Grandstream UCM6302A IP PBX 1500 Users 200 Concurrent Calls PoE+

PRODUCT OVERVIEW

Grandstream UCM6302A Unified Communication & Collaboration Solution

PRODUCT TYPE

IP Phones

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 HighAvailability 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 version 16 open source telephony operating system
• Analog Telephone FXS Ports: 2 RJ11 ports All ports have lifeline capability in case of power outage
• PSTN Line FXO Port: 2 RJ11 ports
• Network Interfaces: Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+
• NAT Router: Yes (supports router mode and switch mode)
• Peripheral Ports: 1 USB 2.0, 1 USB 3.0, 1 SD card interface
• LED Indicators: None

TECHNICAL SPECIFICATIONS

• Brand: Grandstream
• Model: UCM6302A
• Product Type: IP Phones
• 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 HighAvailability 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 version 16 open source telephony operating system
• Analog Telephone FXS Ports: 2 RJ11 ports All ports have lifeline capability in case of power outage
• PSTN Line FXO Port: 2 RJ11 ports
• Network Interfaces: Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+
• NAT Router: Yes (supports router mode and switch mode)
• Peripheral Ports: 1 USB 2.0, 1 USB 3.0, 1 SD card interface
• LED Indicators: None
• LCD Display: 320x240 color LCD with touch screen for Shortcut Keys and Scroll Bar
• Reset Switch: Yes, long press for factory reset and short press for reboot
• Voice-over-Packet Capabilities: LEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier grade Line Echo Cancellation, Dynamic Jitter Buffer, Modem detection & auto-switch to G.711, NetEQ, FEC 2.0, jitter resilience up to 50% audio packet loss
• Voice and Fax Codecs: Opus, G.711 A-law/U-law, G.722, G722.1 G722.1C, G.723.1 5.3K/6.3K, G.726-32, G.729A/B, iLBC, GSM; T.38
• QoS: Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
• API: Full API available for third-party platform and application integration
• Telephone Operating System: Based on Asterisk version 16
• DTMF Methods: In-band audio, RFC4733, and SIP INFO
• Provisioning Protocol & Plug-and-Play: Mass provisioning using AES encrypted XML configuration file, auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP Option 66 multicast SIP SUBSCRIBE mDNS), eventlist between local and remote trunk
• Network Protocols: TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP/HTTPS, PPPoE, STUN, SRTP, TLS, LDAP, HDLC, HDLC-ETH, PPP, Frame Relay (pending), IPv6, OpenVPN
• Disconnect Methods: Busy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect

INTENDED USE / APPLICATIONS

• Business voice communications
• Office, reception, and institutional environments
• Integration with compatible SIP/VoIP systems