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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.
The SBC License for Mediatrix Sentinel 100 (10–99 sessions) enables between 10 and 99 simultaneous SBC sessions on your Sentinel 100 unit. This scalable licensing option is designed for medium to large enterprises requiring robust VoIP edge security and session management.
Key Features:
Scalability: Expand SBC capacity as your network grows.
Security: Add advanced SIP protection and encrypted media streams.
Seamless integration: Fully compatible with Sentinel 100 platforms.
Easy activation: No additional hardware required.
This license is essential for organizations needing reliable and adaptable SBC functionality across multiple concurrent VoIP calls.
Check availability and contact Avanzada7 today at sales@avanzada7.com
The Mediatrix C7 Series offers a series of gateways that combine FXS and FXO interfaces to integrate multiple applications into a single platform. These VoIP Gateways offer excellent value for money and are ideal for connecting small and medium-sized networks to an IP network.
Models C710, C711, C730, C731 and C733 are currently available. Depending on the model it can present from 4 to 8 telephone ports and connect up to 8 analog phones, modems, fax or up to 8 RTC lines or trunks from a PBX to the IP network.
One of Mediatrix's flagship products is the Sentinel, an integrative product that has the capability of a customizable Session Border Controller (SBC) and Gateway. It is ideal for medium and large enterprises that need VoIP deployments with flexible architecture, including fault tolerance, SIP normalization and demarcation point.
The Base Station of our Sentinel initially presents:
Thanks to these 8 slots we will be able to customize our Sentinel as we want, combining the cards and licenses that best suit our needs.
MODULES
SBC Licenses
The Advanced Recovery module from Sangoma provides a high-availability solution for FreePBX and PBXact systems, ensuring seamless service continuity in case of primary server failure.
It allows administrators to configure a backup server that mirrors the primary setup. If a failure is detected, the system automatically switches to the secondary server, minimizing downtime and protecting critical communications.
Key features include:
User-friendly GUI for configuring and synchronizing primary and secondary servers.
Continuous monitoring and automatic failover response.
Integration with Endpoint Manager to update Sangoma S- and D-series phones during switchover.
Support for custom third-party scripts triggered during failover.
This solution is ideal for business environments requiring robust uptime and zero-interruption communications.
Check availability and contact to Avanzada7 at sales@avanzada7.com
The Sangoma Call Accounting Module is a commercial add-on for FreePBX and PBXact systems that enables accurate allocation of call-related costs across your organization. With this module, administrators can track call expenses by user, user group, or trunk, offering complete transparency and cost control in VoIP environments.
Custom call cost assignment for inbound and outbound calls based on trunks, users, or groups.
Multiple rate deck configuration to associate different cost structures per trunk.
Advanced dialing pattern management, with bulk import/export options.
Comprehensive reporting, providing detailed insights into call usage and expenses.
This module is compatible with FreePBX 14+ and PBXact 14+, and requires an annual license.
For pricing or to check availability, contact our Sales Team at sales@avanzada7.com
The UCM RemoteConnect Business Plan by Grandstream is the ultimate solution for businesses seeking to connect and manage up to 200 remote users or devices simultaneously. This plan supports up to 32 concurrent voice, video calls, or conference sessions and includes 5 GB of cloud storage. The innovative Wave app enables seamless collaboration from desktop, web, or mobile platforms.
With always-on automated NAT firewall traversal and integration with the Grandstream Device Management System (GDMS), this plan ensures secure, reliable connections. It also offers advanced system monitoring, real-time alerts, call quality analytics, and customization options for company logos and domains.
Elevate your remote team's productivity and security with this robust and scalable platform. Contact us in Avanzada7 for pricing or download the datasheet for more details.
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.
The Enterprise Plan of UCM RemoteConnect by Grandstream is designed for organizations requiring robust, scalable, and secure communication solutions for remote collaboration. Supporting up to 400 registered users or devices and 64 simultaneous sessions of voice calls, video meetings, or conferences, this plan provides the highest capacity within the UCM RemoteConnect suite.
It includes 10GB of cloud storage, secure NAT firewall traversal, and real-time system monitoring and alerts to ensure seamless and secure connections. Advanced remote administration, integration with WebRTC trunking, custom logo and top-level domain (FQDN), and detailed call quality analytics elevate the user experience. Compatible with the Wave app for desktop, web, and mobile, the Enterprise Plan is ideal for businesses seeking premium communication capabilities.
Contact our Avanzada7 Sales Team today or download the full datasheet for more details.
The UCM RemoteConnect Plus Plan by Grandstream is designed to provide businesses with secure and scalable solutions for remote work. This plan allows up to 50 registered users or devices and supports 8 concurrent remote sessions for voice calls, video calls, or conferences. Features include 1 GB of cloud storage, integration with the Wave app (available on mobile, desktop, and web), and an always-on NAT firewall traversal for enhanced security. With real-time monitoring, alerts, and IT-friendly management tools, this plan ensures reliability with a 99.9% uptime on AWS. Additional capabilities such as comprehensive remote administration, API and SDK support, and WebRTC trunk integration make it a robust choice for modern enterprises. Start optimizing your remote work today! Learn more.
Grandstream UCM RemoteConnect Pro is a cloud-based service designed to provide secure and efficient remote collaboration. Fully compatible with the UCM6300 series, this plan includes unlimited calls and meetings, 2 GB of cloud storage, comprehensive remote administration, and always-on NAT firewall traversal. Additionally, it offers real-time monitoring, alerts, and support for up to 100 registered remote users or devices.
Integrated with the Grandstream Device Management System (GDMS) and featuring the Wave app for desktop, web, and mobile, UCM RemoteConnect Pro ensures a seamless user experience. It’s the ideal solution for organizations seeking to enhance mobility and security for their teams.
Key Benefits:
Discover how UCM RemoteConnect Pro can transform your business communications. Contact Avanzada7 Sales Team or download the datasheet now!
The 1-year renewal license for the FreePBX Commercial Endpoint Manager (EPM) by Sangoma ensures that your system remains current and supported. With this renewal, you'll continue receiving firmware updates and new device compatibility, along with access to official technical support.
Maintain centralized provisioning and configuration for over 300 VoIP devices from top vendors using the same intuitive graphical interface.
This renewal is essential for FreePBX deployments that rely on EPM for efficient endpoint management.
Check availability and request more info now at Avanzada7: sales@avanzada7.com
The SBC License for Mediatrix Sentinel 100 (1–9 sessions) enables between 1 and 9 simultaneous SBC sessions on your Sentinel 100 device. This scalable licensing option is ideal for small to medium-sized businesses that need secure and reliable VoIP session control.
Scalable licensing: Add sessions as your needs grow.
Security: Enhanced SIP session protection and media encryption.
Seamless integration: Designed specifically for the Sentinel 100 platform.
This license is a cost-effective way to expand SBC functionality on Mediatrix Sentinel 100 for secure and flexible SIP traffic management.
The Call Recording Reports module by Sangoma is a commercial add-on for FreePBX and PBXact systems, enabling advanced management and archival of call recordings. With this tool, administrators can:
Search recordings by date, call type, source, destination, duration, and time.
Listen to call recordings directly from the web interface.
Download individual recordings or batch them into compressed (.tgz) files.
Automatically archive recordings monthly to reduce storage space.
Set up email reminders to back up archive files.
Automatically delete older recordings based on retention policies.
This module is ideal for businesses requiring call quality control, training, or legal compliance related to call recordings.
For more details or personalized assistance contact our sales team.
Check availability and request more information now at sales@avanzada7.com
License Mitel RFP V2 (up 10 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 100 Mitel DECT bases)
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)
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:
The FreePBX Endpoint Manager (EPM) commercial module by Sangoma streamlines provisioning and configuration of over 300 VoIP devices—desk phones, wireless units, intercoms, paging systems, gateways, and more—from major brands like Aastra, Algo, Audiocodes, Cisco, Digium, Grandstream, Mitel, Panasonic, Polycom, Sangoma, Snom, Xorcom, and Yealink.
EPM features an intuitive GUI that allows you to:
Create multiple templates per vendor to generate configuration files.
Assign extensions to MAC addresses via extension mapping.
Manage firmware and images per device or template.
Set global settings such as internal/external PBX IPs.
Modify base files for advanced control without being overwritten by updates.
It also includes the UCP for EPM module, empowering users to customize their phone button layouts directly from the User Control Panel (UCP), without altering system-level templates.
Sangoma IP phones built for FreePBX include a lifetime EPM license—no additional purchase needed.
Download the datasheet or contact our sales team for a full list of supported devices and technical specifications.
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.
Use our comparator and choose the correct product
Sangoma renewal FreePBX Zulu 1 usuario (1 año)
Enable up to 99 SBC sessions for secure and efficient VoIP ...
Sangoma Renewal FreePBX Zulu 20 usuarios (1 año)
Automatic failover to keep your VoIP system always operational.
Sangoma FreePBX Softphones 1-User 1-Year License
Assign and monitor call costs by user, group, or trunk efficiently ...
Advanced features for up to 200 users with centralized management and ...
Support for up to 400 users with secure and centralized remote ...
Support up to 50 users with 8 simultaneous sessions for seamless ...
Advanced collaboration and security tools in the cloud for UCM6300 series.
Extend your EPM license for 1 year and ensure uninterrupted updates ...
Unlock up to 9 SBC sessions for efficient and secure VoIP ...
Efficiently manage and archive your call recordings with powerful filters and ...
Activation license Mitel RFP V2 (up 10 Mitel DECT bases)
Activation license Mitel RFP V2 (up 20 Mitel DECT bases)
Activation license Mitel RFP V2 (up 50 Mitel DECT bases)
Activation license Mitel RFP V2 (up 100 Mitel DECT bases)
Patton Licence upgrade 1 session adicional SN5300 (up to 250 sessions ...
Plataforma Centralitas Virtuales para empresas y/o entornos residenciales
Sangoma FreePBX Zulu 20 usuarios (1 año)
Automate the configuration of 300+ VoIP devices and enable users to ...
Digium echo cancel software license
Digium's Fax For Asterisk provides a complete commercial fax solution for ...
The most used voice codec
