Ingate Software SIParator/Firewall comes in a wide range of models to address the needs of small businesses and SMBs, large enterprises and everything in-between. The software-only E-SBC can handle from as few as five simultaneous calls, up to as many as 10,000, depending on the hardware used.

The software-only E-SBC is intended for IP-PBX vendors, system integrators and customers deploying a large number of Ingate products on their own hardware platform.

Enabling Secure SIP Trunking, Unified Communications
Like all Ingate E-SBCs, the Ingate Software SIParator/Firewall enables secure SIP into the network to make trusted SIP trunking and UC possible. It works hand-in-hand with an existing network firewall to allow SIP traffic to traverse the enterprise edge. It can also be configured with firewalling functionality to provide enterprise security for all SIP and data traffic.

Advanced Security Bundled Free with Ingate Products
Intrusion Detection System/Intrusion Prevention System solutions for SIP are bundled free with all Ingate models, including the software version of the Ingate E-SBC. IDS/IPS protects against attacks targeting SIP devices, such as IP-PBXs and SIP phones. IDS/IPS works in tandem with Ingate's existing security features to offer maximum protection.

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.

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.

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.

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.

Twilio Launches VoIP Developer Toolkit for Millions of Web & Mobile Applications

VoIP Monitor — Tue, 26 Jul 2011 20:27:52 GMT

Twilio announces the release of Twilio Client, a software toolkit for web, iOS, and Android development. With Twilio Client, mobile and web developers can now enable voice communications within any application by programmatically embedding two-way audio between their users on websites, inside web and mobile applications, and with traditional landline and office phones.

With Twilio Client, millions of web developers worldwide can now build Skype-like voice capabilities with Twilio’s scalable, reliable communications infrastructure-as-a-service. With just a few lines of code, web and mobile applications can host voice conversations, conference calls, and other forms of rich communication. Twilio Client works simultaneously across platforms, allowing web browsers, mobile phones, and tablets running iOS or Android to communicate seamlessly. The service also integrates with traditional phone service and SMS using Twilio’s existing web service APIs for making and receiving phone calls and text messages.

Twilio Client enables communication experiences in a wide range of existing applications, including email, CRM, gaming, entertainment, news, chat, and social networking. Using Twilio’s simple developer tools, businesses can build sophisticated unified communications solutions such as call centers, office phone systems, call tracking tools, and more.

Over 40,000 developers use Twilio for applications that interact with traditional mobile and landline telephones via the public telephone network. Twilio Client now allows those applications to interact directly with end-users, bypassing the public telephone network and completely reinventing the way developers enable communication, without the constraints of traditional telephones.

Twilio Client is available today as a free Javascript SDK download, with iOS and Android SDKs currently in beta. Twilio Client connections cost ¼ cent per minute.

Citel Announces Portico Telephone VoIP Adapter 5.00.0 Availability

VoIP Monitor — Tue, 21 Jun 2011 22:40:33 GMT

Citel announced the immediate availability of the Portico TVA version 5.00.0, enabling businesses to deploy the award-winning Portico Telephone VoIP Adapter with the latest features and updates. Citel announced that the latest release has been successfully Beta tested and is an updated version of the previous release 4.01.0 – “Citel’s most robust and comprehensive set of new features for the Portico TVA to date.” The availability of the Portico TVA is obtainable on all newly manufactured Portico TVA units, with current systems being fully upgradeable in the field. Previous deployments of the TVA have proved beneficial in easing the transition to VoIP migration for companies such as the Florida Department of Health and the availability of version 5.00.0 will further drive the success ratio of Portico deployments when used to aid in the migration towards a VoIP environment.

Expansion of existing product features.

Many enterprises have already benefited from the great features and functions of the Citel Portico TVA. Duo County Telephone turned to the Portico TVA to facilitate the migration of its consumer base to VoIP with minimal disruption. Since the system permit the re-use of existing phone lines and power, no re-cabling was required.

Version 5.00.0 for the Portico Telephone VoIP Adapter enhances resiliency with redundant SIP soft-switches, using SIP Forum’s SIPconnect standard, while enhancing support for P-Phone Centrex phones with add-on units. Additionally, it provides heightened support of toggle speed-dials and works to simplify the SIP configuration process with an optional ‘Realm’ field.

Portico version 5.00.0 Availability

Version 5.00.0 is a collection of updates driven by customer feedback as part of ongoing Beta testing throughout 2010/2011 and is part of Citel’s commitment to deliver the latest product updates and remain a frontrunner in the field of VoIP migration. All newly manufactured Portico TVA units will be pre-loaded with the most recent software release, and existing Portico units can easily be upgraded in the field. The latest release is available to all TVA users with products under warranty or with extended maintenance contract by contacting Citel Support via www.citel.com. For those customers whose units are no longer under warranty can contact Citel Support for details on purchasing the firmware upgrade. Full details on the new firmware can be found at www.citel.com/Products/Resources/PIBs.asp.

AnVoice VoIP Engine, Enhanced Voice Quality for Android VoIP Applications

VoIP Monitor — Thu, 16 Jun 2011 18:05:48 GMT

Adaptive Digital releases the newest version of AnVoice software, a VoIP Engine for Android. This version includes enhanced voice quality features such as G.722 (wideband audio) with packet loss concealment and dynamic jitter buffer. AnVoice works in a wide variety of Android-based handsets, such as DROID by Motorola, DROID 2 by Motorola, DROID X by Motorola, Nexus One, and Samsung Galaxy in both handset and speakerphone modes.

AnVoice, Adaptive Digital’s VoIP Engine for the Android is a software package that provides voice to Android. It includes all of the voice processing necessary to VoIP-enable an Android application. The core of AnVoice is an Android native-layer application that includes a complete suite of Adaptive Digital’s field-proven telephony, VoIP, and voice quality enhancement algorithms that enable developers to create toll-quality next generation mobile applications for Android/ARM users. The VoIP Engine is supplied with a sample Java application that interfaces to the VoIP Engine native application. The sample application uses the AnVoice API, which in turn uses the Java Native Interface, to setup an RTP/IP to RTP/IP VoIP connection.

VoIP is not a run-of-the-mill Android / Java application. A VoIP on Android application needs to run at both the Java layer and also at the more cumbersome native layer of Android. Writing software at the native layer is complicated not only due to the complexity of Android but also due to the nature of open-source software in general. Additionally, since voice is a real-time phenomenon, a VoIP application requires demanding real-time performance from the cell phones central processing unit as well as from the operating system. Real-time applications currently reside on the bleeding edge of Android technology.

With the current release of AnVoice, Adaptive Digital has made great strides toward a more robust and more universal Android-based VoIP engine.

CounterPath Releases Bria for iPad Version 1.0

VoIP Monitor — Wed, 08 Jun 2011 22:15:55 GMT

Nearly one in four enterprises already use tablets, and by 2013, so will the majority of businesses, according to multiple analyst and vendor surveys of Chief Information Officers and IT managers. As this installed base grows, so does the opportunity for enterprises to leverage the tablet to minimize telephony costs while improving employee productivity and responsiveness.

To enable enterprises and mobile workers to capitalize on this trend, CounterPath announced the worldwide availability of Bria iPad Edition Version 1.0 via the Apple iTunes App Store. This standards-based, service agnostic softphone provides an intuitive user interface for making and receiving calls over a Wi-Fi or a 3G/4G mobile connection, making it the first universal endpoint for iPad.

Bria iPad Edition 1.0 supports multiple VoIP accounts and SIP protocols, Bluetooth headsets and native iOS multitasking. This enables business users and prosumers to access multiple personal and business accounts while simultaneously utilizing other applications or accessing documents on their iPad during a call.

Capable of working with any VoIP service provider, Bria iPad Edition supports hosted VoIP and IP-PBXs, effectively turning the iPad into a mobile desktop phone. Bria includes a list of pre-configured, CounterPath-approved ITSPs, making VoIP account set-up fast and easy for users who either lack a VoIP provider or already use a CounterPath-approved ITSP.

Unlike other VoIP voice applications marketed for the iPad that were introduced primarily for the iPhone, Bria iPad Edition is designed specifically for all Apple iPads. This iPad-centric foundation means that Bria iPad Edition leverages the device's large, high-resolution display, providing a richer user experience that enhances the features found in the Bria iPhone Edition.

Bria iPad Edition's separated navigation panes make it effortless to find contact information and initiate calls. A one-touch popover dialpad is elegantly minimized yet easily accessible at all times, while the iPad's native keyboard is also available for entering alphanumeric SIP URL/addresses.

Other key features of Bria iPad Edition 1.0 include:

A customizable user interface specifically designed for the iPad. A visually engaging double-pane design logically organizes application features and enables simple and smooth transitions between tabs and sub-tabs. Includes options for custom background images and custom colors.
Bluetooth connectivity. When the iPad is paired with a compatible Bluetooth headset, the Bria client can be used to speak and listen providing the ability to be away from the iPad up to the range of the headset.
iTunes Auto Pause. When a call is placed or received, the audio from the iPad's iPod music application is automatically paused.
A wide variety of deskphone features, including call hold, transfer, conferencing, display and history, as well as contacts and a voicemail indicator.
iOS4 multitasking, enabling access to other applications without interrupting calls. Bria iPad Edition softphone runs in the background and can be returned to quickly by a top-of-screen tap.
Advanced security and audio features. Secure call signaling and audio encryption plus codec support for G.711, G.722 wideband audio for lifelike sound, GSM, iLBC and an optional G.729 codec that is available for purchase within the app or through the Apple Store.

According to Infonetics Research, the VoIP services market reached nearly $50 billion in 2010. With more than 19 million iPads sold worldwide, Apple currently holds the majority of the growing tablet market. "CounterPath's Bria iPad Edition meets the needs of these rapidly converging markets, serving both the prosumer and enterprise consumer segments," Carothers said.

Like Bria iPhone Edition, Bria iPad Edition 1.0 integrates seamlessly with other CounterPath desktop and mobile VoIP solutions, as well as with enterprise and carrier infrastructure equipment from major vendors. CounterPath also develops customized white-label versions of Bria iPad Edition for carriers, OEMs and enterprise customers worldwide.

A version of Bria iPad Edition that includes support for video calls, SMS, Instant Messaging and presence will be available later this year.

Bria iPad Edition is available immediately in theApple iTunes App Store or via CounterPath's online store at www.counterpath.com/store.

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.

CounterPath Announces ITSP Partner Program for Mobile Softphones

VoIP Monitor — Wed, 25 May 2011 17:32:49 GMT

CounterPath has launched an online certification program for Internet Telephony Service Providers. The CounterPath ITSP Partner Program enables ITSPs to certify their service with CounterPath's mobile softphone, Bria iPhone Edition, and provide a pre-provisioned service option within the softphone client.

By participating in the ITSP Partner Program, ITSPs can provide their customers with a premier softphone option that is easy to activate and use. The program also helps ITSPs expand their market reach by leveraging the CounterPath softphone user base.

Registration for the CounterPath ITSP program is free via the CounterPath website. ITSPs receive an Acceptance Test Procedure to complete and, once approved, the ITSP Partner will appear in a list of pre-provisioned VoIP providers within the mobile softphone application. CounterPath Certified ITSP Partners also gain exposure on CounterPath's website and receive a CounterPath Certified logo for use in their customer collateral.

The CounterPath ITSP program is available immediately for certification with CounterPath's Bria iPhone Edition softphone. Certification with the company's Bria Android Edition softphone will be available soon, with the award-winning Bria desktop softphone to follow.

Goober Networks Delivers SDK for VoIP and Video Applications

VoIP Monitor — Mon, 16 May 2011 16:39:01 GMT

Goober Networks announces the availability of VIVO Engine, a VoIP and video software development kit for real-time communication over IP. The SDK includes everything necessary for expanding applications with capabilities such as VoIP and video and includes SIP signaling software, which is the standard for real-time IP communications, as well as NAT/FW and video rendering.

Goober's VIVO Engine is available as a white-label turnkey solution to OEMs or as separate modules, which offers customers the flexibility to adapt to their specific design needs and provide unprecedented services to empower the offerings of social networking sites such as Facebook, LinkedIn and mySpace. One such customer, ICQ, is the pioneer of Instant Messaging with over 34 million global monthly users. ICQ belonged to AOL before and was sold to DST (now Mail.ru Group) in 2010.

At the heart of the VIVO Engine are HD video and HD voice codecs. Widely recognized as the premiere market solutions in terms of HD video and HD audio quality, these codecs run on any available platform. Additionally Goober offers the carrier grade backbone (as a rentout) necessary to enable such services as video conferencing, paid VoIP calls, and free video conferencing with up to six participants. Offering unparalleled voice and video quality and voice and video optimization functions, the SDK is available for Windows, Mac, Linux, iOS and Android.

Goober's VIVO Engine SDK features a fully-integrated network, video, and voice engine as well as a f lexible, high-level API for a simple integration. Based on the variety of VoIP and video codecs utilized by Goober Networks, quality can be adjusted to specific bandwidth and include HD resolution. The VIVO Engine SDK also includes SIP as signaling standard, network optimization functions for a clean and stable IP connection, broad- and narrowband codecs for VoIP, and several video codecs.

Unified Communications Finally Unified with New 3CX Phone System v10

VoIP Monitor — Wed, 11 May 2011 16:41:21 GMT

3CX announces the availability of 3CX Phone System version 10. This new release includes a new user-friendly web-based app, 3CX MyPhone. Other PBX’s that include unified communications do not make features such as conferencing easily accessible for the end user. The new 3CX MyPhone portal makes it easier for users to take advantage of advanced unified communications and VoIP features by bringing conferencing, presence, transferring of calls and other rich features together in one easy to use application.

New Advanced User Portal – 3CX MyPhone

The new user portal – 3CX MyPhone – is web-based, but leverages Silverlight to provide a rich –“desktop like” user-friendly experience. With 3CX MyPhone, users can easily manage their extension from their netbook or desktop and avoid having to use a cryptic phone interface. 3CX MyPhone works in tandem with softphones, IP phones and even iPhone and Android smartphones, retaining flexibility for users to choose what phone device best suits them.

3CX MyPhone Allows Users to:

See status/presence of other users
Launch calls from a web page, outlook or phonebook
Transfer calls to other users.
Divert calls to voice mail
Monitor queue calls (if they have the rights to)
See missed / inbound / outbound calls
Create and manage conference calls
Configure call forwarding options when away/out of the office

More Manageability and Security Features

Version 10 has also gained enhanced management and security features. Administrators have finer control over blacklisting of clients trying to register against the server. It is possible to automatically block outbound calls during night time or when the office is closed, as well as block outbound calls to countries or continents to which a company never makes calls. 3CX server event notifications we’re expanded, and administrators can now be notified by e-mail of critical system events such as a losing connection to a remote office or SIP trunk.

Improved PBX Performance

3CX Phone System v10 is the only native SIP PBX for Windows. This focus on Windows allows us to optimize our code for the platform. 3CX Phone System v10 performance was improved, whilst significantly reducing memory and processor usage. 3CX Phone System can run virtualized or alongside other server applications – thus not requiring a dedicated server!

Improved IP Phone & Smartphone Integration

3CX Phone System is SIP standard based and integrates with a range of popular SIP phones. In 3CX v10 integration was further beefed up with seamless support for PnP deployment of IP Phones, softphones and smartphones. Administrators just plug in an IP Phone or ask the user to install the 3CX smartphone app on their smartphone (via the Appstore/Android Market). The new clients will be visible in the management console and allow the administrator to push out the settings automatically.

Improved IP Phone Management

3CX Phone System has gone further in easing the life of the network administrator. IP Phones can now be managed end to end from the 3CX Management Console. Vendor Independent! IP phone firmware versions can be checked and upgraded remotely – either one by one or in batch mode. This significantly reduces administration time and increases control for the administrators.

Improve Customer Service

3CX Phone System can now match caller IDs to customer names, and show these in 3CX MyPhone or even on the IP Phone. Employees will be able to answer the phone knowing the customer’s name and significantly increase customer service.

What’s new in 3CX Phone System V10?

All Versions

Easier to use 3CX MyPhone that incorporates 3CX Assistant
3CX MyPhone is now Silverlight based, incorporating the old assistant and is therefore much easier to deploy
3CX MyPhone now also supported on Macs
Notification to Windows Event log, 3CX Event Log and optionally by e-mail on key events (show Event log)
Ability to blacklist IPs
Ability to block extensions being registered from outside the network
You can configure that particular extensions can not be used outside of the local network
Ability to limit extensions registering externally to use tunnel (which provides additional layer of security via password)
Ability to record prompts via the phone – simply click on prompt, enter extension name and record prompt
Reduced memory footprint and elimination of extra service
Class of Service – Specify extension group in outbound rules to easily create Class of Service rules
Themes support in 3CX MyPhone / 3CX Assistant

Small Business, Pro and Enterprise Editions

Improved conferencing page in 3CX MyPhone allows for easier setup of conference calls
Queue tab in 3CX MyPhone which gives an overview of the status of queues
Improved BLF page – now allows provisioning of Speed dials, custom codes
Complete G722 support (Requires G722 capable phone)
Server will query phone book to match Caller IDS to a specific name

Use this V10 Demo Key to activate enterprise features: I8VM-OXW5-GRI6-EIV4

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.

F-ROUTE Adds Business Features to SessionTalk SIP VoIP Softphone for iPhone

VoIP Monitor — Thu, 21 Apr 2011 15:52:56 GMT

F-ROUTE recently released the latest update to their SIP VoIP Softphone for the iPhone. SessionTalk 3.1 features now include Dual Line, Call Waiting, Call conferencing as well as Transfer and Attended Transfer. This adds to an already impressive list of features including iOS4 Multitasking and Background support, Incoming call Push Notifications, Native contacts integration and a highly intuitive iPhone like user interface.

SessionTalk enables businesses as well as the mobile worker to use their iPhone as an IP-PBX extension, allowing them to access the same phone services and features as if they were in the office. The application can help boost worker productivity in a number of ways and show a fast return on investment. For example, a product development expert working from home or from a remote site would be able to join in a phone conference with managers at headquarters and collaborate on a new product design.

The highly intuitive iPhone like design will also appeal to the casual VoIP user as they are presented with an interface they are very familiar with. Also to help users who are new to VoIP there are video tutorials available on the company's website.

SessionTalk 3.1’s other key features include:

Full SIP compliance, enabling use with any SIP-compliant server and hundreds of ITSPs.
Excellent audio quality with echo cancellation.
Unlimited multiple accounts.
Voicemail waiting indicator, detailed call history, favorites and ringtones.
Support for DTMF, which lets users enter numbers to access an auto attendant.
Bluetooth support (iPhone only).
Optional customized branding available for enterprises and ITSPs.

F-ROUTE still have many improvements and additional features planned for SessionTalk including G.729 codec support, call recording and SMS support which will be available in the next few weeks. The company will also have an Android version available later this year. SessionTalk 3.1 is sold exclusively through the following link on the iTunes Store for $4.99.

VoxOx Makes PC Magazine’s ''Best Free Software of 2011'' List

VoIP Monitor — Tue, 12 Apr 2011 17:29:29 GMT

VoxOx has been named to PC Magazine’s “Best Free Software of 2011” list for the second consecutive year. This year’s list highlights the top free downloadable software applications in 35 categories. VoxOx is featured as one of five free services in the “VoIP and conferencing” category, which also includes Skype.

PC Magazine features editor Eric Griffith writes: “Every type of communication you can imagine is part of the VoxOx unified hub: phone, e-mail text chat, SMS texting, social networking, you name it—even, dare we say it, faxing. It also translates for you on the fly and takes messages. The product of a phone company, VoxOx provides you with a free phone number to use for incoming calls.”

Additionally, Departures, a luxury travel publication, has named VoxOx to its “2011’s Best Gadgets” list, a compilation of the most exciting and innovative tech products to look forward to this year. VoxOx is listed as 2011’s Best Gadget for Communication. The Departures article states:

“Instant messaging, SMS, Skype, email, Facebook: The newly updated VoxOx desktop program puts all those modes of communication into one manageable window so that staying up-to-date with friends and colleagues no longer requires checking multiple devices. In addition, VoxOx offers a bevy of other communication tools, including on-the-fly translation of e-mail. The only charge is for outbound texts, faxes and calls.”

“We’re thrilled to receive not one, but two editorial nods by such distinguished publications as Departures and PC Magazine,” said Bryan Hertz, CEO of Telcentris. “VoxOx is a one-of-a-kind service, and each time we are recognized it reaffirms our commitment to technological innovation and betterment of communications.”

These two designations mark the latest in a string of awards and accolades for VoxOx, including:

"Best Software / Service" at CES 2010 – Computer Shopper
“50 Disruptive VoIP Services (2010)” – Heavy Reading
“Best Free Software of 2010” list – PC Magazine
“Greatest Gadget Finalist” – GadgetFest 2009
“Best New Product / Service 2009” – American Business Awards, “The Stevies"
“Best Consumer Offering (2009)” – TMC’s ITEXPO West

VoxOx is a free service with software available on both PC and Mac operating systems.

Voip-Pal.Com Announces the Launch of Its New PointsPhone Platinum Suite Which Includes an Antivirus Program for North America Android Users

VoIP Monitor — Mon, 11 Apr 2011 14:43:48 GMT

Voip-Pal.Com announces the availability for North America Android users of its PointsPhone Platinum Suite package which bundles an Antivirus Program. The Antivirus Program has been designed specifically to guard smartphones against spyware, malware and viruses.

The PointsPhone Platinum Suite is now available in North America only for all Google Android smartphones, including the new high-performing Motorola ATRIX and Samsung's Galaxy 4G. The Platinum Suite will include a one year subscription to a state-of-art antivirus program, substantial domestic and international airtime minutes, enhanced Voice Mail features and a free North America phone number. The antivirus program is truly comprehensive, providing automatic real-time protection against viruses, spyware and other malware. Memory cards are automatically scanned when they are inserted to prevent transfer of malware and viruses. A GPS locating system is available to track down lost or stolen smartphones. A Remote Backup feature is available to backup all data stored on the smartphone in case the smartphone is lost or stolen.

Users will register and purchase a Platinum Suite for $29.95. A User account and password will be created and users will have immediate access to their account for airtime minutes, Voice Mail and virtual phone numbers. A registration number for their Antivirus Program will be sent as a text message to their smartphone which will provide access to download their Antivirus Program within 48 hours (to complete fraud check). A free North America Phone Number is allocated. The PointsPhone Mobile App for the Android is downloaded free from the Android Market App Store (search 'pointsphone').

For more information on PointsPhone products and services please click on link: http://www.youtube.com/watch?v=-vSnIpCsnS0

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.

F4W's Technology Improves Quality, Increases VoIP Connections via Satellite and Cellular by 500%

VoIP Monitor — Wed, 30 Mar 2011 17:07:26 GMT

F4W announces a radical change in the way VoIP performs over low bandwidth or high latency connections, such as cellular or satellite. A tremendous barrier to accessing the power and flexibility of VoIP during a disaster event or in remote locations is the inability of traditional VoIP to connect and sustain a connection over a satellite or 3G / 4G cellular data connection due to the high latency of those types of networks. Add security concerns via the implementation of third-party devices and the ability to support a full-scale operation with VoIP becomes impossible.

However, all that has changed with F4W's Core technology. This software revolutionizes the use of VoIP, allowing for 10, 20, 30 or more simultaneous voice calls with no degradation in quality over any available network connection. Furthermore, F4W's Core technology is entirely secure, adding no burden to the call or requiring no additional hardware or software to implement.

F4W's technology has been developed from years of real-world, in-the-field emergency response experience, whether with a system or side-by-side with a customer, F4W has participated in every major event in the past six years.

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

RHUB Communications Adds HD VoIP to Web Conferencing and Remote Support Appliances

VoIP Monitor — Tue, 01 Mar 2011 16:26:41 GMT

RHUB Communications announces the addition of Hi-Definition VOIP capabilities to TurboMeeting, the company’s flagship line of web conferencing and remote support appliances.

With TurboMeeting 4.3’s High-Definition VoIP softphone, users can conduct internet-based audio conferences with more than 200 participants from both PC and Mac computers. The softphone doubles the voice sampling rate from 8kHz of traditional phones to 16kHz, resulting in improved sound quality. The softphone’s capabilities include echo-cancellation, firewall traversal, automated audio device detection and auto-reconnection. Users can also record entire web conferences, including audio, screen updates and webcam video. In addition, TurboMeeting 4.3 includes a recording converter, allowing users to convert their recordings to a variety of standard formats, including FLV, WMV, and AVI.

In addition to the VoIP functions, TurboMeeting 4.3 extends LDAP integration by allowing user privileges to be defined by user groups, further facilitating TurboMeeting deployment for larger enterprises.

Pricing and Availability

TurboMeeting 4.3 is available today.

Tone Software Reveals Holistic Management Approach as Key to Leveraging Maximum VoIP Performance in Complex Converged Environments

VoIP Monitor — Tue, 01 Mar 2011 16:24:28 GMT

While networks are the backbone for VoIP communications, they are also the Achilles heel, presenting a challenging set of circumstances. The core issue is data versus voice. Data traversing today's networks are forgiving of such things as latency, delay and packet loss. Not so with voice. Voice requires clear, noiseless call quality without interruption or interference, and these same common network conditions can seriously degrade VoIP quality and performance.

Yet, more and more businesses are relying on VoIP based technologies to fuel work group collaboration and drive advanced business processes -- with the expectation of reducing costs, increasing productivity and gaining significant ROI. The promise of VoIP is very attractive -- but quickly erodes if no one can hear you. High availability and near flawless call quality to both internal and external users are requirements of businesses implementing IP telephony, however, these standards can be elusive. Poor call quality negatively impacts both employees and customers -- and is often difficult to overcome.

According to Tone Software the strategy for success is a holistic approach that enables the interdependence of voice and the network to be viewed and managed as a whole. Otherwise, diagnosing VoIP issues can be a moving target. For instance, call servers are often at the top of the suspect list when trying to get to the root of frustrating VoIP quality and service issues, but frequently the perpetrator can be found in the complex underlying network infrastructure. Capacity and network traffic levels occurring at the time of voice degradation are frequently the culprit -- but without the tools to pinpoint these conditions, support teams waste valuable time and resources chasing after the wrong quality issue -- leaving the true cause of the problem undiagnosed and unresolved.

Reflecting on what companies are doing to establish and execute successful VoIP QoS management in a converged environment, Tone sees three steps in common. First, implementing a management strategy where both network and voice teams have insight into the impact of their actions, while working together to achieve optimum VoIP communications. Essentially, end-to-end visibility and control over both the voice and network domains that comprise VoIP.

Next, those successful in leveraging maximum benefits from VoIP are utilizing deep VoIP QoS metrics to rapidly find and resolve the correct quality imparing voice issues. Such analytics are combined with network performance metrics so voice technicians can relate quality problems to other relevant network performance events, faults and issues occurring and contributing to voice quality degradation.

In an environment where the only constant is change, proactive monitoring and management of voice traffic levels, new applications, network configuration changes and the like, is the third step to achieving VoIP quality.

By taking these steps, telecommunications and IT managers are efficiently and effectively addressing the frustrating VoIP quality issues plaguing their converged voice environments -- once and for all.

At the forefront of providing the tools that ensure VoIP quality in a converged network, is Tone's ReliaTel solution which allows enterprises and managed service providers (MSPs) to proactively manage both VoIP call quality and the underlying converged network infrastructure in a holistic manner, through a single pane of glass, eliminating the need to access and manually associate diagnostics from multiple applications.

ReliaTel provides a platform-agnostic, comprehensive VoIP QoS and converged network management system in a unified solution, currently available in a fully-hosted, premises-based or turnkey deployment and licensing option. ReliaTel provides deep analytics and improved visibility that spans both the voice and data domains, providing voice and network support teams with the answers and capabilities needed to effectively assure VoIP call quality and service levels that consistently support and drive the business.

Confirmed Interoperability Between Brekeke’s Telephony Platform Products and JeraSoft VoIP Carrier Suite

VoIP Monitor — Wed, 23 Feb 2011 16:13:45 GMT

Brekeke has confirmed interoperability between JeraSoft VoIP Carrier Suite and Brekeke’s telephony platform products, including Brekeke SIP Server and Brekeke PBX. JeraSoft VCS serves telecom operators worldwide with customer usage of up to 500 million minutes per month. This new interoperability between our products allows service providers that use Brekeke’s platform products to add billing capability with minimal deployment time.

Brekeke Software is committed to delivering products that meet growing operational needs and scalability for telephony service providers. As demand for VoIP telephony increases from businesses and home users, the telephony platform needs to provide scalability to respond to demand, reliability to provide uninterrupted service, and high quality to ensure cost-effective operation. JeraSoft’s VoIP billing system meets these goals and brings many additional benefits to service providers, not only with their rich feature set and convenient module options, but also with their proven quality.

A configuration sample for connecting the JeraSoft VoIP Carrier Suite with Brekeke SIP Server is available at Brekeke’s wiki site here: http://wiki.brekeke.com/wiki/Jerasoft-Development-Jerasoft-VCS

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

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

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

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

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

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

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

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

Aptela Extends VoIP Market Leadership with New Call Control Functionality

VoIP Monitor — Wed, 16 Feb 2011 17:07:46 GMT

Aptela announces the availability of Aptela v5.3, the latest release of the company’s cloud-based calling platform. Aptela v5.3 offers businesses the ultimate control to manage their business communications.

“With the release of Aptela v5.3, our customers now have the enhanced call control capabilities they need to easily manage their business communications, no matter their location,” said Ann Santorios, Vice President of Product and Business Development for Aptela. “We are pleased to continue to offer the latest innovations in calling functionality to meet the needs of small businesses and mobile workers on-the-go.”

Major enhancements delivered in v5.3 include:

Find Me List Management – Allows users to easily create and manage a Find Me List within Aptela’s intuitive User Dashboard.
Follow Me – Temporarily overrides an established Find Me List, redirecting all calls to a designated phone number or extension. This provides users with the ability to quickly change their standard Find Me List to be reachable at one specific number; perfect for unexpected situations such loss of reception on a mobile phone and needing to redirect calls to an alternate number.
Do Not Disturb – Temporarily overrides a Find Me List, redirecting all calls to voicemail.
Smart Find Me List – Granting the ultimate power over how calls reach users, Find Me Lists can be created for designated contacts, colleagues or department calls. Many customers use this as a shortcut for their bosses, VIP customers or family members.
Department Recording – Expands call recording to Departments, allowing a business to enable/disable call recording at the department level. This is an ideal tool for coaching and training.
Firefox Plug-in – Provides users with click-to-call capabilities. The plug-in adds the Aptela icon right to the Firefox toolbar, allowing access from one’s browser. Firefox is a trademark of the Mozilla Foundation.

As an innovator of hosted PBX and VoIP, Aptela gives small businesses the tools required to intuitively manage and take control of their business communications through VoIP phone systems. Written in Erlang/OTP, and using CouchDB as its data store, the v5 platform demonstrates the power of cloud computing with horizontal scaling, robust fault-tolerance and a remarkably high degree of concurrency. The platform delivers a robust set of APIs that unifies communications for Aptela’s customers by integrating calling into their existing business processes. All combined, Aptela has rebuilt the new platform from the ground up utilizing the technologies best suited to deliver a high quality customer experience.

Aptela v5.3 is the latest release of Aptela’s innovative calling platform.

Viber Turns to SPIRIT for Quality HD Mobile VoIP Calling on the iPhone

VoIP Monitor — Tue, 15 Feb 2011 16:51:48 GMT

SPIRIT DSP announces that its TeamSpirit Voice Engine Mobile is powering the new Viber application that offers free HD VoIP calls from the iPhone.

Available in the Apple App Store, Viber's free mobile VoIP app can make calls over 3G and Wi-Fi connections, bypassing the use of any cellular voice minutes; calls are free worldwide. Unlike other VoIP services such as Skype, Viber requires no registration or need to log-in to make calls. Viber product releases for Android, BlackBerry and Symbian are now in the works supported by SPIRIT.

SPIRIT's TeamSpirit Mobile enables HD-quality voice and video communications on a broad range of mobile devices, addressing all impairments inherent with IP networks - such as congestion, echo, noise suppression, latency, delay (jitter), packet loss, lip synchronization, etc. - to secure high quality voice over Wi-Fi, LTE/WiMAX and 3G/4G networks. It includes highly optimized standard voice and video codecs, such as G.711, G.723, G.729, iLBC, SILK, H.263, H.264 and SPIRIT's patent-free wideband IP-MR codec. TeamSpirit mobile voice engine is available on all popular smartphone operating systems including iPhone iOS, Android, Windows Mobile and Symbian.

Unicoi Systems Exhibits at Embedded World 2011: InstaVoIP Embedded to be Unveiled

VoIP Monitor — Mon, 14 Feb 2011 18:02:43 GMT

Unicoi Systems announces the coming unveiling of its InstaVoIP Embedded software solution at Embedded World 2011, on March 1-3 in Nuremberg, Germany. InstaVoIP Embedded will provide developers with a platform-independent software suite for adding VoIP or Radio over IP functionality to virtually any embedded device.

InstaVoIP Embedded builds on Unicoi’s industry leading Fusion Embedded RTP and SIP protocols by adding a full-featured Call Manager, Voice Engine, and Information Subsystem, providing developers with a comprehensive VoIP software suite designed for use in products such as IP Phones, VoIP ATAs, RoIP Gateways, security call-boxes and many others. InstaVoIP Embedded comes in C source code, thus allowing it to run on virtually any embedded operating system or processor.

CommuniGate Systems launches Pronto! Mobile

VoIP Monitor — Thu, 10 Feb 2011 19:20:59 GMT

CommuniGate Systems launches Pronto! Mobile to join the Pronto! suite of applications; supporting telecom operators to deliver white-label flexible HD voice and data applications Berlin, Germany, 10th February 2011 – CommuniGate Systems, the leader in super efficient, massively scalable Unified Communications technology, today launches Pronto! Mobile. The next generation of CommuniGate Systems’ acclaimed Unified Communications application suite Pronto! version 4.0 now adds Pronto! Mobile, a high quality mobile application enabling operators to deliver value added communication applications under their own brand to business and residential subscribers.

Pronto! Mobile is a secure white-label mobile VoIP application that includes network address book, instant messaging, group chat, file transfer and conference calling, which is owned, branded and controlled by the operator. According to recent research by Allot Communications, demand for global mobile data bandwidth soared by 200% in 2010. With VoIP and IM the second-fastest growing services up by 87%, operators need to take advantage of this huge demand.

Pronto! Mobile extends the award-nominated Pronto! feature set to iPhone and Android devices enabling operators to deliver compelling HD Unified Communication applications to their subscribers on desktop, web and mobile devices that retain and grow subscriber numbers and loyalty, whilst increasing their brand awareness and equity.

The Pronto! application suite is today being used to empower operators with Skype-like communication services, enabling subscribers to chat, send files, and make and receive phone calls in super-clear HD Voice, but under in the operator’s own branded application.

Pronto! Mobile will be on display at Mobile World Congress next week. www.communigate.com/mwc

Media5 to Showcase Media5-fone at Mobile World Congress 2011

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

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

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

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

Floreat Announces FloVFoIP Integrated Software Suite of Host and Embedded VoIP and FoIP Solutions

VoIP Monitor — Thu, 10 Feb 2011 18:47:44 GMT

Floreat announces the immediate availability of FloVFoIP, an integrated software suite of host and embedded VoIP and FoIP solutions, as well as product-engineering services for global markets. This will effectively meet the growing demand for Multimedia products that support more than just Voice. The main applications are IP enabled SOHO and other gateways, Media consoles, next generation Internet-enabled Fax and Voice machines, Green Fax, Smart Phones, Tablets, VoIP Fax, T.38 VoIP and other specialized applications. In addition to feature enhancements and reduced time-to-market, this suite of FloVFoIP software and associated services lower manufacturers’ costs by more than 50% relative to other solutions.

With increasing e-communications and declining PSTN deployment, it was assumed that Fax would disappear; to everyone’s surprise, its presence has grown steadily to meet the requirements of corporations, lawyers, hospitals and even home-users. With VoIP now widely-deployed and accepted, many VoIP users need reliable Fax communication over their VoIP lines. This is made possible by products such as our FloFoIP and FloVFoIP that enable Voice and Fax communication over the Internet.

With the goal of meeting any application or communication requirement, Floreat has integrated its VoIP and FoIP software products, using 20 man-years worth of proven legacy voice and fax technology, deployed by more than a dozen industry leaders. In addition to making the integrated VFoIP work seamlessly, Floreat’s engineering team has taken care of quality related issues at the code level and thus enhanced the user-experience. FloVFoIP, our integrated VoIP and FoIP software solution, include an integrated VoIP and FoIP Framework; G.711, G.72x, iLBC codecs; VAD/CNG; G.168 Echo Canceller; Jitter Buffer management; PLC, DTMF; CP tones; G.167 AEC; Voice-Fax discrimination; T.38 FoIP relay with V.17, V.29, V.27ter, V.21ch.2 Group3 fax modems; Terminating T.38, T.30 Fax Application Software with Tiff support; QoE and performance enhancements; support of call control stacks – SIP, H.323, and others. FloVFax also supports Fax and Voice over RF links, and enables Green Fax that eliminates ink, paper and thermal transfers on mobile and portable communication devices.

Floreat also offers product-engineering services for software, firmware and hardware development. Floreat’s team has developed this expertise while enabling and supporting customers’ products with its software. The product-engineering services include software porting on embedded processors and DSPs, software optimization for footprints and CPU load, customization such as integration within any specified OS environment, development of device drivers, prototype hardware-design, and support of various AFEs. Our well-equipped test-lab facilitates testing and accommodates other development requests.

These unique, robust FloVFoIP offerings provide our customers a single source for their Voice and Fax requirements and overcome the challenges inherent in other solutions, such as incomplete vendor-offerings that do not permit a combined VoIP-FoIP product, lack of inter-operability with the numerous gateways and 100+ fax machine models available on the market, and inflexible incumbent (and obsolescing) hardware chipsets.

Processors Supporting Integrated VoIP and FoIP Software (FloVFoIP):

ARM 7/9/9E/Cortex A8
MIPS
TI C5000 and C6000
Intel Pentium fixed and floating point, Atom
Marvell XScale
ADI Blackfin (BF53x)
PowerPC, STM, SuperH cores
CEVA DSPs

Tone's ReliaTel Delivers Comprehensive SIP Management for Converged VoIP Environments

VoIP Monitor — Wed, 09 Feb 2011 18:17:07 GMT

Tone Software announced the immediate availability of SIP management through its ReliaTel VoIP quality of service and Converged Infrastructure Management software. The SIP management facility strengthens the ReliaTel core competency of managing voice quality, network service levels and the physical voice and data infrastructure from end to end, across virtually any convergence technology mix.

With SIP trunking permeating corporate telephony networks at a rapid pace, SIP technology is becoming a foundation for deploying and implementing unified communications, mobility, collaboration and telepresence applications. Further, SIP trunks are often justified based on expected cost-savings for enterprises. As a result, effectively managing SIP trunks is imperative to ensure the service levels of the SIP transport layer, the quality of voice traffic as it enters the corporate network, and the expected financial ROI.

Tone's ReliaTel solution provides a platform-agnostic, comprehensive VoIP QoS and converged infrastructure management system in a unified solution, currently available in a fully-hosted, premises-based, or turnkey deployment and licensing option. ReliaTel's SIP management provides significant value in SIP deployments, enabling enterprises to monitor the Quality of Service of their VoIP call traffic as it enters their corporate WAN/LAN, as well as the physical health and performance of the SIP connected devices and links.

ReliaTel's ability to monitor and analyze VoIP QoS in real-time provides immediate awareness of quality degradation occurring at the SIP trunk, and the specific metrics and facts related to the quality or service issue. Armed with this critical data, voice teams can engage SIP providers earlier and work directly with their carrier or Internet Telephony Service Provider to resolve the correct quality problems, rather than wasting time chasing the wrong root cause of quality issues they assume are originating within their own network. As a result, quality issues are resolved more rapidly, with far less negative impact to overall business communications.

ReliaTel converged network management also enables users to monitor and manage their physical SIP architecture -- including session border controllers, gateways and softswitches -- for device health, link status and utilization statistics. This ensures the physical network is available and capable of moving the voice traffic between SIP trunks and the corporate VoIP network.

Further, ReliaTel's strong VoIP QoS and service level reporting capabilities enable voice teams to easily track trends in voice quality, availability, utilization and performance of their SIP trunks over time. These metrics can be invaluable in validating the SLA compliance of their SIP carrier or ITSP, or to document non-compliance with SLA commitments.

Available immediately, ReliaTel SIP management is provided within the core capabilities of the ReliaTel VoIP QoS Management and Converged Network Management core application.

4PSA Demonstrates Cloud Unified Communications at ITEXPO East 2011

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

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

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

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

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

CommuniGate Systems Launches Pronto! Version 4.0

VoIP Monitor — Thu, 03 Feb 2011 18:16:54 GMT

CommuniGate Systems announces the next generation of their acclaimed Unified Communications application suite, Pronto! version 4.0. The new and improved features are available in both the Flash-based web version, Pronto! Web, and the new desktop client Pronto! Pro. Both web 2.0 and desktop versions use the same super efficient code base, and have an API to integrate to other business productivity applications. Pronto! integrates all types of communication into a single dashboard, allowing operators to offer subscribers one application, one experience, under one brand.

Pronto! is the only fully Unified Communication application to support telecom operators to deliver compelling, flexible voice and data applications under their brand, to their business and consumer subscribers wherever they are, using whichever internet connected device they choose. Powered by the award winning CommuniGate Pro server technology, Pronto! boasts High Definition Voice VoIP and Video, mobile VoIP, social network integration (including Twitter, Facebook), virtualized IP PBX, HD conference calling, instant messaging & presence, email, address books, calendaring, collaboration, mobile synchronization, personal file storage and more.

Pronto! Web is built on Adobe Flash technology and available to users via any internet browser. It is entirely customizable and modular, allowing the operator to deliver the services they want, under their brand.

Pronto! Pro delivers the Pronto! experience to the desktop utilizing the Adobe AIR environment and supports a white-labeled Skype-like communicator mode enabling subscribers to chat, send files, and make and receive phone calls in super-clear HD Voice.

4PSA VoipNow 2.5.1 Simplifies Equipment Provisioning and Creates Opportunities for Wholesale Carriers

VoIP Monitor — Tue, 01 Feb 2011 21:00:55 GMT

4PSA announces a new release of the award-winning VoipNow Professional software. The latest version introduces a redesigned provisioning module that simplifies equipment provisioning, a new functionality marketed as Service Provider Template for the use of VoipNow partners, as well as many other improvements that enable service providers to increase the quality of their services and offer more functionality to their customers.

Another important feature of VoipNow 2.5.1 is the Service Provider Template that creates new opportunities for many companies in the industry. With the help of such templates, wholesalers offering SIP trunking services can easily market their offer to service providers using VoipNow and help them with the technical setup of the SIP trunks.

Many service providers have trouble in selecting reliable wholesale carriers, especially on the global market. Thanks to these new facilities, carriers can present their offer to service providers using VoipNow and, at the same time, they can automate the SIP trunk setup process with predefined rules. Service Provider templates can be digitally signed to avoid tampering, which means that they can be safely distributed over the Internet.

Incoming and outgoing call rules have been improved with new matching rules and actions. One particularly interesting functionality allows mobile devices on the public telephone network to act like internal PBX extensions during a call, without using native VoIP connectivity. Other improvements included in the new version feature integration with external MWI servers, call queue enhancements such as "Whisper mode" or call confirmation by call agents. Furthermore, CallAPI has also been enriched with new resources for developers.

Current customers with a valid update subscription can upgrade to VoipNow 2.5.1 free of charge in only a few minutes. The new version is also shipped with easy-to-use images for VMware, Parallels Virtuozzo or Amazon EC2 Cloud.

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

SPIRIT Powers QIP - A Unified IM Service for PC and Mobile Users

VoIP Monitor — Fri, 21 Jan 2011 19:26:15 GMT

SPIRIT DSP announces that its TeamSpirit 3.1 Voice and Video Engine is now powering a new version of QIP – a unified IM, serving about 20 million users in Russia and other countries.

The new version of the IM, called “QIP Infium” allows users to send instant messages and make PC-to-PC and PC-to-mobile IP voice and video calls. QIP Infium is also integrated with popular social networks, and allows sending SMS.

“QIP is a unified Russian IP service that now includes VVoIP-client," says a QIP representative. "Thanks to innovative SPIRIT technologies, QIP offers the full range of cutting edge features for Internet communications via a single application. High quality and reliable performance on any network and with any terminal equipment allows our subscribers to make IP calls wherever they are.”

“TeamSpirit is a proven voice and video over IP software platform for cross-terminal unified communication applications, such as QiP Infium and a number of other popular IP services provided by the largest world carriers, including Korea Telecom and China Mobile,” said SPIRIT's Chairman Andrew Sviridenko. “TeamSpirit enables telecom operators and service providers to offer interoperable multipoint videoconferencing IP services for the most popular PC and mobile endpoints, including Windows, Mac, Android and iOS platforms. SPIRIT’s record-setting H.264SVC-based video server software supports up to 1000 concurrent video channels on a standard $4,000 PC, drastically cutting down video hardware infrastructure costs and offering HD quality videoconferencing services to millions of users.”

VoxOx Releases its Latest Version Named ''New VoxOx''

VoIP Monitor — Thu, 06 Jan 2011 20:10:31 GMT

VoxOx announces the official release of its latest version, dubbed the “new VoxOx.” With this release, the VoxOx desktop software application presents a dramatically redesigned look, feel and feature integration. New functionality includes: VoxOx Call Connect – a service that allows a user to trigger a VoIP call on any device; Voicemail transcription – the first ever to be integrated into a desktop messenger style application; Contact merging; Expanded user web portal (my.voxox), and more. VoxOx has also debuted a new visual brand platform, which will serve as the foundation for the brand’s fresh strategic direction, focusing on the concepts of unification and liberation.

Software Overhaul
In earlier versions, VoxOx has been designed to present the richest possible feature set, offering consumers an unbridled power tool. The latest release of the VoxOx software builds on this foundation, and hones in on streamlining the user flow while integrating features into a powerful, new user experience. This new release has a fully redesigned UI, sleeker look and feel, and a complete reorganization of existing functionality. Some features have also moved from within the desktop software to a newly expanded online user portal and web store at http://my.voxox.com.

Technology Innovation & New Features
The majority of the new VoxOx functionality purports the idea of unification, starting at its communicator core with the debut of the industry’s first fully integrated messaging window. In past versions, each mode of conversation was delivered in a new window; however, in this release, all forms of communication are threaded seamlessly in a single conversation with a contact, allowing users to switch between communication modes, networks, even languages – all in one window. The release also includes a new free voicemail transcription service that integrates with the messaging window in an unprecedented way, incorporating text transcription of voice messages into IM, text, and email formats. This is coupled with the ability to select a free U.S. phone number nationwide (rather than have one assigned at random, as with previous releases). VoxOx has also launched an innovative form of contact merging that enables users to reach a person through any mode or network from a single contact view, thereby providing the ultimate interactive address book experience.

Furthermore, VoxOx has introduced a unique software and service integration that allows a user to place VoIP calls from any calling device – mobile, office phone, home phone, computer desktop, etc. – all through a service called VoxOx Call Connect. A combination of VoxOx’s built-in softphone, web and SMS callback features, this new service enables consumers to initiate a VoIP call (at cheap calling rates) between any two calling devices worldwide, even landlines. Within the desktop software, VoxOx Call Connect allows users to select the calling device from which they would like to place their VoIP call, and then, auto-dial any contact in their VoxOx buddy list on any of their respective phones. VoxOx Call Connect also enables users to initiate VoIP calls away from the computer through its SMS service or by calling into the service from any phone.

New Branding, New Initiatives
With the debut of the VoxOx technological advancements comes a new brand identity. The new identity consists of a fresh VoxOx visual brand platform, which features a new logo with an embracing, double-crescent avatar that forms a subtle “V” and a bold new web site to match. The new brand direction also includes cause awareness activities built around the tagline “Speak Free,” and entertainment-themed marketing initiatives, which will commence at the Consumer Electronics Show, January 6-9, 2010 in Las Vegas, Nevada, and unfold further in 2011. Flash crowds, iPhone giveaways, petition drives and related activities will make an appearance at the show.

Voice2Phone Releases Auto Dialer - Simple Voice Broadcasting via VoIP

VoIP Monitor — Mon, 03 Jan 2011 19:06:35 GMT

Voice2Phone announces the release of Auto Dialer tool. The product automates the process of broadcasting voice message by phone making it much easier to perform phone surveys, send voice notifications or reminders to clients.

Auto Dialer is automatic phone broadcasting software working both via VoIP service providers and through hardware PBX systems, or via software PBX including Asterisk. The program takes a list of phone numbers and dials all of them simultaneously (or one by one, depending on the VoIP provider) playing the pre-recorded message to the line and recording an answer if needed.

Thanks to flexible configuration, Auto Dialer detects and accurately handles answering machines and voicemail boxes. It stays on line and waits until the respondent will be ready to receive the voice message. It is also capable of recording the voice answer and the keys pressed on the phone, which is indispensable in phone surveys.

What distinguishes the product from alternatives is its Wizard-like interface. Following the steps of the configuration Wizard, a user easily sets up VoIP providers settings, records an audio message to be replayed to the line, and imports a list of phone numbers. The entire configuration process lasts literally 2-3 minutes.

All in all, Auto Dialer is an ideal solution for mass phone notification, customer surveys and reminders with virtually endless number of possible applications.

3CX Version 9 Service Pack 5

VoIP Monitor — Mon, 13 Dec 2010 20:44:51 GMT

3CX is pleased to announce service pack 5 for version 9. This update is mainly a bug fixing update and the last one for 2010. This service pack is available for download from the 3CX Winforms Management console / Service pack updates and will upgrade your 3CXPhone System V9 to build 15536. The update takes between 5-10 minutes once downloaded depending on the machine spec of your 3CX Phone System server. Update is 20Mb in size.

Important Notes:
Plug and Play: Plug and Play is only available for Yealink and TipTel phones at the moment. With this feature, creating option 66 in DHCP is no longer needed and you don’t need to know the IP Address of the phone or log in to the phone’s browser.

Right click on a detected Yealink or tiptel phone from the Phones Page in the management console and select “Add Extension” or “Add Existing Extension”.
Fill in all the required information and select the appropriate template for your phone model from the Provisioning Tab. Click on Apply and OK to save changes. 3CX Phone System will send a SIP NOTIFY message to the phone with the provisioning URL inside. The Phone will accept this and provision automatically.

Change Log Version 9 SP5 – Version 15536

Added: Support for Plug and Play for Yealink and Tiptel
Added: VoIP Provider templates VoIP Talk, Phonzo
Added: Ability to delete multicast entries in Phones Page (New detected devices)
Improved: Yealink Templates – Line keys are now provisioned to Line 1 for multiple Outbound calls
Improved: Cisco/Linksys Templates and added sidecar templates
Improved: Cisco/Linksys improved template modifying resync time on failure
Improved: Added new Aastra templates with sidecar
Improved: Languages for Management console
Fixed: SiP ID in bulk extension csv importing is blank
Fixed: Updating of outbound parameters in Blind Transfers
Fixed: If Forwarding rule is set to End Call, Busy prompt is now played (If option to play prompts is selected)
Fixed: Originator Caller ID is now kept when a call is forwarded to an extension which has RMS (Ring my Mobile Simultaneously)
Fixed: Problem in transfers when an extension with RMS
Fixed: Call termination improvements on disconnect or transfer
Fixed: Transfer call from Queue when queue member is configured to RMS
Fixed: Infinite Queue polling and ghost calls in queue
Fixed: Queue polling problem when caller disconnects call – call used to remain polling in the queue.
Fixed: Bug in queue that used to not poll agents when one of the agents removes option RMS
Fixed: Call disconnected when call is made using Gateways/VoIP Providers to an extension with RMS
Fixed: PBX was not sending BYE to disconnect a call made to an extension with RMS
Fixed: Issue in priority of extensions in a Ring group
Fixed: Queue announcing incorrect position.
Fixed: Queue poll call rejected by extension (hardphone) with RMS did not disconnect the Mobile phone too
Fixed: Extension profile changes, updates and Options Log in/out of queues
Fixed: Issue in Extension set to out of office outside of office hours when only Queue status was supposed to change
Fixed: Disable External calls when Out of office and remove option back in office hours
Fixed: Fax Devices can be selected when an ATA is configured from 3CX Management console
Fixed: Extensions in some scenarios used to remain set to AWAY profile
Fixed: Bug in adding of extensions in Phones Page
Fixed: Bug in holidays and configuration of Office/Specific office hours
Fixed: Bug in Rebound / Call Screening – Call screening recorded name was not being played to call recipient
Fixed: Bug in BLF order when an extension is provisioned with BLF
Fixed: Changing public IP for provisioning now updates all provisioning files
Fixed: Delete extension’s provisioning file when the extension is deleted
Fixed: Bug showing duplicate entries in Time Provisioning dropdown section

Known Issues
1.Yealink phones pick up calls from Queue will pick up wrong leg of call. Reported and fixed in Yealink nest firmware update
2.When a call to an extension with RMS is recorded and answered from the mobile phone, you will get a small audio file with no audio.
3.Queue position will not be accurate if there are a lot of ringing polling calls in the queue
4.If you have your extension configured with RMS, and you try and trigger a makecall from the Assistant, this will ring your own number
5.Polycom users: G722 will not work directly with the latest Polycom firmware and users will not be able to transfer calls. Options to fix this temporarily are to bind the extensions to media server or put PCMU/PCMA at the top of the codec list (edit templates). G722 native support will be available in Version 10

Truphone Expands Its Suite of Applications to the Desktop and Laptop

VoIP Monitor — Fri, 03 Dec 2010 18:14:48 GMT

Truphone announces that its mobile VoIP service is now available for desktops and laptops. The new desktop application has been unveiled at the same time as the company's new brand name Tru and its refreshed website.

Tru's new desktop application can be easily downloaded from www.truphone.com/applications allowing customers to stay in touch with the people that matter to them using voice, instant messenger or SMS text messaging, for free or at low cost, using their device's Internet connection.

Tru's customers will be able to transfer their PC calls seamlessly onto their mobile phones, providing the ultimate in convenience and allowing customers to stay connected wherever they are.

The launch of Tru App for Desktop comes hot on the heels of Tru's new calling plans, offering rates that are 60% cheaper than its competitors across its top 30 calling destinations. The launch is part of the company's strategy to allow customers to stay connected wherever they are at the lowest possible cost, in whichever way is most convenient to them.

The Tru App for Desktop is compatible on Windows XP/7 and Vista Mac OS X and all major distributions of Linux operating systems, and is based upon the popular features found within the company's mobile applications, allowing desktop users to:

Make and receive calls in the most convenient and cost effective way

One account across their Tru App for Desktop and Tru App for Mobile account means that incoming calls will ring simultaneously on both the desktop and mobile application, allowing the user to pick up on whichever devices is most convenient
Transfer your call from the PC to your mobile so that you can keep moving whilst talking
Purchase an incoming number for your Tru account so that friends can ring you wherever you are

Keep in touch with contacts from all their social networks in one place:

Integrate all their social and personal networks including Tru's TruFriends, Skype, Facebook, AIM, GoogleTalk, MSN, with real-time updates and synchronisation between devices

See who is online at a glance

Determine online status and presence of other Tru users, and assign your own avatar

Share documents and files by dragging and dropping directly into the app

The launch expands the company's portfolio of popular mobile calling applications for the iPhone, iPod, iPad, Android, Nokia and RIM Blackberry devices, allowing users to continue their conversation whether they are at their PC or on the move.

3CXPhone 5 Adds Video, BLF to Free Soft Phone in Smartphone Look

VoIP Monitor — Wed, 01 Dec 2010 16:47:53 GMT

3CX announces a new version of its popular free VoIP soft phone for Windows, 3CXPhone 5. New features added to 3CXPhone 5 include standards based video support, multiple SIP profiles, BLF and the ability to provision and manage all soft phone installations network wide. 3CXPhone 5 will remain completely free.

“The soft-phone is now a serious desk phone option for businesses. They are easy to manage, save on administration and are free. They also save electricity. With 3CXPhone 5, soft phone installs can be easily provisioned and managed network wide. 3CXPhone is also very easy to setup as a remote extension, allowing users to connect into the company phone system from wherever they are” said Nick Galea, 3CX CEO.

New features in 3CXPhone 5:

Video Support
Video calls can be established with the click of a button. 3CXPhone 5 video support is standards based and has been tested with popular video phones from Yealink and Grandstream.

Multiple Profiles
It is now possible to register against multiple PBX’s or VoIP providers at the same time. This allows seamless connection to multiple offices or the ability to make or pick up calls from different VoIP providers from a single 3CXPhone install.

BLF and Speed Dials
3CXPhone now includes the ability to configure up to 20 BLF lights or speed dials. BLF lights can be used to monitor other extensions and provide single click transfer to that extension.

Zero Admin
3CXPhone 5 is now much easier to manage in a network. It can be provisioned remotely with all relevant configuration information such as authentication, SIP Server details, tunnel connections and the phonebook. Users can install 3CXPhone by clicking on an HTTP link. The soft phone will be installed and automatically provisioned with the correct information so that the user can use the phone immediately. In addition, network administrators can push out software updates automatically to 3CXPhone 5, without having to visit each workstation.

Completely Free
3CXPhone is provided completely free of charge to individuals and organizations including commercial entities. All features (including call transfer) are enabled. This makes it easy for companies to deploy it on any Windows desktop without having to worry about cost or licensing hassles.

Remote Extensions are easy with the 3CX Tunnel
3CXPhone can easily be configured as a remote extension, allowing users away from the office to easily connect to the corporate phone system. The unique tunnel feature proxies all SIP & RTP traffic over a single port and makes firewall and NAT configuration a breeze. It is also possible to configure it as a remote extension using ‘direct SIP’.

Jabra Headset support

3CXPhone also supports Jabra headsets, making it a good option for call centers.
3CX Phone – key features:

Choose from several popular smart phone interfaces
Supports multiple SIP profiles
Video Support
Ability to provision the phone
Ability to remotely update the phone
BLF – monitor other extensions and transfer calls with a single click
Speed dials – configure
FREE – no license fees
Supports G.711, GSM and iLBC codecs
STUN support for NAT/firewall traversal
Jabra headset support
Works with 3CX Phone System, Asterisk and popular VoIP providers

VoX Communications' Mobile VoIP App is Now Available on Ovi Store for Symbian Phones

VoIP Monitor — Mon, 22 Nov 2010 16:59:40 GMT

VoX Communications has a downloadable mobile VoIP app on the Ovi Store by Nokia for use on Symbian mobile phones. Users of Symbian devices can download the VoIP app from VoX Communications at http://ovistore.com/search?q=vox or by going to store.ovi.com from their smartphone.

Symbian smartphone users will be able to download the VoX mobile VoIP plan with a simple signup process that will deliver the install software package straight to the device. Consumers worldwide can use the app to make calls to the United States for $.02 per minute, or less, if an unlimited plan is purchased.

Other benefits of downloading the VoX app from the Ovi Store include:

The app works on GSM or WiFi, enabling dual-mode functionality
Low cost US calling
Low cost international calling
Seamless integration with the Nokia N900 contact list
Provides users with a second line for inbound and outbound calls that also supports anonymous call rejection, call blocking and call forwarding
Allows you to add a toll free number
Allows you to add several virtual numbers
Number porting is available
Does not use up your monthly cell phone plan voice-minutes

Counterpath Officially Releases Bria Android Edition Mobile VoIP Softphone

VoIP Monitor — Thu, 11 Nov 2010 16:53:32 GMT

CounterPath is pleased to announce the official launch of Bria Android Edition. An enterprise-grade VoIP application, Bria Android Edition is a highly secure, standards-based softphone that works over both 3G and Wi-Fi networks. Highlights of Bria Android Edition's features include:

Support on any SIP-compliant server to enable rapid implementation
Multi-tasking support for background operation, such as fielding incoming calls while using other applications
Audio codecs G.711a/u, iLBC and GSM
Option to purchase with or without G.729 codec
Automatic codec selection to ensure optimal call quality
Deskphone-class options such as rejecting, holding, forwarding, merging and splitting calls, as well as attended and unattended transfers
A detailed call history pane that displays dialed, answered and missed calls, along with ability to delete entries
Advanced security settings, including audio encryption
NAT traversal to balance security and ease-of-use
The ability to work with the native dialer
Integration of user contacts stored on the device
Voicemail
Optional customized branding for graphic assets and SIP settings available for enterprises and telephony providers.

Bria Android Edition is available for $7.99 USD from the Android Market or CounterPath's online store. Customers have the ability to upgrade the client to include G.729 support for an additional $8.99 USD.

Voip-Pal.Com Announces OS6 Version of Their BlackBerry App

VoIP Monitor — Thu, 04 Nov 2010 15:38:46 GMT

Voip-Pal.Com has posted a new upgrade of their PointsPhone Mobile App for the BlackBerry which now is compatible for the OS6 version of the new BlackBerry Torch and all previous versions of the BlackBerry as well. The PointsPhone Mobile App for the BlackBerry is available on www.PointsPhone.com and users can now download for free the PointsPhone Mobile App for BlackBerry. A user account must be registered.

The PointsPhone Guard will be available for delivery as a stand-alone App on our retail website. It will also be available as an Add-On for all new and existing clients. In addition, a new PointsPhone Platinum package will be available that will provide a complete telecommunications solution for Smart Phone users.

The PointsPhone Platinum package will retail for $29.95 and include a Free Virtual Number, a Free Smart Phone App (for Apple iPhone, BlackBerry, Symbian, Android or the new Microsoft Smart Phone), Talk Minutes, and a One Year subscription to the PointsPhone Guard.

Send Free SMS to 2.5 Billion Asians with Vopium

VoIP Monitor — Tue, 02 Nov 2010 16:46:40 GMT

For the rest of the year, you can send free SMS text messages to Asia if you have the Vopium application installed on your mobile phone. The application is easy to install, and in addition to the free SMS service, Vopium also offers low-cost international calls of a quality far exceeding what you may know from Skype.

With Vopium on your mobile phone, you can send free SMS text messages for the remainder of the year to your friends, family and other people in most Asian countries, such as India, Japan and Thailand, enabling you to send text messages to more than 2.5 billion people in Asia.

This offer gives everybody a chance to find out how easy it is to send an SMS to friends and colleagues if you have Vopium installed on your mobile phone. The free offer applies to most countries in Asia, but we also offer low-cost SMS rates to the rest of the world. For instance, it only costs EUR 0.09 to send an SMS to the USA, says Tanveer Sharif, CEO at Vopium.

Vopium works on more than 900 mobile phone models

Vopium is a free application that you can download to your mobile phone. All you have to do is enter your mobile number at http://www.vopium.com. You then receive an SMS text message containing a link which activates the Vopium application. You can choose to download the Vopium app directly from app stores, whether you have an iPhone, Android, Blackberry or Nokia phone.

Once Vopium is installed, you can send free or low-rate SMS and make low-rate international high-quality phone calls directly from your mobile phone. Vopium is designed to run on more than 900 types of mobile telephones, so even users of older phones can benefit from the offer of free SMS to Asia.

Vopium works via Wi-Fi and from your PC, where the offer is completely free, as well as from the mobile network where, depending on subscription type, there may be a small data charge to the provider. For subscribers with a flat data rate, the offer is, of course, absolutely free. Everyone can send Free SMS to the following countries;

Afghanistan, Bangladesh, Bhutan, Brunei, Cambodia, India, Japan, Laos, Malaysia, Mongolia, Nepal, Philippines, Singapore, Sri Lanka, Taiwan, Thailand, Vietnam, Pakistan and Indonesia.

Digium Releases Asterisk 1.8 Open Source Telephony Platform

VoIP Monitor — Thu, 21 Oct 2010 17:25:23 GMT

Digium announces the release of Asterisk 1.8, a significant update to the software that developers, integrators, resellers and others around the world use to create customized, feature-rich and cost-effective business phone systems. Asterisk 1.8 includes more than 200 enhancements, including new security features, integration with IPv6 and extensive additions to ISDN-BRI functionality. Asterisk 1.8 is designated as a Long Term Support release, indicating four years of support from Digium. Hundreds of members of the open source Asterisk community will take an in-depth look at the software's new capabilities at next week's AstriCon Conference & Expo, which runs from October 26-28 near Washington, D.C.

Technologists working with small and large businesses, government agencies and municipalities, call centers and carriers have downloaded Asterisk more than two million times over the past 12 months. It can be used to create nearly any type of phone system or voice application, with some of the most popular uses being IP PBXs, voice gateways, voicemail, interactive voice response, conference bridges and automatic call distributors for call centers.

Multitude of New Features Boost Security, Scalability, ISDN-BRI Support

Asterisk 1.8 includes contributions from hundreds of community developers, as well as the Digium development team. For a comprehensive list of additions and changes, and to download the new version, visit http://www.asterisk.org. A few of the updates include:

Secure real-time transport protocol support-New end-to-end VoIPencryption of signaling and media to compliment the existing encrypted signalingsupport.
Security event framework-Modular capability for collecting and distributing security events within Asterisk.
Extensive additions to ISDN-BRI functionality-Call completion services, connected party identification, ETSI advice of charge, message waiting indicator, call rerouting and call deflection.
Session initiation protocol changes-Substantial increase in the speed of registrations, transport layer security improvements and more flexible network address translation handling.
IPv6 support-Integration with next-generation networks.
Calendar integration-Support for Microsoft Exchange, CalDav and iCalendar.
Channel event logging-Enhanced call tracking and logging for better audit trail and billing purposes.
XMPP distributed messaging-Better scalability for message waiting and device state.
Improved internationalization and localization-Asterisk offers improved handling of concatenated audio playback (dates, numbers).
Google Talk and Google Voice support-Inbound and outbound support for Google Talk and Google Voice calling.
High-resolution timestamps for call data records-Carrier and enterprise users can track call times to the microsecond.
Better support for voice codecs-16 kHz signed linear media streams are now supported. Additional HD voice codecs supported.
PacketCable NCS 1.0 support-Allows cable companies to use Asterisk as an option to create business services.
Default de-noise for conference bridge calls-Conference calls will sound clearer.
ConfBridge application enhancements-DAHDI hardware is no longer required to use this software feature. New call conferencing application that does not require the DAHDI kernel interface to operate.
Pitch shift functions-The pitch of audio, including of callers' voices, can be manipulated.
Multicast RTP paging-Extremely efficient and scalable method for handset paging.
Faster development and more robust unit testing-Digium has implementedAgile development and a new automatedtestingframework. The Agile process streamlines development and gives Asterisk users a better view into development plans.

Asterisk 1.8 is released under the GNU General Public License. It is free of charge and available for download at http://www.asterisk.org.

Edgewater Networks Delivers IP Phone Monitoring and Configuration Management Enhancements in EdgeView Version 10.5

VoIP Monitor — Wed, 13 Oct 2010 16:19:10 GMT

Edgewater Networks announces the availability of EdgeView VoIP Support System software version 10.5 which includes enhancements that reduce operating expense for network operators and improve end-user satisfaction. The award-winning EdgeView VoIP Support System is used by leading service providers and enterprise organizations that maintain and manage converged voice and data networks. EdgeView provides visibility into VoIP call quality performance, auto-provisioning of IP phones, simplified device administration and ongoing VoIP network monitoring.

EdgeView software version 10.5 delivers the following major enhancements:

Support for VoIP call quality statistics from Aastra and Polycom IP phones. The EdgeView system will store and present important VoIP call quality statistics including Mean Opinion Score provided by the IP phone endpoints.
Network operators can retrieve call quality statistics directly from the IP phones as well as from all of the EdgeMarc monitoring points in the network with a single integrated view. A typical “on-net” VoIP call between two locations would result in a total of six call quality scores that represent each monitoring point in the call path. In the case of a poor quality call this type of reporting enables the network operator to significantly reduce problem resolution times by quickly identifying which portion of the network is responsible for negatively affecting the end user experience.
Scheduled firmware upgrades for EdgeMarc Network Services Gateways at the node and group level. This feature enables network operators to schedule maintenance or feature upgrades for off-hours maintenance windows.
IP phone registration state is now displayed and retrievable via search to facilitate troubleshooting.
The ability to compare device backup configuration files has been extended to Edgewater Networks’ EdgeConnect Managed Power over Ethernet Switches and 3rd party IP Phones being managed by the EdgeView VoIP Support System. This powerful feature enables network operators to compare known working configurations with the current running configurations of EdgeMarcs, EdgeConnects and IP phones to quickly determine if a new configuration has caused problems.

The software release contains over 20 other enhancements and a detailed release note can be found in the support knowledge base at www.edgewaternetworks.com.

8x8 Announces Release of Virtual Office Pro 2.0 Upgraded with New Features

VoIP Monitor — Mon, 04 Oct 2010 16:39:20 GMT

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

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

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

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

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

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

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

Ingate Bundles Free SIP IDS/IPS Security Module with Firewall, SIParator Enterprise SBCs

VoIP Monitor — Mon, 04 Oct 2010 16:19:28 GMT

Ingate announces that Intrusion Detection System/Intrusion Prevention System solutions for SIP are now bundled as a free software module on Ingate Firewall and Ingate SIParator products. Firewalls and SIParators already in use can be upgraded for free.

With IP attacks to steal VoIP service a genuine threat, weak passwords still providing an opportunity for malicious activity and the potential for overloading VoIP systems a possibility, IDS/IPS has become a crucial security measure for enterprise deployments of SIP. Ingate is giving its SIP IDS/IPS module to all users of Ingate Firewalls and SIParators to protect against attacks targeting SIP devices, such as IP-PBXs. IDS/IPS works in tandem with Ingate's existing security technologies, further strengthening security for VoIP, SIP trunking, Unified Communications and other SIP applications.

Maximum VoIP Security

In addition to SIP IDS/IPS, Ingate also offers an optional Enhanced Security software module. Enhanced Security offers the ability to encrypt all SIP communications which makes it impossible for eavesdroppers to retrieve the original media streams or signaling.

The Enhanced Security Software Module includes:

Transport Layer Security which authenticates communication parties and encrypts the signaling on the public side, even if it is in the clear on the LAN.
Secure Realtime Transport Protocol, which adds encryption when the voice media streams are transported outside the enterprise LAN. When combined with TLS it further shields users from eavesdroppers, hackers and spoofers.

Availability

IDS/IPS is available now as a free software download for all Ingate Firewalls and SIParators.

For more information visit Ingate at ITEXPO West, October 4-6, 2010 at the SIP Trunk-Unified Communications Summit -- free to all ITEXPO attendees, in room 403A of the Los Angeles Convention Center.