Description
- 4 FXS and 4 FXO Port
- Up to 350 concurrent calls
- 4 Video Conference rooms and up to 40 parties with 1080p HD
- Voice Conference for up to 200 participants
- Zero configuration provisioning of Grandstream SIP endpoints
- Built-in conferencing & meetings platform; supports desktop, Wave app, and SIP endpoints
- Wave for Android & iOS 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 when using UCM RemoteConnect
- Enhanced reliability with support for Hot Standby High-Availability
- 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
Details | |
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Manufacturer's Product Code | UCM6304 |
Analog Telephone FXS Ports | 4 RJ11 Ports (All ports have lifeline capability in case of power outage |
PSTN Line FXO Ports | 4 RJ11 Ports (All ports have lifeline capability in case of power outage |
Network Interface | Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+ |
NAT Router | Yes (supports router mode and switch mode) |
Peripheral Ports | 2*USB 3.0, 1*SD card interface |
LED Indicators | Power 1/2, PoE, FXS, FXO, LAN, WAN, Heartbeat |
Reset Switch | Yes, long press for factory reset and short press for reboot |
Voice-over-Packet Capabilities | LEC with NLP Packetised 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 |
Video Codecs | H.264, H.263, H263+, H.265, VP8 |
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 |
Telephony Operating System | Based on Asterisk version 16 |
DTMF Methods | In-band audio, RFC2833, 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 |
Media Encryption | SRTP, TLS, HTTPS, SSH, 802.1X |
Universal Power Supply | Input: 100 ~ 240VAC, 50/60Hz; Output: DC 12V, 2A |
Dimensions | 485mm(L) x 187.2mm(W) x 46.2mm(H) |
Mounting | Wall mount & Desktop |
Caller ID | Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT |
Polarity Reversal/Wink | Yes, with enable/disable option upon call establishment and termination |
Call Center | Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/work-load, in-queue announcement |
Customisable Auto Attendant | Up to 5 layers of IVR (Interactive Voice Response) in multiple languages |
Maximum Call Capacity | Users: 2500; Concurrent calls: 350; Max concurrent SRTP calls: 150 |
Maximum Attendees of Conference Bridges | 4 Video Conference rooms and up to 40 parties with 1080p HD; Voice Conference: Up to 200 parties |
Wave Mobile App | Allows Android & iOS users to join UCM-hosted meetings & communicate with other users/solutions registered to the UCM6300 |
Call Features | Call park, call forward, call transfer, call waiting, caller ID, call record, call history, ringtone, IVR, music on hold, call routes, DID, DOD, DND, DISA, ring group, ring simultaneously, time schedule, PIN groups, call queue, pickup group, paging/intercom, voicemail, call wakeup, SCA, BLF, voicemail to email, fax to email, speed dial, call back, dial by name, emergency call, call follow me, blacklist/whitelist, voice conference, video conference, eventlist, feature codes, busy camp-on/ call completion, voice control |
Firmware Upgrade | supported by Grandstream Device Management System (GDMS), a zero-touch cloud provisioning and management system, It provides a centralised interface to provision, manage, monitor and troubleshoot Grandstream products |
Details | |
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Power Supply | 12VDC, 2A PSU |
Mounting | Rack mount & Desktop |
Ethernet Cable | Included |
Specifications
Details | |
---|---|
Manufacturer's Product Code | UCM6304 |
Analog Telephone FXS Ports | 4 RJ11 Ports (All ports have lifeline capability in case of power outage |
PSTN Line FXO Ports | 4 RJ11 Ports (All ports have lifeline capability in case of power outage |
Network Interface | Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+ |
NAT Router | Yes (supports router mode and switch mode) |
Peripheral Ports | 2*USB 3.0, 1*SD card interface |
LED Indicators | Power 1/2, PoE, FXS, FXO, LAN, WAN, Heartbeat |
Reset Switch | Yes, long press for factory reset and short press for reboot |
Voice-over-Packet Capabilities | LEC with NLP Packetised 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 |
Video Codecs | H.264, H.263, H263+, H.265, VP8 |
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 |
Telephony Operating System | Based on Asterisk version 16 |
DTMF Methods | In-band audio, RFC2833, 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 |
Media Encryption | SRTP, TLS, HTTPS, SSH, 802.1X |
Universal Power Supply | Input: 100 ~ 240VAC, 50/60Hz; Output: DC 12V, 2A |
Dimensions | 485mm(L) x 187.2mm(W) x 46.2mm(H) |
Mounting | Wall mount & Desktop |
Caller ID | Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT |
Polarity Reversal/Wink | Yes, with enable/disable option upon call establishment and termination |
Call Center | Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/work-load, in-queue announcement |
Customisable Auto Attendant | Up to 5 layers of IVR (Interactive Voice Response) in multiple languages |
Maximum Call Capacity | Users: 2500; Concurrent calls: 350; Max concurrent SRTP calls: 150 |
Maximum Attendees of Conference Bridges | 4 Video Conference rooms and up to 40 parties with 1080p HD; Voice Conference: Up to 200 parties |
Wave Mobile App | Allows Android & iOS users to join UCM-hosted meetings & communicate with other users/solutions registered to the UCM6300 |
Call Features | Call park, call forward, call transfer, call waiting, caller ID, call record, call history, ringtone, IVR, music on hold, call routes, DID, DOD, DND, DISA, ring group, ring simultaneously, time schedule, PIN groups, call queue, pickup group, paging/intercom, voicemail, call wakeup, SCA, BLF, voicemail to email, fax to email, speed dial, call back, dial by name, emergency call, call follow me, blacklist/whitelist, voice conference, video conference, eventlist, feature codes, busy camp-on/ call completion, voice control |
Firmware Upgrade | supported by Grandstream Device Management System (GDMS), a zero-touch cloud provisioning and management system, It provides a centralised interface to provision, manage, monitor and troubleshoot Grandstream products |
Expected Delivery Times
North Island - 2-6 Working Days
South Island - 3-10 Working Days