Digium TE210

The TE210 is a digital telephony card for communication interfaces that are compatible with voice mail, VoIP gateways and call conferencing. It is dual span port and works on PCI 3.3V bus type which is normally 64-bit. The card can function in T1/E1/J1 environment. It reduces the CPU usage by ten times and can increase the card density per server. This ensures faster speed and increased efficiency.

The Digium TE210 is fully integrated with the Asterisk PBX environment and is compatible with most of the Digium hardware and Asterisk software applications. The card can easily work with the traditional PBX and the modern VoIP applications. The ideal environment for this card to function is with Digium hardware and Asterisk software, which also gives an option to customize the card as per the needs of the business and create various telephony configurations according to the needs of the individual or business.

The card supports North American and Euro standard protocols, along with the PPP, HDLC, and Frame Relay data modes. Both FXS/FXO ports are compatible with the card. The Digium TE210 card is designed to provide up to 60 voice and data channels in the T1/E1/J1 interfaces. It is available on per card per port basis and allows a mix of E1 and T1 circuits. It can also be used with a compatible 5.0V PCI slots.

The Digium TE210 comes with an optional combination with VPMOCT064 Octasic echo cancellation module. It provides echo cancellation of up to 1024 taps or 128ms across all the 60 channels supported in the E1 mode or 48 channels for T1/J1 modes. The scalability of this card is increased through bus mastering architecture. The card is compatible with open source Asterisk as well as Asterisk Business Edition and AsteriskNOW. The Digium TE210 represents the next generation data and voice communication.

Digium Analog Cards

Digium analog cards are designed to be used over Public Switched Telephony Network or PSTN. These cards are made to work with the telephony software Asterisk. The purpose of these cards is to provide communication services on analog interfaces without facing any system conflicts. The cards use Digium VoiceBus technology and connect Plain Old Telephone Service or POTS with analog interfaces to enable Voice over Internet Protocol.

The cards can support 4-ports, 8-ports or 24-ports using a computer. While working with analog cards, separate gateways are not needed for the ports with the FXS or FXO module. Digium analog cards are available in six variants. TDM2400P is a full length PCI card and can support up to 24 ports. TDM800P and TDM410P are half length PCI cards can be used to support 8 and 4 ports respectively.

There are express cards in this category as well. The AEX2400, a full length card, can support 24 ports. On the other hand, AEX800 and AEX410  are half length cards and can support 8 and 4 cards respectively.  All the cards can be used along single FXS and quad FXO modules and are customizable according to individual requirements.

All the Digium analog cards are designed according to the industry standards to maximize system compatibility so that no errors or echo is recorded during communication. Additionally, the HPEC or High Performance Echo Cancellation software is available with the cards that may improve the user's experience of telephonic conversation. The users also have the choice to use VPMADT032 module for echo cancellation.

Digium analog cards are mostly used along with Asterisk software and Linux operating system for optimum results. These cards provide ultimate business Voice over Internet Protocol solutions to their clients. All the cards come with a 5 year warranty and are covered under the Exceptional Satisfaction Program.

48 Port Netgear Switches

One of the well known 48 Port Netgear switches model is the ProSafe 48 Port 10/100 Smart Switch and 2 Gigabit Ports FS750T2. This model is known for its 2 10/100/1000 Mbps copper gigabit ports and 2 Small Form-factor Pluggable (SFP) GBIC slots. The high density switch offers great value for money by providing easy browser-based management, high port density and great gigabit capability. These features include adding two gigabit uplinks to a growing business, and installing high port density in one unit box.

Capabilities relating to key switch management need to be added for the purpose of cutting down the overall cost of the device. These Netgear switches can simultaneously and quickly deliver all kinds of video, image and multimedia files within no time. Right links are made on the basis of support received from Auto Uplink technology, which is facilitated by 50 shielded RJ-45 ports known for automatically negotiating to the highest speed. Greater distance is obtained by fiber connectivity, which comes from the two hot-swappable Small Form-factor Pluggable (SFP) GBIC slots.

Simple Smart Switch Management is offered by the intuitive web-browser interface that helps in configuring ports and setting up port trunks. It also facilitates monitoring of switch performance, VLANs and traffic prioritization. Users can manage their Smart Switch by using the SNMP-based software. These switches are easy to install and can be used on a daily basis. Moreover, these Netgear switches help in extending the managed networks and adding basic management to the unmanaged networks. 

The Netgear switches cover swift transfer right from the edge of the network to the core, while delivering full-speed and non-blocking packets. The packet forwarding occurs through the forty-eight 10/100 ports and two Gigabit ports, which are known for powerful performance. Users have the option of obtaining fiber connectivity for greater distance based upon the two SFP GBIC fiber slots.

Asterisk PBX

As a small business enterprise, if you are looking for ways to expand your business in the competitive market, then Asterisk PBX by Digium is your key to success. Voice over Internet Protocol based communications are increasingly being used for networking with business clients and customers. Border-less communication is best described by IP telephony as you can communicate with a large group of people located at distant places. To build your ordinary computer system into a Private Branch Exchange communication platform, Asterisk PBX can be used.

The software is basically PBX implemented that supports IP PBX systems, conference servers and also VoIP gateways. The interface software connects PBX with analog lines and vice versa. It provides all the features that are expected from a PBX along with other services like voice mail, call conferencing, interactive voice over sessions and call queuing. Similar to PBX, this software allows you to make calls using attached telephones and also make connections with Public Switched Telephone Network (PSTN) and Voice over IP services.

A mix of traditional and VoIP telephony services, Asterisk is one of the first open source PBX software packages. Asterisk PBX runs on Linux, BSD and Windows, without any need for additional hardware for VoIP. To make use of the Asterisk PBX, all one needs is some kind of PSTN gateway or card. The system supports interactive voice sessions, enhanced messaging services, information services, auto dialing, auto provisioning, caller ID customization and call transferring. It supports VoIP and analog telephone handsets for intensive networking.

A business telephone system can be easily built using Asterisk software as a platform for PBXs. The Asterisk PBX integrated system is widely appreciated for its open design, flexibility, extensibility, open source licensing and other distinct features. Digium has also introduced AsteriskNow, a customized Linux installation including FreePBX to support PBX.

Hence the next time you think of a big business venture you should include Asterisk PBX in your plan.

Digium Digital Cards

In today’s competitive world, each company is putting in endless efforts to deliver something unique to its customer in order to stay ahead of its competitors. Incorporating latest technology and designs in manufacturing the products is becoming the key to success for every business. Digium is one such renowned name in the industry which is the creator and primary developer of industry’s first open source PBX known as Asterisk. Additionally, the company offers hardware and software products that provide flawless telephony solutions to various telecommunication providers and enterprises.

Digium has designed, developed and manufactured PC based telephony cards that are used for extending the functionality of TDM to voice processing and VoIP gateways of Asterisk. Digium digital cards are ISO 9001:2001 certified and are RoHS compliant. These cards are backed by the 'Exceptional Satisfaction Program' a service offered by the company.

Digium digital cards are the digital telephony interfaces that are cost effective and high in performance. These cards are designed in such a way so as to allow Asterisk to easily connect with T1, E1 and J1 digital lines that are also known as trunks. Digium digital cards comprise one or more ports, each of which is connected to an individual digital circuit. This feature of Digium digital cards enables signaling translations between T1 and E1 equipment. In addition, these cards also allow inexpensive T1 channels banks to easily connect to E1 circuits. Installing the Digium digital cards helps in reduction of CPU usage and helps in increasing card density per server as the bus-mastering TE cards improve the input and output speed over old architectures.

Digium digital cards are compatible with both traditional telephony systems like PBX and emerging technologies like VoIP. These cards are in compliance with the industry standard telephony and data protocols and are designed to support line-side and trunk-side interfaces. In addition, the Digium digital cards also support the advance call features. 

Digium digital cards are thus paving way for new generation telecommunications ranging from traditional PBX to recent VoIP gateways.

Yealink T20P Manual

The Yealink T20P package comes with an IP phone set, phone stand, power adapter, handset, handset core, Ethernet cable and a user manual for reference. To begin with, the user has to assemble the phone and attach the hand set and power adapter with the phone. The user can then connect the Ethernet cable in order to access the Internet, after reading the manual.

The user manual guides the user through the installation and configuration of the phone. For configuration, the user can press the OK button on the phone and find out the IP address of the phone on entering the status page. The phone would try to connect to the available DNS server and default network. The user can now change the user name, password, display name and other account settings along with the SIP server and port.

With the help of the user manual, the user can easily change the language, date, time and also adjust the volume of the Yealink T20P. The LCD screen of the phone displays numerous icons in order to show the status of the phone. This may include voice mail, call forwarding, message and call mute/hold icons. The user manual enables the user to understand this concept as well.

While reading the user manual, one may learn to define the expected pattern of digits while dialing a number. This includes automatically adding the country code as a prefix to a dialed number. The Dial Now enables the user to define the specific length of a number so that the next time it is dialed; the user need not press the send button.

In short, the user manual is an essential document that must be read before for better understanding as well as easy installation. It also aids in making optimum usage of the features of the Yealink T20P phone.

CyberData SIP Office Ringer

The CyberData SIP Office Ringer is SIP-enabled VoIP device that is highly coveted in the market. This device uses Power-over-Ethernet (PoE 802.3af) for power input, which makes it safer and economical. Using a single cable connection, the SIP Office Ringer can be connected into the existing local area networks (LANs). The device is functional under the temperature range of -30°C to 55°C.

This VoIP device is known for supporting SIP voice paging and priority-based multicast broadcasts. It also provides an audible ring indication when it works as a part of a ring group. The CyberData SIP Office Ringer is also loaded with a night ringer function. The dual speed of the device is around 10/100 Mbps. As the device is compatible with all kinds of IP-PBX servers, the installation of the ringer is easy.

For security purpose, the device is loaded with intrusion sensors. It helps the system to monitor network activities and detect malicious activities or policy violations. The firmware and other software for CyberData SIP Office Ringer can be downloaded online, which helps in upgrading the device from time-to-time for better functioning. With Solarwinds TFTP, the users can set a fixed IP address or an IP address range in the server. User can also adjust the server operation to only receive or only send files.

The CyberData SIP Office Ringer is extremely user friendly as the volume and microphone sensitivity of the network can be adjusted as per the requirement. Switching of the device is based on dry contact relay as the signals are of low level. In this office ringer, audio files with several formats like RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit and mono 8000 Hz can be uploaded. The device is available in the dimensional specifications of 5" x 5" x 2.5" and comes with a warranty of 2 years.