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Permita que su personal trabaje de forma remota utilizando la aplicación móvil Sangoma Connect para dispositivos iOS y Android. Los empleados podrán disfrutar de la colaboración de voz y vídeo de múltiples participantes de Sangoma Meet utilizando su extensión comercial sin dar a conocer la información de su dispositivo personal. El proceso de incorporación intuitivo envía un correo electrónico generado automáticamente a cada usuario en el momento de activarse el PBX con instrucciones sencillas para que descarguen la aplicación con inicio de sesión instantáneo. Así de sencillo.
UCM RemoteConnect allows businesses to easily build a secure collaboration solution for remote workers and devices. It offers a companion cloud service for the UCM6300 series that provides always-on, automatic NAT firewall traversal to ensure secure connections by remote users. UCM RemoteConnect provides powerful audio and video collaboration tools to remote users through Grandstream’s Wave desktop, web and mobile app, and SIP endpoints integrated with the UCM6300 series. This cloud service provides 99.9% reliability by running on Amazon Web Services (AWS) while offering zero-touch configuration and IT-friendly management. UCM RemoteConnect is fully integrated with the Grandstream Device Management System (GDMS), is setup and managed at ucmrc.gdms.cloud and provides cloud storage, diagnosis tools, reports and alerts. By providing a full ecosystem of remote collaboration tools, services and management for the UCM6300 series, UCM RemoteConnect is the ideal platform for any organization looking to securely support remote workers.
Founced in 2002, Grandstream Networks is the leading manufacturer of voice/video telephony and IP video surveillance systems. Grandstream addresses the needs of small and medium businesses through products that reduce the costs of communications, increasing security in order to improve productivity. Their products based on the standard SIP protocol offer wide interoperability in the industry, unrivaled features and flexibility.
Grandstream's IP Executive terminals are the next generation of IP Phones. They offer the highest telephony functions for businesses that need specific features like multiple lines and SIP identities, high-quality audio, integrated PoE and broad interoperability. They represent the perfect choice for businesses that need a high quality multi-line executive IP Phone at an affordable cost.
Grandstream's IP Standard Desktops are a new generation of ideal IP Phones as a working tool for small and medium businesses.
High-quality audio, customizable application service, soft keys, call waiting, call forwarding and transfer, tripartite conference and acoustic echo cancellation are some of the functions that have these great Grandstream handsets.
The Grandstream DECT series is presented as the new generation of high-quality, easy-to-use wireless IP phone. It offers a great variety of functionalities and a wide range of radio coverage.
Its compact size, excellent voice quality and excellent features make it a tandem with excellent value for money.
Grandstream offers several videoconferencing solutions at the user's disposal. We find, on the one hand, the series of Videophones IP of Grandstream, that combines in one device the benefits of an IP telephone with videoconference and the functionality of an Android tablet.
Incorporating a tablet's solutions gives you access to a multitude of Android apps, including business productivity such as Skype, Microsoft Lync, GoToMeeting, etc. A device that offers a multiplatform solution for all business communication and productivity needs (voice, video, data and mobility).
On the other hand, Grandstream also offers GVC3200, an Android based revolutionary videoconferencing solution, which supports multiple conferencing protocols from which we also find an adapted version for needs of small and medium businesses, the GVC3202.
Grandstream Analog Telephone Adapters (ATA) offer the ability to deploy commercial voice over IP services over an analog phone. Grandstream offers the following models to the user:
Grandstream video suveillance cameras are ready for any outdoor or indoor environment, where weather and light conditions are constantly changing. It lenses allow an adaptation according to the monitoring needs of the users allowing to monitor nearby areas, entrances to buildings, etc.
Grandstream has a wide range of Gateways ideal for any business that wants to incorporate the new IP technology into their daily activity. In addition, it is presented as the ideal solution to monetize the investment made in an analog telephone equipment, fax and tradicional PBX systems.
- GXW4004/4008 series represents the ideal solution for companies that wants to connect one or more lines from a traditional PBX to a VoIP telephone system with 4 and 8 FXS ports respectively.
- The GXW4104/4108 models convert SIP/RTP IP calls into traditional PSTN calls. Witn 4 and 8 FXO ports respectively, the installation is identical on both models and offers complete interoperability with IP PBX systems, softswhitches and SIP servers.
- The GXW4216/4224/4232/4248 has 16/24/32 and 48 analog FXS telephony ports respectively. In addition, they incorporate advanced security protection, excellent voice quality, easy provisioning and excellent performance in handling high volumes of voice calls.
In the portfolio of Grandstream, we also find this type of solutions, ideal for meeting rooms or meetings of any office thanks to its excellent mobility and audio quality. It has bluetooth, WiFi, 7-way conference and connection of a second GAC2500 unit in "Daisy-chain" mode with a secondary speaker and another microphone of expansion. That is why its flexibility and mobility are the main differentiating qualities of this audio conference equipment.
In addition, being based on Android 4.4 supports Google App Store applications such as Google Hangouts, Skype, Skype for Business (Lync), Internet browser, Adoble Flash, Twitter, Facebook, Youtube, Google Calendar, import / export data from Mobile phone via Bluetooth, etc. API/ SDK available for advanced custom application development.
treeMT: Multi Tenant Platform para empresas y/o entornos residenciales
treeMT es una plataforma virtual, donde poder crear y gestionar tantas centralitas virtuales como se necesiten, (hasta un máximo de 2.000 extensiones entre todas). Avanzada 7 garantiza además el mantenimiento de la Plataforma Multi Tenant, incorporando nuevas features y correcciones en versiones sucesivas.
Otra de las ventajas de treeMT es que es una máquina virtual que le permite trabajar con su Hypervisor favorito sin tener que adaptar su hardware de trabajo.
treeMT está dirigido a operadores VoIP que necesitan ofrecer Centralitas Virtuales en una plataforma de gestión propia sin cambiar de operador (manteniendo sus carrier habituales).
Caracteristicas Principales:
Patton Licence upgrade 1 session adicional SN5300
The SN5300 connects to the IP-PBX or UC system in the Enterprise’s LAN and to an Internet telephony service provider (ITSP), creating a single conduit for multimedia components including voice, video, and data.
Patton Licence upgrade 1 session adicional SN5300 (up to 250 sessions in Patton SN5300)
License Mitel RFP V2 (up 100 Mitel DECT bases)
To enable users enjoy the advantages of using IP networks together with the DECT technology, Mitel has developed radio fixed parts (RFP) with IP interfaces for integrating DECT into IP networks. These radio fixed parts (RFP L32 IP and RFP L34 IP) are connected to the network like IP terminals. Voice is conveyed via VoIP to the radio fixed part and from the RFP to the air via DECT.
Therefore, employees can always be reached via their call numbers, regardless of whether they are in a branch of the company or at the head office. Using the same IP connections for data and telephony saves additional infrastructure and, thus, costs.
No matter the size and range of the IP network, a single Open Mobility Manager (OMM) is enough to manage all RFPs of the multi-cellular DECT network. This is installed on any of the RFPs by software. The OMM manages up to 256 RFPs and 512 handsets.
The base station RFP L42 WLAN allows the integration of mobile data transmission via WLAN in parallel using the same network. Central administration of the DECT and WLAN network available via a browser interface.
License Mitel RFP V2 (up 50 Mitel DECT bases)
License Mitel RFP V2 (up 10 Mitel DECT bases)
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:
Fax For Asterisk
Digium's Fax For Asterisk is a commercial facsimile (Fax) termination and origination solution designed to enhance the capabilities of Open Source and commercial Asterisk as well as Switchvox. Fax For Asterisk bundles a suite of user-friendly Asterisk applications and a licensed version of the industry's leading fax modem software from Commetrex. Fax For Asterisk provides low speed (14400bps) PSTN faxing via DAHDI-compatible telephony boards as well as VoIP faxing to T.38-compatible SIP endpoints and service providers. Licensed on a per-channel basis, Digium's Fax For Asterisk provides a complete, cost-effective, commercial fax solution for Asterisk users.
FREE 1st License:
Additionally, each open source or commercial Asterisk system is eligible to receive from Digium, a single channel of Fax For Asterisk, called Free Fax For Asterisk, for no cost. Free Fax For Asterisk is provided under license as-is, without technical support, and is available to all Asterisk users as a free, zero cost purchase from the Digium webstore. Only one channel of Free Fax For Asterisk may be used with an installation of Asterisk. If you require multiple channels of Fax capability or if you require Digium's technical support, you may purchase channels of Fax For Asterisk.
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.
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Plataforma Centralitas Virtuales para empresas y/o entornos residenciales
Patton Licence upgrade 1 session adicional SN5300 (up to 250 sessions ...
Activation license Mitel RFP V2 (up 100 Mitel DECT bases)
Activation license Mitel RFP V2 (up 50 Mitel DECT bases)
Activation license Mitel RFP V2 (up 20 Mitel DECT bases)
Activation license Mitel RFP V2 (up 10 Mitel DECT bases)
Digium echo cancel software license
Digium's Fax For Asterisk provides a complete commercial fax solution for ...
The most used voice codec
