Spanish English Portuguese
Licenses, updates and accessories of switchboards.
Issabel® is an Open Source Software to establish Unified Communications. About this concept, Issabel® goal is to incorporate all the communication alternatives, available at an enterprise level, into a unique solution..
Telephony was the traditional way that lead communications the last century, that’s why many Companies and users focus their requirements on their necessities to establish telephony communications on their organizations, and confuse unified communications “distros” with a telephone exchange system. Issabel®, not only provides telephony, it integrates other communication alternatives to make your organization environment more productive and efficient. .
Licensing in Issabel®
Issabel® is an open source entrepreneur tool relased under the license GPLv2. You are free to use it for commercial of personal purposes subject to the conditions as described under its License.
Issabel® doesn’t have a cost related with licensing or functionality.
All the Issabel® versions are full versions without limitation in its use or its features. Nor the addition of modules or users, in an Issabel® implementation have a cost involved to the integrator, organizations or enterprises in desire to use Issabel®.
We work every day of the year thinking how to improve your communications and designing new versions of Issabel®.
If you are interested in a personalized solution, please contact our comercial department: sales@avanzada7.com
Front Panel of NLX4000 for TE133
Rack Mount Kit
Epygi's Rack Mount Kit is designed with you in mind. The smaller, more modular QX products integrate seamlessly with the Rack Mount Kit, ultimately reducing clutter and maximizing space. With the exception of the 19-inch QX2000, Epygi's line of QX products simply slide into the rack mount and are secured with a thumb screw.
The kit, purchased separately from the IP PBXs and Gateways, includes two DC power cables for power redundancy. Power redundancy can be used on all QX products with the exception of the QXFXS24.
Please refer to the technical data sheets for our IP PBXs to see the maximum number of Gateways that can be interconnected.
In 2000, Epygi Technologies was one of the first IP PBX manufacturers for the small to medium-sized business market. Since then, we have continued to be a leader in the telephony industry. Epygi, a worldwide provider of award-winning IP PBXs and Gateways, was founded with the idea that every company, regardless of their size, should have the opportunity to own the latest in feature-rich and economically-priced VoIP technology.
Epygi’s headquarters is in Plano, Texas, where all of our sales, marketing, executive development and manufacturing takes place. We also have an international location in Yerevan, Armenia, where our R&D, product design and software development occurs. Throughout the company, we employ over 40 engineering, administrative and marketing professionals, with sales and support presence in 40 countries across the globe. We believe in fostering a tight knit corporate community where innovative ideas are encouraged, curiosity is welcomed and teamwork is valued. Our employees are truly our greatest asset.
You may be wondering, “Why ‘Epygi’? What does it mean?”. The story begins in the year 2000 with our founder searching for a company name similar to the word “apogee,” which means “the farthest or highest point.” It was through the meaning of this word that the name, “Epygi,” was finally developed. Today, we continue to strive for the “highest point,” by providing your business with the most innovative telephony solutions in the industry.
Epygi QX DC power cable
Cable to offer Redundant Power solution between two modules Epygi QX. With this cable, a module QX can offer power to another module, in case one of them loses AC power. It's clever way to provide redundant power at combine two or more modules Epygi QX.
Digium Front Panel MC100
OptiCaller is an application for Mobile PBX with unique call optimization. It allows the user to make calls in a flexible, and above all, more cost-effective manner (average 40-95% cost savings). OptiCaller also makes it easy to manage PBX functions, e.g. call diversions and presence status, directly from the mobile phone.
The architecture is operator independent, which means that the application works regardless of mobile operator. The application has very flexible configuration options and is adaptable to suite the majority of PBX systems on the market. Configuration and deployment of clients (OTA) is easily handled by a powerful provisioning system.
OptiCaller always takes control over how each call is connected and is thus giving the user the opportunity to call in the most cost efficient way. The call can be connected in three different ways
The call method can either be predefined or decided at each call with our Always Ask functionality. It is also possible to create rules that control the call method.
OptiCaller is compatible with Asterisk and many more.
Ampliación de garantía a 3 años para la Switchvox AA305
Ampliación de garantía a 3 años para la Switchvox AA65
Garantia Extendida 3 años para AA355
Rather than the 64kbit/s required for a standard, uncompressed G.711 PCM audio data stream, the G.729 codec compresses the payload to 8kbit/s. Bandwidth calculations for a VoIP call should consider signaling and packet overhead as well, which varies according to network topology. In a typical Ethernet environment and utilizing the SIP or IAX signaling protocols, a G.711 call will consume about 87.2kbit/s while a typical G.729 compressed call will consume about 31.2kbit/s.
A practical example is the number of calls that may be carried across a standard 1.5 megabit/s T1 link. When using uncompressed G.711 audio, one can expect 18 concurrent calls across a T1. And, when using G.729 compression and Digium's IAX2 Trunking, instead of SIP, signaling protocol, one can expect about 140 concurrent calls across the same link.
Digium's implementation of the G.729 Codec in software allows Asterisk to transcode (compress and decompress) audio to and from formats other than G.729. Many business-class IP telephones and VoIP gateways include support for G.729. With the Digium G.729 Codec for Asterisk, those devices can now exchange calls with Asterisk directly.
Without the capability to transcode G.729, Asterisk can only pass-through G.729 data between endpoints. This means that basic station to station calling can be made to work, but the advanced PBX features of Asterisk such as Call Conferences, DTMF digit collection, Call Recording and more will not work without Digium's licensed G.729 Codec.
Multiple versions of G.729 are defined according to industry standards. Asterisk, and Digium's G.729 implementation support G.729 Annex A, or G.729a. Aster isk and Digium's G.729 implementation do not support G.729 Annex B, or G.729b.Digium's software G.729 Codec utilizes the power of the host system's CPU to perform its transformations. Therefore, the transcoding capacity, in terms of simultaneous channels/transcodes, is determined by the performance of the host server. Digium's internal testing indicates that 60 concurrent G.729 calls/transcodes require a system equivalent to a dual Intel Xeon at 1.8GHz. Further testing indicates that 80 concurrent G.729 calls/transcodes require something equivalent to a dual Intel Xeon at 2.8GHz.
Digium's G.729 Codec for Asterisk is licensed on a per-channel basis. A channel is defined as a single connection from an endpoint to an Asterisk applicatio n, or a bi-directional call between two endpoints attached to Asterisk. Customers may use the licensed G.729 Codec in conjunction with Asterisk and any combination of Digi um telephony interface boards.
The G.729 Codec is provided with support from Digium's Technical Support organization for Linux x86 and x86_64 environments. Digium also provides builds for other platforms, but without support.
Digium®, Inc., the Asterisk® Company, created and is the innovative force behind Asterisk, the most widely used open source telephony software. Since its founding in 1999 by Mark Spencer, Digium has become the open source alternative to proprietary communication providers, with offerings that cost as much as 80 percent less. Digium offers Asterisk software free to the open source community and offers Switchvox, Digium's Unified Communications solution to power a broad family of products for small, medium and large businesses. The company's product line includes a wide range of telephony hardware and software to enable resellers and customers to implement turnkey VoIP systems or to design their own custom communications solutions. At Digium, we're changing the way businesses communicate.
Asterisk is the world's most popular open source telephony project. Under development since 1999, Asterisk is free, open source software that turns an ordinary computer into a feature-rich voice communications server. Asterisk makes it simple to create and deploy a wide range of telephony applications and services.
Code for Asterisk, originally written by founder and CTO, Mark Spencer of Digium, Inc., has been contributed from open source software engineers around the world. Currently boasting over two million users, Asterisk supports a wide range of TDM protocols for the handling and transmission of voice over traditional telephony interfaces, featuring VoIP packet protocols such as SIP and IAX among others. It supports U.S. and European standard signaling types used in business phone systems, allowing it to bridge between next-generation voice-data integrated networks and existing infrastructure.
The Digium cards in the TE series are high performance, cost effective, with digital telephone interfaces that support the T1 and E1 environments. The environments are selectable on a base ny card or by port. This characteristic allows the translation of the signaling between the equipment T1 and E1, and allows to connect banks of economic channels T1 with circuits E1.
Sometimes it may happen that your IP communications systemn is echoed, this can happen as a result of the waiting times that a VoIP system often has, unlike an analog system. As a result, your conversation may suffer an echo.
Although many devices already incorporate the echo canceller, Digium offers the Digium High Performance Echo Cancellation (HPEC) hardware. Echo cancellation hardware is also advantageous when handling large volumes of calls or high numbers of channels that would otherwise strain the CPU which would result in poor audio potential quality. Here are some of its features:
La licencia permitirá aumentar las funcionalidades de su equipo Quadro con la capacidad de almacenar las llamadas que se cursen en el sistema.
Estas llamadas podrán ser grabadas automáticamente o a voluntad del usuario; almacenadas temporalmente en el equipo o enviadas vía FTP a otra hubicación.
Call Recording is a powerful feature allowing the system to record all calls made from and to the IP extensions of the PBX. This allows a user to record selected calls both automatically and by special request from the Web GUI or directly from the phone. The recordings could be stored either on the IP PBX (and be reviewed on the Quadro) or be uploaded to an external file storage for further processing. Call Recording is a purchasable feature priced per recording port and sold in groups of ports available on the QuadroM IP PBX products, including 8L, 12Li, 26x, 26xi and 32x.
QuadroM32x: Four 8 port licenses can be purchased for a total of 32 recording ports.
Recording can be set to record all calls or restricted based on called/caller party number or based on the digits dialed.
Record calls automatically or after pressing the Record button on the handset. Recording status displayed on Aastra & snom phones, as well as displayed on Quadro GUI.
Recorded files: .wav files using G.711
Saved, viewed or played back locally on the Quadro GUI
Saved, viewed or played back on an ftp server Optionally prompt for password before playback
The Barge-in Feature Pack is comprised of three independent features designed to help supervisors or managers of a call center or in an office environment monitor and coach their employees. These features are tools supervisors can use to participate in conversations between employees/agents and customers. In addition, supervisors can monitor their employees’ performance or customers’ behavior. The supervisor can coach an employee while he/she is en-gaged with a customer on a phone call without the customer knowing, or the supervisor can participate in a three-way call and assist both at the same time or just monitor the call of an employee and customer
Silent Monitoring Feature This feature allows the supervisor to listen to a call between an internal extension (employee) and an external call (the customer).
Agent Whisper Feature With the Agent Whisper feature, the supervisor can dial *92+extension number and listen to the conversation but only speak and be heard by the internal extension (employee) not the external call (the customer).
Barge In A supervisor will be able to join an established call between the employee and customer and have a three-way call.
Administrative Assistant To further enhance the abilities of an Administrative Assistant, the Barge-In Feature Pack adds some very interesting communication options. For example, if the executive would like the assistant to take notes during a call, but ensure they cannot be heard, silent monitoring would be used. Whisper could also be used if the assistant needed to interrupt the executive if there was a more urgent situation.
Standard hunt groups and simultaneous call distribution are not sufficient for the demands of true call center environments. Advanced methods of call distribution are required, such as Skills Based Routing or Least Active Agent. Grouping of the agents is also a key advantage, allowing call centers to logically group resources together to clearly define responsibilities and expertise of the agents. Automatic Call Distribution (ACD) is a purchasable feature only available on the QuadroM IP PBX products, including 8L, 12Li, 26x, 26xi and 32x.
Features:
Ampliación de garantía a 3 años para AA60
Ampliación de 1 usuario para centralitas AA305 ,AA355, AA60 y AA65
Ampliación de 5 usuarios para centralitas AA305 ,AA355, AA60 y AA65
Ampliación de 25 usuarios para centralitas AA305 ,AA355, AA60 y AA65
Ampliación hasta 100 usuarios para centralitas AA305 y 355
Ampliación de garantía a 3 años para AA350
Use our comparator and choose the correct product
Para NLX4000
Metal Grip for NLX4000
Compatible with OpenVox A2410E
Compatible with A8B
Epygi QX rack mounting kit
Compatible with A4B
Front Panel ISS-ENTRY for B601
Sangoma B500 front panel for MC100
Compatible with MiniUCS B601
OptiCaller is an application for Mobile PBX with unique call optimization
Front Panel ISS-450 for B601
Front Panel NLX4000 for B601
Front Panel de NLX400 para A100
Cable alimentación Molex SATA para Elastix ELX-025
Cable de conexión para discos SATA ELX5000
The most used voice codec
Epygi license for call recording (4 users)
Epygi license for call recording (8 users)
Call center advanced features (agent whisper, silent monitoring and supervisor 3-way ...
Expansion license for ACD full management with Epygi PBXs
Ampliación 1 usuario para centralitas Switchvox
Ampliación 5 usuarios para centralitas Switchvox
Ampliación a 25 usuarios para centralitas Switchvox
Ampliación a 100 usuarios para centralitas Switchvox