Grandstream UCM6300A Audio Series IPPBX

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$463.00 +GST
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20 In Stock
1 - 4$463.00
5 - 9$428.28
10 - 19$416.70
Grandstream UCM6300A Asterisk 16, 250 users/50 concurrent calls, NO FXO/FXS PORTS, NO VIDEO CAPABILITY
Package Contents
The Grandstream UCM6300 Audio series 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. The UCM6300 Audio Series supports up to 1500 users and includes a built-in instant messaging (IM), voice/web conferencing platform, and the free Wave App that allows users to communicate and collaborate from desktops, mobile devices, IP phones, and other SIP endpoints. It supports UCM RemoteConnect cloud service for remote users to offer a best-in-class hybrid platform that combines the control of an on-premise IP PBX with the remote access and system manageability of a cloud solution. By offering a high-end unified communications and collaboration solution packed with a suite of mobility, security, instant messaging, voice conferencing and collaboration tools, the UCM6300 Audio series provides a powerful business communication platform for any organization.

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  • 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
Manufacturer's Product CodeUCM6300A
Analog Telephone FXS PortsNone
PSTN Line FXO PortsNone
Network InterfacesThree self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+
NAT RouterYes (supports router mode and switch mode)
Peripheral Ports1*USB 3.0, 1*SD card interface
LED IndicatorsNone
LCD Display320x240 color LCD with touch screen for Shortcut Keys and Scroll Bar
Reset SwitchYes, long press for factory reset and short press for reboot
Voice-over-Packet CapabilitiesLEC 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 CodecsOpus, 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
QoSLayer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
APIFull API available for third-party platform and application integration
Telephony Operating SystemBased on Asterisk version 16
DTMF MethodsIn-band audio, RFC4733, and SIP INFO
Provisioning Protocol & Plug-and-PlayMass 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
Disconnect MethodsBusy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect
Media EncryptionSRTP, TLS, HTTPS, SSH, 802.1X
Universal Power SupplyInput: 100 ~ 240VAC, 50/60Hz; Output: DC 12V, 1.5A
Dimensions270mm(L) x 175mm(W) x 36mm(H)
WeightUnit Weight: 705g; Package Weight: 1131g
Temperature & HumidityOperating: 32 - 113ºF / 0 ~ 45ºC, Humidity 10 - 90% (non-condensing)Storage: 14 - 140ºF / -10 ~ 60ºC, Humidity 10 - 90% (non-condensing)
MountingWall mount & Desktop
Caller IDBellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT
Polarity Reversal/WinkYes, with enable/disable option upon call establishment and termination
Call CenterMultiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/ workload, in-queue announcement
Customizable Auto AttendantUp to 5 layers of IVR (Interactive Voice Response) in multiple languages
Maximum Call CapacityUsers: 250; Concurrent calls (G.711): 50 Max; Concurrent SRTP calls (G.711): 50
Maximum Attendees of Conference Bridges3 meeting rooms and up to 50 parties
Call FeaturesCall 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 UpgradeSupported 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
PoE (802.3af/at)

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