Grandstream

• UCM6202 supports up to 500 users/extension and 50 concurrent calls
• Integrated 2 FXO ports for PSTN trunk and 2 FXS ports for analog telephone with lifeline capability
• up to 50 SIP trunk accounts
• up to 3 conference bridges with password-protected
• up to 25 attendees per bridge simultaneous PSTN or IP participants
• 2(two) Gigabit network ports with Integrates PoE, USB, SD card
• Supports up to a 5-level IVR (Interactive Voice Response)
• Built-in call recordings server; recordings accessible via web user interface
• Built-in Call Detail Records (CDR) for tracking phone usage by line, date, etc.
• Supports multi-language auto-attendant and call queue to efficiently handle incoming calls
• Strongest possible security protection using SRTP, TLS and HTTPS encyrption
• Supports any SIP video endpoint that uses the H.264, H.263 or H.263+ codecs
• Auto Discovery and Zero Configuration of Grandstream SIP endpoints

Grandstream UCM6304 Key Features

• Built in 4 FXO ports (PSTN Line) and 4 FXS ports (Analog Device); All ports have lifeline capability in case of power outage.
• Up to 15 Simultaneous Video Conference Rooms, up to 200 simultaneous participants in all room combined, up to 9 video feeds in all conference room
• Voice Conference : Up to 200 parties (G.711)
• Supports up to 2000 users and up to 300 concurrent calls
• Up to 200 SIP trunk accounts
• Zero configuration provisioning of Grandstream SIP endpoints
• Built-in conferencing & meetings platform; supports desktop, Wave app, and SIP endpoints
• Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & solutions
• API available for third-party integrations, including CRM and PMS platforms
• Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
• Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
• Automated NAT firewall traversal service facilitates secure remote connections
• Supports Full-Band Opus voice codec and H.264/H.263/ H.263+/VP8 video codec, jitter resilience up to 50% packet loss
• Compatible with GDMS for cloud setup, management and monitoring
• Based on Asterisk* version 16 open source telephony operating system

Grandstream UCM6308 Key Features

• Built-in 8 FXO ports (PSTN Line) and 8 FXS ports (Analog Device); All ports have lifeline capability in case of power outage.
• Up to 25 Simultaneous Video Conference Rooms, up to 300 simultaneous participants in all room combined, up to 9 video feeds in all conference room
• Voice Conference : Up to 300 parties (G.711)
• Supports up to 3000 users and up to 450 concurrent calls
• Up to 200 SIP trunk accounts
• Zero configuration provisioning of Grandstream SIP endpoints
• Built-in conferencing & meetings platform; supports desktop, Wave app, and SIP endpoints
• Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & solutions
• API available for third-party integrations, including CRM and PMS platforms
• Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
• Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
• Automated NAT firewall traversal service facilitates secure remote connections
• Supports Full-Band Opus voice codec and H.264/H.263/ H.263+/VP8 video codec, jitter resilience up to 50% packet loss
• Compatible with GDMS for cloud setup, management and monitoring
• Based on Asterisk* version 16 open source telephony operating system

Grandstream UCM6300A Audio Series IP PBX

The Grandstream UCM6300A allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifies fundamental business communications needs, including voice, instant messaging (IM), voice meetings, audio web meetings, data, analytics, mobility, facility access,intercoms and more.

Grandstream UCM6300A Key Features

• Supports up to 250 users and up to 50 concurrent calls
• Zero configuration provisioning of Grandstream SIP endpoints
• Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints
• Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices
• API available for third-party integrations, including CRM and PMS platforms
• Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
• Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
• Automated NAT firewall traversal service facilitates secure remote connections
• Enhanced reliability with support for Hot Standby High-Availability and local dual deployment
• Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
• Compatible with GDMS for cloud setup, management, and monitoring
• Based on Asterisk* version 16 open source telephony operating system

Grandstream UCM6302A Key Features

• Supports up to 500 users and up to 75 concurrent calls
• Zero configuration provisioning of Grandstream SIP endpoints
• Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints
• Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices
• API available for third-party integrations, including CRM and PMS platforms
• Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
• Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
• Automated NAT firewall traversal service facilitates secure remote connections
• Enhanced reliability with support for Hot Standby High-Availability and local dual deployment
• Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
• Compatible with GDMS for cloud setup, management, and monitoring
• Based on Asterisk* version 16 open source telephony operating system

Grandstream UCM6304A Key Features

• Supports up to 1000 users and up to 150 concurrent calls
• Zero configuration provisioning of Grandstream SIP endpoints
• Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints
• Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices
• API available for third-party integrations, including CRM and PMS platforms
• Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
• Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
• Automated NAT firewall traversal service facilitates secure remote connections
• Enhanced reliability with support for Hot Standby High-Availability and local dual deployment
• Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
• Compatible with GDMS for cloud setup, management, and monitoring
• Based on Asterisk* version 16 open source telephony operating system

Grandstream UCM6308A Key Features

• Supports up to 1500 users and up to 200 concurrent calls
• Zero configuration provisioning of Grandstream SIP endpoints
• Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints
• Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices
• API available for third-party integrations, including CRM and PMS platforms
• Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
• Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
• Automated NAT firewall traversal service facilitates secure remote connections
• Enhanced reliability with support for Hot Standby High-Availability and local dual deployment
• Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
• Compatible with GDMS for cloud setup, management, and monitoring
• Based on Asterisk* version 16 open source telephony operating system