Cart (0)
Close
-
Unlock your business potential with Grandstream UCM6510
Rated 0 out of 5Features
-
(✓) Supports up to 2000 users and 200 SIP trunk accounts, up to 200 concurrent calls and up to 64 conference attendees.
-
(✓) 1GHz quad-core Cortex A9 processor.
-
(✓) 1GB DDR3 Ram, 32GB Flash.
-
(✓) 1 Integrated T1/E1/J1 interface, 2 PSTN trunk FXO ports, 2 analog telephone/Fax FXS ports with lifeline capability.
-
(✓) Gigabit network ports with Integrates PoE, USB, SD card, integrated NAT router.
-
(✓) Comprehensive security protection using SRTP, TLS and HTTPS with hardware encryption accelerator.
-
(✓) Quickly setup and provision Grandstream endpoints using the Auto-Discovery and Zero Config feature within the product’s web user interface.
-
-
UCM6308
Rated 0 out of 5Features
-
(✓) Supports 3000 users and 450 concurrent calls and Max 300 concurrent SRTP calls.
-
(✓) Zero configuration provisioning of Grandstream SIP endpoints.
-
(✓) 10 Video Conference rooms and up to 80 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus) Voice Conference: Up to 300 parties.
-
(✓) Built-in conferencing & meetings platform; supports desktop, and SIP endpoints.
-
(✓) Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & solutions.
-
(✓) 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+/H.265/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.
-
-
UCM6308A
Rated 0 out of 5Features
-
(✓) Supports 1500 users and 200 concurrent calls and Max 150 concurrent SRTP calls.
-
(✓) Zero configuration provisioning of Grandstream SIP endpoints.
-
(✓) 9 meeting rooms and up to 150 parties.
-
(✓) Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints.
-
(✓) 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.
-
-
UCM6304
Rated 0 out of 5Features
-
(✓) Supports 2000 users and 300 concurrent calls and Max 200 concurrent SRTP calls.
-
(✓) Zero configuration provisioning of Grandstream SIP endpoints.
-
(✓) 8 Video Conference rooms and up to 60 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus) Voice Conference: Up to 200 parties .
-
(✓) Built-in conferencing & meetings platform; supports desktop, and SIP endpoints.
-
(✓) Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & solutions.
-
(✓) 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+/H.265/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.
-
-
UCM6304A
Rated 0 out of 5Features
-
(✓) Supports 1000 users and 150 concurrent calls and Max 120 concurrent SRTP calls.
-
(✓) Zero configuration provisioning of Grandstream SIP endpoints.
-
(✓) 7 meeting rooms and up to 120 parties.
-
(✓) Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints.
-
(✓) 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.
-
-
UCM6302
Rated 0 out of 5Features
-
(✓) Supports 1000 users and 150 concurrent calls and Max 100 concurrent SRTP calls.
-
(✓) Zero configuration provisioning of Grandstream SIP endpoints.
-
(✓) 6 Video Conference rooms and up to 30 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus) Voice Conference: Up to 150 parties .
-
(✓) Built-in conferencing & meetings platform; supports desktop, and SIP endpoints.
-
(✓) Wave for Android, iOS, Chrome and Firefox browsers allows communication with all UCM6300 users & solutions.
-
(✓) 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+/H.265/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.
-
-
UCM6302A
Rated 0 out of 5Features
-
(✓) Supports 500 users and 75 concurrent calls and Max 75 concurrent SRTP calls.
-
(✓) Zero configuration provisioning of Grandstream SIP endpoints.
-
(✓) 5 meeting rooms and up to 75 parties.
-
(✓) Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints.
-
(✓) 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.
-
-
UCM6208
Rated 0 out of 5Features
-
(✓) UCM6208 supports up to 800 users and 100 concurrent calls
-
(✓) Auto Discovery and Zero Configuration of Grandstream SIP endpoints
-
(✓) Integrated 8 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability and up to 200 SIP trunk accounts
-
(✓) 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 encryption
-
(✓) Supports any SIP video endpoint that uses the H.264, H.263 or H.263+ codecs
-
-
UCM6204
Rated 0 out of 5Features
-
(✓) UCM6204 support up to 500 users and 75 concurrent calls
-
(✓) Auto Discovery and Zero Configuration of Grandstream SIP endpoints
-
(✓) Integrated 4 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability and up to 200 SIP trunk accounts
-
(✓) 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 encryption
-
(✓) Supports any SIP video endpoint that uses the H.264, H.263 or H.263+ codecs
-
-
UCM6202
Rated 0 out of 5Features
-
(✓) UCM6202 support up to 500 users and 50 concurrent calls
-
(✓) Auto Discovery and Zero Configuration of Grandstream SIP endpoints
-
(✓) Integrated 2 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability and up to 200 SIP trunk accounts
-
(✓) 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 encryption
-
(✓) Supports any SIP video endpoint that uses the H.264, H.263 or H.263+ codecs
-