Month: May 2011

Panasonic Announces New Range of SIP Terminal Phones

On the 19th and 20th of May at the financial heart of Europe, in the German city of Frankfurt, Panasonic SIP Phones, in partnership with Broadsoft who specialise in IP voice software, introduced the new generation of Panasonic SIP Terminals at the Panasonic European SIP Convention 2011, aimed to manage the global communications of thousands of users. These devices operate as an open standard, developed for the transmission of multimedia applications over IP networks. They are true data processing centers as much as they are Panasonic SIP Phones, however, they are prepared to work just as well in the home as in business environments. Panasonic has prepared the launch of more than about ten SIP Phones all with their own design, unique technology and environmentally friendly ethos. Likewise, it brings to the forefront their new audio systems that produce high definition signals for all of their new range of telephones, which permits the conversations be clearer than those found in previous calls. The purpose of the meeting was to show some of the solutions and features that will emerge in the market throughout the year. The two series, Panasonic UT and the Panasonic TGP, incorporate large LCD screens with a simple interface, Wideband HD Sound, certification for use with Digium Asterisk and Broadsoft for BroadWorks and an easy configuration which also allows the use of a wide range of...

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Series of “How to” Videos Demo XE Series LCD Functionality

The XE series of Xorcom’s Asterisk-based IP-PBX appliances features a multi-function Liquid Crystal Display, or “LCD”, touch panel. This feature allows the system administrator to perform several of the most common administrative functions directly on the front panel of the PBX. Below are the first three videos in the series: How to Discover / Configure IP Address for Xorcom XE Series How to Map IP-PBX FXS Extension Numbers to Physical Ports How to Restore the IP-PBX to Factory Default Settings Xorcom XE2000 The XE2000 is an Intel Atom based Asterisk® IP PBX with PSTN / telephone ports. The stand-alone XE2000 features the Elastix™ Asterisk distribution and may be equipped with up to 32 analog FXS/FXO ports, up to 8 BRI ports, and/or a single E1/T1 PRI, T1 CAS or E1 R2 port in a single, 19″ 2U chassis. Additional PSTN / Analog phone ports can be provided by connecting external Astribank units via the USB2 ports, for a total of up to 160 PSTN / Analog phones ports, up to 200 users and up to 45 concurrent calls. Hardware Processor: Intel Atom D525 Dual Core 1.8 GHz RAM: 1 GB Hard disk: From 250GB 2.5″ (optional upgrade to 500GB) RAID1: Dual hard drive for increased system reliability Fan: 3 redundant fans Echo Cancellation module: Voice enhancement & echo cancellation (optional) USB: 2 external USB 2.0 I/O Ports (model-specific): Input/Output...

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2N® VoiceBlue Next VoIP Gateway gains Cisco certification!

The 2N® VoiceBlue Next new generation VoIP GSM gateway was successfully certified in April 2011 for compatibility with the products of Cisco Systems. It is our pleasure to inform you that 2N TELEKOMUNIKACE has successfully obtained certification of compatibility with the products of Cisco Systems, Inc. for its new generation VoIP Gateway 2N® VoiceBlue Next version 01.00.04. This certificate has great value for the company as preparation for the audit took over a year and required fulfilment of the highest quality criteria. This certificate is further evidence for our customers that the products and services of 2N TELEKOMUNIKACE are of the highest...

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D150 Voice Transcoding Series Launched by Sangoma

New Series Provides Greater Voice Transcoding Deployment Capability and Flexibility in VoIP Solutions Market Sangoma(R) Technologies Corporation (TSX VENTURE:STC), a leading provider of hardware and software components that enable or enhance IP Communications Systems for both voice and data, today launched 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...

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Microsoft Lync Server 2010 Compatibility with Polycom KIRK

Direct Interoperability with KIRK 50/60/7000 Handsets and the KWS6000 The first-ever professional  multi-cell DECT solution with direct interoperability to Microsoft Lync Server 2010 KIRK Wireless Server 6000 The Polycom KIRK Wireless Server 6000 from Polycom is a scalable IP solution for organisations wanting to give wireless handsets to up to 4096 employees. It is suited to organisations needing to provide business and mission critical on-site mobile speech and message services around an SME, very large enterprise, or campus type organisation. The KWS6000 is a wireless handset IP solution which scales up to 4096 handsets and 1024 simultaneous conversations and is easy to install and maintain. Base stations work on PoE No infrastructure cabling – components connect to the existing LAN Works with KIRK enterprise quality handsets Can provide mobile speech services in a satellite office or subsidiay Can cover a large area – supports up to 6 repeaters Provides robust and reliable communications for business and mission critical applications The KWS6000 is a SIP enabled server.  The wireless link between the handsets and base stations uses DECT protocol thus ensuring highly reliable, robust and secure communications into the VoIP network.  Coverage can be expanded by using KIRK repeaters which provide the flxibility to cope with complex coverage requirements at minimal cost. The KWS6000 works with all series of KIRK handsets.  KIRK handsets are designed and built for use in...

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