Grandstream UCM6300 Audio Series IP PBXs
UCM6300 series of IP PBXs come in four different models, each delivering services for a particular business size. All of the models incorporate advanced voice features : All of the most innovative and popular telephony call features including call park, call forward, call transfer, call waiting, call record, call history, IVR, music on hold, direct inward dial, call back, and speed dials, and many more. chose your model
UCM series register hundreds of extensions in low price.
The UCM6300 Audio Series makes it simple for businesses to implement scalable integrated communication and collaboration solutions. IP PBXs serve as the central hub for all of an organization's essential telephony and conferencing services, such as phone calls, instant messaging, in-person meetings, web-based audio conferences, data analytics, mobility, and physical access.
Intercoms and various other mass communication systems. The UCM6300 Audio Series facilitates unified communication and collaboration across desktops, mobile devices, IP phones, and other SIP endpoints through the Wave App's integrated IM, voice/web conferencing, and support for up to 1,500 users. It enables the UCM RemoteConnect cloud service for remote users in order to provide the best-in-class hybrid platform, which combines the control of an on-premise IP PBX with the remote access and system-manageability of a cloud solution. The UCM6300 Audio series provides an all-inclusive business communication platform with a complete suite of mobility, security, instant messaging, voice conferencing, and collaboration technologies to large enterprises requiring a unified communications and collaboration solution.
UCM6300A
- Up to 250 users
- Up to 50 concurrent calls using the G.711 codec
- 3 concurrent conference bridges with up to 50 participants
UCM6302A
- 2 RJ11 FXS and two RJ11 FXO ports
- Up to 500 users
- Up to 75 concurrent calls using the G.711 codec
- 5 concurrent conference bridges with up to 75 participants
UCM6304A
- 4 RJ11 FXS and four RJ11 FXO ports
- Up to 1000 users
- Up to 150 concurrent calls using the G.711 codec
- 7 concurrent conference bridges with up to 120 participants
UCM6308A
- 8 RJ11 FXS and eight RJ11 FXO ports
- Up to 1500 users
- Up to 200 concurrent calls using the G.711 codec
- Up to 7 concurrent conference bridges with up to 150 participants
- Dual power supplies for redundancy
- Rack mountable
UCM 6301 | UCM 6302 | UCM 6304 | UCM 6308 | |
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Analog Telephone FXS Ports | 1 RJ11 Port | 2 RJ11 Ports | 4 RJ11 Ports | 8 RJ11 Ports |
All ports have lifeline capability in case of power outage, number of ports can be explained by peering with an FXS gateway | ||||
PSTN Line FXO Ports | 1 RJ11 Port | 2 RJ11 Ports | 4 RJ11 Ports | 8 RJ11 Ports |
All ports have lifeline capability in case of power outage, number of ports can be explained by peering with an FXS gateway | ||||
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 3.0, 1*SD card interface | 1*USB 2.0, 1*USB 3.0, 1*SD card interface | 2*USB 3.0, 1*SD card interface | |
LED Indicators | None | Power 1/2, FXS, FXO, LAN, WAN, Heartbeat | ||
LCD Display | 320x240 color LCD with touch screen for Shortcut Keys and Scroll Bar | 128x32 dot matrix graphic LCD with DOWN and OK buttons | ||
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 | |||
Telephony 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 | |||
Media Encryption | SRTP, TLS, HTTPS, SSH, 802.1X | |||
Universal Power Supply | Input: 100 ~ 240VAC, 50/60Hz; Output: DC 12V, 1.5A | 1x DC 12V Power Jack Input: 100~240VAC, 50/60Hz;Output: DC 12V, 2A | 2x DC 12V Power Jack Input: 100~240VAC, 50/60Hz;Output: DC 12V, 2A | |
Dimensions | 270mm(L) x 175mm(W) x 36mm(H) | 485mm(L) x 187.2mm(W) x 46.2mm(H) | ||
Weight | Unit Weight: 705g; Package Weight: 1131g | Unit Weight: 725g; Package Weight: 1221g | Unit Weight: 775g; Package Weight: 1621g | Unit Weight: 2538g; Package Weight: 3463g |
Temperature & Humidity | Operating: 32 - 113ºF / 0 ~ 45ºC, Humidity 10 - 90% (non-condensing) | |||
Storage: 14 - 140ºF / -10 ~ 60ºC, Humidity 10 - 90% (non-condensing) | ||||
Mounting | Wall mount & Desktop | Rack mount & Desktop | ||
Multi-Language Support | -Web UI: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish | |||
-Customizable IVR/voice prompts: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, | ||||
Russian, Swedish, Turkish, Hebrew, Arabic, Nederlands | ||||
-Customizable language pack to support any other languages | ||||
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/ workload, in-queue announcement | |||
Customizable Auto Attendant | Up to 5 layers of IVR (Interactive Voice Response) in multiple languages | |||
Maximum Call Capacity | Users: 250 | Users: 500 | Users: 1000 | Users: 1500 |
Concurrent calls (G.711): 50 | Concurrent calls (G.711): 75 | Concurrent calls (G.711): 150 | Concurrent calls (G.711): 200 | |
Max concurrent SRTP calls | Max concurrent SRTP calls | Max concurrent SRTP calls | Max concurrent SRTP calls | |
(G.711): 50 | (G.711): 75 | (G.711): 120 | (G.711): 150 | |
Maximum Attendees of Conference Bridges | 3 meeting rooms and up to 50 parties | 5 meeting rooms and up to 75 parties | 7 meeting rooms and up to 120 parties | 9 meeting rooms and up to 150 parties |
Wave App | Free; Available for desktop (Windows 10+, Mac OS 10+), web (Firefox and Chrome Browsers) and mobile (Android & iOS), allows users to join UCM-hosted meetings, communicate with other users/solutions and make/receive calls using SIP accounts registered to a UCM6300 Audio series IP PBX | |||
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 meeting, 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 centralized interface to provision, manage, monitor and troubleshoot Grandstream products | |||
Internet Protocol Standards | RFC 3261, RFC 3262, RFC 3263, RFC 3264, RFC 3515, RFC 3311, RFC 4028. RFC 2976, RFC 3842, RFC 3892, RFC 3428, RFC 4733, RFC 4566, RFC 2617, RFC 3856, RFC 3711, RFC 5245, RFC 5389, RFC 5766, RFC 6347, RFC 6455, RFC 8860, RFC 4734, RFC 3665, RFC 3323, RFC 3550 | |||
Compliance | FCC: Part 15 (CFR 47) Class B, Part 68 | |||
CE: EN 55032, EN 55035, EN 61000-3-2, EN 61000-3-3, EN 62368.1, ES 203 021, ITU-T K.21 | ||||
IC: ICES-003, CS-03 Part I Issue 9 | ||||
RCM: AS/NZS CISPR 32, AS/NZS 62368.1, AS/CA S002, AS/CA S003.1/.2 | ||||
Power adapter: UL 60950-1 or UL 62368-1 |
Data sheet
Codec | 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 |
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Dimensions | 270mm(L) x 175mm(W) x 36mm(H) |
Display | 320x240 color LCD with touch screen for Shortcut Keys and Scroll Bar |
DTMF | In-band audio, RFC4733, and SIP INFO |
RJ45 Ports | Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+ |
Mounting | Wall mount & Desktop |
NAT Router | Yes (supports router mode and switch mode) |
Operating humidity | Humidity 10 - 90% (non-condensing) |
Operating system | Based on Asterisk version 16 |
QoS | Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS |
Users | Supports up to 1500 users and up to 200 concurrent calls |
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