Cloud-based virtual PBX and accessories!

Call resources
Compatible with standard SIP phones.
– Telephone interface with user language: English,
Brazilian Portuguese, Spanish, French, and Italian.
– WebRTC Softphone.
– Windows, Mac, iOS and Android Soft Clients.
– Multi-device (5 simultaneously)
Multiple Bosses / Multiple Secretaries + Call List.
– Support for Occupied Indicator (BLF).
Free Seats
– Call back.
– Call flow control.
Call forwarding (busy, no call)
(unconditional response).
Call monitoring.
Call pickup.
– On-call parking.
– Call recording (record, search, listen).
– Call recording audit and download
files.
Call monitoring.
– Call spy.
– Call transfer (blind, assisted).
Call on hold.
– Call-back services.
Caller ID.
Call Inquiry (External and Internal)
– Connected number ID.
– 3-party teleconference.
– Conference bridge for 30 parties.
– DISA (Direct access to the internal system).
– DND (Do Not Disturb).
– Internal Direct Dialing Numbers (DIDs).
Find me / follow me.
– Hunting/touch groups.
– Direct line/Hotline.
Call waiting/music on hold.
– Outbound/inbound routes and route by identifier
calls.
– PIN sets.
– Dialing restriction per device.
– Restriction of incoming calls based on
Caller ID.
– Restriction on outgoing calls.
– Blacklist (PSTN, Local, Long Distance LDN).
– Blacklist (collect call, 0300, 0500, 0900).
– DID Blocking.
– Capture groups.
– Personal Assistance (IVR).
– Personal Recordings/Notes.
Remote Replacement.
Remote call pickup.
Redial
– Speed dial.
– Video call.
– Wake-up calls.
Up to 5 devices per user.
– Personal call log (billing/CDR).
– Personal extension settings.
– Personal and/or Multilevel IVR.
– UC (Chat, Video and Sharing).
– Transmission of alert messages in sets
SIP.

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Our Call Center!

Agent login/logout.
Free seats.
– Various reports.
Call monitoring.
– Call queues.
Call recording.
– Caller name lookup.
– CDR (record of details of
call).
– Conference (on-the-fly).
– Customer account codes.
– Hot table.
– Search groups.
Multi-level IVR (IVR)
Automated attendants.
– Satisfaction survey
TTS and ASR
Listen, officer.
Collection groups.
Advanced Routing
– Skill base routing
– Routing to Agent
Time-based routing.
– Queue priorities.
– VIP list for the queue.
– Touch group strategies.
– Communicating.
Whisper to the agent.
– Dialing/Outgoing Calls
Call queue priority
lost
Protocol ID

Reports:

– Incoming Calls
– Per Hour.
– Per Day.
– Per Week.
– Por Mês.
– Por Fila.
– Por Grupo.
– Por Agente.
– Chamadas Realizadas
– Per Hour.
– Per Day.
– Per Week.
– Por Mês.
– Por Fila.
– Por Grupo.
– Por Agente.
– Chamadas Atendidas
– Por Fila.
– Por Grupo.
– Por Agente.
– Transferidas.
– Chamadas Não Atendidas
– Por Fila.
– Por Grupo.
– Por Agente.
– Duração da Chamada
– Tempo total de chamada.
– Taxa total de tempo de
chamada.
– Tempo médio de chamada.
– Tempo de espera.
– Tempo médio de espera.
– Tempo máximo de espera.
– Tempo de Fila
– Tempo de espera previsto.
– Por Fila.
– Tempo de Toque
– Taxa de tempo de toque.
– Tempo máximo de toque.
– Agentes
– Disponibilidade.
– Pausas.
– Retiradas.
– Chamadas Atendidas.
– Chamadas Não Atendidas.
– Chamadas Desconectadas.
– Desconexão pelo Chamador.
– Transferidas.
– Nível de Serviço.
– Causa da desconexão.
– Taxa de abandono.
– Abandono pelo chamador.
– Relatórios personalizados.

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Recursos do Agente: 

– Interface gráfica do usuário.
– Login/Logout do agente.
Free seats.
– Pausa, Retirar.
– ID de fila, identificador de chamadas,
informações de espera.
– Transferência para
IVR/Fila/Agente/Supervisor.
– Multi-guia.
– Transferência com um clique / Clique para
discar.
– Posição do chamador na fila.

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Recursos de supervisão: 

– GUI da Web
– Estatísticas em tempo real
– Alarmes e relatórios
– Status da fila.
– Situação dos agentes
– Relatórios históricos
– Fluxo IVR

Micro and small businesses need solutions that bring more agility and economy in communication. For them, Intelbras offers Impacta 16, Impacta 40 and Impacta 68i plants. One for every need, with essential functions in the business routine.

• Intelligence and Economics: The switch directs calls to a specific preprogrammed operator, prioritizing cost savings when making calls to fixed, mobile, DDD, DDI and VoIP.
• Agility: The switch identifies who is calling and directs the call to any extension, group, or branch that is pre-registered to answer, based on part of the number (eg, DDD) or the entire number. Agility and assertiveness for you and your customers.
• Reliability: The Impacta line is ensured by strict quality control in development, production and after sales, the latter present throughout Brazil to ensure fast, effective and value-added service.

Features:

• Flexible numbering plan
• Follow me external
• Ticket
• Conference
• DISA (auto attendant)
• Caller ID
• Ethernet interface
• Executive Line
• Music waiting
• Least Cost Route
• Automatic row selection
• Hotel application solution
• Call Forwarding to Outside Numbers
• Person search function between smart terminals

More compact

With an optimized design, 66% smaller than Impacta line units, the UnniTI 2000 uses only 2 Us 19 ”rack height. In addition, the integrated RJ45 connectors make it easy to install on a wall, rack or shelf.

 Web manager with responsive interface

Manage UnniTI 2000 from different devices without sending / receiving schedules. Compatible with major browsers, it has a responsive interface and great usability on tablets and smartphones, and even displays automatic update alerts, without having to download from the site.

Features:

• Manages up to 120 IP extensions, 96 analog extensions and 60 IP lines.¹
• New layout with RJ45 connectors that streamline wall, rack or shelf installation.
• Responsive interface web manager: compatible with major browsers and optimized for tablet and smartphone use
• 2 options for hardware switch connection: external or internal
• Embedded IP CPU: Eliminates the need to purchase a SIP card
• Multilevel DISA: Auto Answer Programming, with menus and submenus.¹
• Web Configuration Interface: Manages UnniTI from different devices without sending / receiving schedules.
• GSM 2G / 3G² connections: the only centers in the market that allow up to 24 GSM 2G / 3G² connections.
• Integrated voicemail, no card required.¹
• Download network monitoring logs via web programmer.
• Monitors events via SNMP in real time.
• Integrated firewall: prevents attacks and ensures more security and control of network threats at the switch.
• Significant reduction in mobile costs by consulting number portability.
• Sending and receiving bulk SMS via email via GSM 2G / 3G² interfaces.
• Recording up to 8 calls simultaneously.¹
• Function Integrated person search, no plates required.

¹ Available through software licensing. ² Check availability of 3G technology in your region with mobile operator

Technical Specifications
» Tela colorida de 2,0” (240×320)
» Wi-Fi integrado de 2,4 GHz e 5 GHz
» Suporte para roaming IEEE802.11k/v/r
» Função de vibração para notificações e chamadas
» Até 16 teclas DSS
» Botão push-to-talk (PTT) configurável para discagem rápida
» Proteção contra quedas de 1,8 metros de altura
» Suporta criptografia WPA/WPA2-PSK e 802.1X EAP para segurança das comunicações
» Bluetooth 5.0 integrado, com suporte para fone de ouvido sem fio
» Até 9 horas em conversação e até 200 horas em standby

*Os dados de consumo de bateria são testados em laboratório e a situação de uso pode variar devido aos diferentes ambientes.

Interface de transmissão
Padrões de telefonia: DECT

Faixas de frequência:
1880 – 1900 MHz (Europa), 1920 – 1930 MHz (EUA),
1910 – 1920 MHz (Brasil), 1786 – 1792 MHz (Coreia),
1893 – 1906 MHz (Japão), 1880 – 1895 MHz (Taiwan)

Número de canais: 10 (Europa), 5 (EUA, Brasil ou Japão), 3 (Coreia), 8 (Taiwan)

Alcance em ambientes externos: até 400 metros (DP735/DP730/DP725) ou até 350
metros (DP722/DP720)
Alcance em ambientes internos: até 50 metros

Economia de energia: Modo ECO

Periféricos:
3 indicadores de LED: alimentação, rede, DECT
Botão de emparelhamento/paginação
Uma porta Ethernet de 10/100 Mbps com detecção automática e PoE integrado

Protocolo/padrões SIP:

RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP/RARP, ICMP, DNS (registro A, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP-MED, LDAP, TR-069, 802.1x, TLS, SRTP Codecs de voz G.711μ/a-law, G.723.1, G.729A/B, G.726-32, iLBC, G.722, OPUS, DTMF em banda e fora de banda (em áudio, RFC2833, SIP INFO), VAD, CNG, PLC, AJB

Recursos de telefonia:

Espera, transferência, encaminhamento, conferência de 3 vias, agenda para download (XML, LDAP, até 3000 entradas), chamada em espera, log de chamadas (até 750 registros), resposta automática, plano de discagem flexível, redundância de servidores e failover Qualidade de serviço Qualidade de serviço de camada 2 (802.1Q, 802.1P) e camada 3 (ToS, DiffServ, MPLS).

Safety Controle de acesso de usuário e administrador, autenticação baseada em MD5 e MD5-sess, arquivo de configuração com criptografia AES de 256 bits, TLS, SRTP, HTTPS, controle de acesso a mídia 802.1x, autenticação DECT e criptografia.

Vários idiomas:

Inglês, tcheco, alemão, espanhol, francês, hebraico, italiano, holandês, polonês, português, russo, turco, árabe, chinês simplificado, chinês tradicional, japonês, coreano, eslovaco, sérvio

Upgrade/provisionamento:

Upgrade de firm are via TFTP/HTTP/HTTPS, provisionamento em massa usando um arquivo de configuração XML com criptografia AES ou TR-069 ;

Várias contas SIP
Até dez (10) contas SIP distintas por sistema;
Cada aparelho pode ser mapeado para qualquer conta SIP;
Cada contas SIP pode ser mapeada para qualquer aparelho.

Ramais em grupo
Opções flexíveis quando vários aparelhos compartilham a mesma conta SIP:
Modo circular: todos os telefone tocam sequencialmente a partir do telefone ao lado
do que foi atendido por último
Modo linear: todos os telefones tocam sequencialmente na ordem predeterminada
Modo paralelo: todos os telefones tocam simultaneamente e, depois que um deles é
atendido, os telefones disponíveis podem fazer novas chamadas
Modo DID: Vários telefones podem ligar para a mesma conta, mas todas as chamadas
recebidas são encaminhadas para um único telefone.

Uso eficiente da energia e energia verde
Fonte de alimentação universal Entrada 100-240 VCA 50/60 Hz; Saída 5 VCC 1 A;
Conexão micro-USB;

PoE: IEEE 802.3af Classe 1, 0,44 W–3,84 W
Conteúdo da embalagem Unidade da base, fonte de alimentação universal; cabo Ethernet; Quick Start Guide (Manual de Início Rápido), declaração do GPL.

Dimensões (A x L x P) 140,31 x 64.98 x 105 mm
Peso Unidade da base: 140g; Fonte de alimentação universal: 50 g; Embalagem: 370g
Temperatura e umidade Operação: -10 a 50 ºC (14 a 122 ºF); Armazenamento: -20º a 60 ºC (-4 a 140 ºF); Umidade: 10% a 90% sem condensação

Conformidade FCC, CE, RCM, IC

Technical Specifications

» Tela colorida de 2,4 polegadas e 320×240 pixels.
»Support for up to 4 SIP accounts
»Audio conference with up to 6 participants
» HD audio for clearer calls
»Dual Gigabit ports, integrated PoE, 2 Gigabit ports for high-speed connection
»Supports G.722 and Opus audio codecs, HD audio on speakerphone and handset.
» Supports 6-way audio conferencing.
» Up to 12 DSS keys, each DSS key can be set as a line/BLF/speed dial key
»Ideal for use in a variety of business settings.
»Available for integration with security products, it can be easily customized into various security solutions to reduce investment.

THE HT812 é um avançado adaptador de telefone analógico (ATA) com 2 portas FXS e um roteador NAT
Gigabit integrado. Desenvolvido com a tecnologia de gateway/ATA SIP líder do setor da Grandstream,
com milhões de unidades implementadas no mundo inteiro, esse ATA avançado oferece qualidade de voz
excepcional em vários ambientes, forte criptografia com um certificado de segurança exclusivo para cada
unidade, provisionamento automatizado para implementação em massa e gerenciamento de dispositivos,
além de excelente desempenho de rede para uso no seu negócio, e, espalhar ramais utilizando seus aparelhos telefone analógicos existentes.

• Suporta 2 perfis SIP através de 2 portas FXS e duas portas Gigabit.
• Inclui um roteador NAT integrado que suporta velocidades de roteamento de até 100 Mbps.
• Tecnologia de criptografia de segurança TLS e SRTP para proteger chamadas e contas.
• As opções de provisionamento automatizado incluem arquivos de configuração TR-069 e XML.
• Suporta conferência de voz de 3 vias.
• O servidor SIP de failover alterna automaticamente para o servidor secundário caso o servidor principal perca a conexão.
• Suporta o protocolo T.38 Fax para criação de Fax sobre IP.
• Suporta uma ampla variedade de formatos de identificação de chamadas.
• Utilize com a série UCM de PABX IP da Grandstream para provisionamento com configuração zero.

More control

Available in 2 versions: with or without lock key. The keyed model is ideal for those who need more control, allowing keypad locking and preventing calls.

Table or wall position

Full can be used in table or wall positions. You choose the best option.

Features:

• T / P switch: Changes the signaling type of the phone from multi-frequency (T) to decadic (P) and vice versa (not available on Full and TC 20 models).
• Adjustable Ringer Volume:
  • Premium: H (high) and L (low).
  • TC 20 and TC 50 Premium: high, low and mute.
  • TC 500 and Full: high, medium and low.
• Ring: Changes the ring tone (available on Full and TC 50 Premium models).
• Mute: By holding down this key you can talk without the caller hearing your conversation.
• Redial / Redial: Redials the last number used (maximum 32 digits).
• Flash / R: Used to access services offered by the telephone company or PBX. Standard flash time: See label on product base.
• Pause / Pause: Adds a pause while dialing (not available on TC 20 and Full models).
• Mode / Tone: Changes the dialing mode temporarily from decadic to multifrequency (not available on TC 20, Full and TC 50 Premium models).
• * and #: used to access services offered by the telephone company or PABX (multifrequency signaling).
• Handset rest: If you need to leave a call waiting, simply place the handset just above the cradle. This feature is only used when the phone is wall mounted (not available on TC 20 model).
• Key model (optional): Locks the keypad (not available for TC 20 and TC 50 Premium models).
• T / PABX / P Switch: Available on TC 50 Premium model only. Selects audio between public line and PBX. If you use the telephone connected to a PBX, select the switch on the PBX option for better audio comfort.

Six SIP accounts and Gigabit network interface

The TIP 638V supports up to 6 SIP accounts, allowing calls to be made over the internet. The model is homologated following Anatel standards and requirements, ensuring greater compatibility with systems that support SIP 2.0. In addition, the Gigabit interface ensures that the full internet potential of the customer's network structure is utilized.

More efficiency

With IP technology, you can make internet phone calls, save on infrastructure, have more flexibility to change desktops, for example without having to change your extension or pull wires, and have many other advantages. By using the SIP 2.0 communication protocol, IP phones are compatible with leading open platforms in the market and with Intelbras IP telephone exchanges.

Video calls

The TIP 638V has 1.3 MP camera, which allows the user to make quality video calls, expanding the ways of communication.

HD audio perfection

Excellent HD audio quality, free from interference or noise, and with digital volume adjustment.

 Agility and versatility

USB, micro USB, SD card and HDMI ports enable communication with other devices, giving the device more versatility. In addition, the phone has an exclusive RJ9 headset jack.

Economy and flexibility in installation

With PoE support, data and power are transmitted over the same cable, saving on infrastructure investment. In addition, IP technology enables everything to be integrated on the same platform, reducing call costs as well.

Features:

• Video call camera
• Touch screen display
• Gigabit network interface
• Color display
• IEEE 802.3af PoE Support
• Six SIP Accounts
• SIP 2.0 Protocol
• 140 color-coded soft keys (green and red)
• USB, micro USB, SD card and HDMI ports
• 2 10/100/1000 Mbps network ports
• HD Voice
• 3 ways of answering (handsfree, headset and handset)
• Compatible with CHS 55 headset and EVC 1000 videoconferencing

Características Principais

Áudio HD no viva-voz (full duplex) e no monofone
8 Teclas DSS no visor principal
Até 35 teclas DSS
Suporte a conferência local de 6 vias inerente
Portas Gigabit Duplas com PoE integrado
Administração WEB e auto provisionamento
Design elegante e 2 modos de ajuste de ângulos de 45 e 55°
Compatível com a maioria dos PABX IPs SIP do mercado

Rede

⚫ Físico: Ethernet de 10/100/1000 Mbps, porta de rede dupla
para bypass de PC a rede LAN
⚫ IP Mode: IPv4/IPv6/IPv4&IPv6
⚫ Configuração de IP: Estático / DHCP / PPPoE
⚫ Controle de acesso à rede: 802.1x
⚫ VPN: OpenVPN
⚫ VLAN/LLDP/CDP/QoS
⚫ IEEE802.1Q, IEEE802.1P, ToS e DSCP
⚫ RTCP-XR (RFC 3611), VQ-RTCPXR (RFC 6035)
⚫ Suporte a autenticação com método MD5/MD5-sess
⚫ Suporte a ICE para comunicação ponto a ponto (Peer-to-Peer)
⚫ Suporte a chamadas via Proxy e chamadas SIP em modo peerto-peer

Protocolos

⚫ SIP2.0 sobre UDP/TCP/TLS
⚫ RFC 2474
⚫ RFC 2833
⚫ RFC 3261
⚫ RFC 3263
⚫ RFC 3264
⚫ RFC 3265
⚫ RFC 3550
⚫ RFC 4566
⚫ RFC 5359
⚫ RFC 6086
⚫ RTP/RTCP/SRTP
⚫ STUN
⚫ DHCP
⚫ CDP
⚫ LLDP-MED
⚫ Denial of Service (DoS)
⚫ PPPoE
⚫ 802.1x
⚫ Flooding
⚫ L2TP
⚫ OpenVPN
⚫ SNTP
⚫ FTP/TFTP/HTTP/HTTPS/TR-069
⚫ AES128 & AES256

More efficiency

With IP technology, you can make internet phone calls, save on infrastructure, have more flexibility to change desktops, for example without having to change your extension or pull wires, and have many other advantages. By using the SIP 2.0 communication protocol, IP phones are compatible with leading open platforms in the market and with Intelbras IP telephone exchanges.

 Discreet and ergonomic design

Specially designed for user comfort and usability, as its handset is ergonomic for better audio pickup on calls.

HD audio perfection

Through HD Voice technology it is possible to receive and make calls with higher voice quality than other terminals. Ideal for companies wishing to improve service quality through IP technology.

Make it easy with Graphic Display and Speakerphone

With the graphical display, you look up your call history and see who is calling. And with the Speakerphone function, you get the freedom to speak and jot down information at the same time.

More comfort to talk

Has connector and answer key for use with RJ9 headset.
Answer calls with just one tap and speed up your routine.

Economy and flexibility in installation

With PoE support, data and power are transmitted over the same cable, saving infrastructure investment. In addition, IP technology enables everything to be integrated on the same platform, reducing call costs and utilizing a single network point for both the IP terminal and the computer.

 Features:

• 132 × 64 pixel display
• Speakerphone, Mute and Redial
• Headset Jack (RJ9)
• 6 programmable two-color LED keys
• Bridge Function »Power over Ethernet (PoE) Support
• Ring indicator LED and missed calls
• DND (do not disturb), History, Phonebook, Conference, Hold, Flash, Mute, and Redial keys
• Voicemail and LED Headset Keys
• Navigation Keys
• Ringer, handset, speakerphone and headset volume adjustment
• 10 types of customizable doorbell
• Record of 10 calls originated, answered, missed and diverted
• Agenda for 1000 names / numbers, can be imported or exported as a file
• Voice over data prioritization (802.1p / Q, ToS / DSCP) ensuring superior audio quality
• Trilingual Menu: Portuguese, English and Spanish
• SIP 2.0 Protocol (RFC 3261)
• Remote Firmware and Configuration Update
• Automatic Bivolt Power Supply

More efficiency

With IP technology, you can make internet phone calls, save on infrastructure, have more flexibility to change desktops, for example without having to change your extension or pull wires, and have many other advantages. By using the SIP 2.0 communication protocol, IP phones are compatible with leading open platforms in the market and with Intelbras IP telephone exchanges.

HD audio perfection

Excellent HD audio quality, free from interference or noise, and with digital volume adjustment.

 Make it easy with Display blacklight and Speakerphone

With the backlit display, you look up your call history and see who's calling. And with the Speakerphone function, you get the freedom to speak and jot down information at the same time.

More comfort to talk

Has connector and answer key for use with RJ9 headset.
Answer calls with just one tap and speed up your routine.

Economy and flexibility in installation

With PoE support, data and power are transmitted over the same cable, saving infrastructure investment. In addition, IP technology enables everything to be integrated on the same platform, reducing call costs and utilizing a single network point for both the IP terminal and the computer.

 Features:

• Support for a SIP Account
• 128 × 32 pixel graphic display
• Dedicated conference key
• Menu key to access basic terminal settings
• Voicemail², mute, speakerphone, and headset answer keys all with LED signaling
• Audio, ringer, flash and redial volume keys
• 2 10/100 Mbps Ethernet ports
• Possibility of table or wall installation
• LED bell signaling
• QoS Support Voice Quality Assurance¹

¹ OpenSource platform compatible. ² Voicemail is a feature of the IP server.

Flagged keys

The TI 830 i has LED signaling keys for status of extensions, trunks and other functions.

To see better

The TI 830 i digital terminal features a 128 × 64 pixel graphical display with contrast adjustment, which provides users with more comfort.

 Features:

• Infrared sensor optical hook
• Speakerphone and quick extension access
• Headset with dedicated connection (RJ9)
• Keys for audio volume adjustments
• Display navigation keys
• 10 programmable keys with LED signaling
• Flash, Redial, Calls, Calendar, Erase and + key
• Programmable Call Pickup key
• Headset, handsfree, mute and recall key (LED)
• DISA setup key
• Keypad lock for terminal cleaning
• Alert beeps
• Registration and consultation of answered, missed and originated calls
• Update via flash drive through Central Connect⁺ and Modulare⁺
• 100 calendar positions
• 128 × 32 pixel graphic display
• Programmable Bells