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GS-UCM6302

Grandstream VoIP PBX 2x FXS and 2x FXO

Grandstream - Grandstream VoIP PBX 2x FXS and 2x FXO, UCM6302, 150 simultaneous calls, 1000 SIP clients, LCD Display, Opus supported, 2 RJ11 ports, Three self-adaptive Gigabit ports, 1x USB 2.0, 1x USB 3.0, 1x SD card Interface, 320 x 240 Colour LCD with Touch Screen
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Prices, images and descriptions are for reference purposes only & may change without prior notice. PLEASE RELY ON ABOVE DESCRIPTIONS AND NOT ON IMAGES. GS-UCM6302 Grandstream VoIP PBX 2x FXS and 2x FXO is available from Laptop Direct and it is sold as new from our VoIP handsets phones range of products. For reference purposes, the price of R6460 is valid 2026-09-25, and while stocks last, may be valid until . The price displayed is a guide for reference purposes only, and may change without prior notice. Reference prices are an invitation for potential buyers to make an offer to purchase, which we may reject without reason. A quote with final pricing is sent before a purchase can be made. T&C's apply. E&OE. Laptop Direct. You are looking at products found in the VoIP handsets phones category.
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Specification for the entire series

Important information: Kindly note that specifications differ between models. Features, options and accessories displayed below this paragraph are not included with this product unless it has been specified in the above description.


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GS-UCM6302

Grandstream VoIP PBX 2x FXS and 2x FXO

Grandstream VoIP PBX 2x FXS and 2x FXO, UCM6302, 150 simultaneous calls, 1000 SIP clients, LCD Display, Opus supported, 2 RJ11 ports, Three self-adaptive Gigabit ports, 1x USB 2.0, 1x USB 3.0, 1x SD card Interface, 320 x 240 Colour LCD with Touch Screen

Overview:

The UCM6300 series allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifies all business communication on one centralized network, including voice, video calling, video conferencing, video surveillance, web meetings, data, analytics, mobility, facility access, intercoms and more. The UCM6300 series supports up to 3000 users and includes a built-in web meetings and video conferencing solution that allows employees to connect from the desktop, mobile, GVC series devices and IP phones. It can be paired with the UCM6300 ecosystem to offer a hybrid platform that combines the control of an on-premise IP PBX with the remote access of a cloud solution. The UCM6300 ecosystem consists of the Wave app for web and mobile, which provides a hub for collaborating remotely, and UCM RemoteConnect, a cloud NAT traversal service for ensuring secure remote connections. The UCM6300 series also offers cloud setup and management through GDMS and an API for integration with third-party platforms. By offering a high-end unified communications and collaboration solution packed with a suite of mobility, security, meeting and collaboration tools, the UCM6300 series provides a powerful platform for any organisation.

Features:

  • Supports up to 1000 users
  • Supports up to 150 concurrent calls
  • Zero configuration provisioning of Grandstream SIP endpoints
  • Built-in conferencing & meetings platform; supports desktop, Wave app, and SIP endpoints
  • Wave for Android, iOS, Chrome and Firefox browsers 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
  • 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

Specifications:

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Specificationsspacer
Product CodeGS-UCM6302spacer
Analog Telephone FXS Ports2 RJ11 Ports
All ports have lifeline capability in case of power outage
PSTN Line FXO Ports2 RJ11 Ports
All ports have lifeline capability in case of power outage
Network InterfacesThree self-adaptive Gigabit ports (switched, routed or dual card mode) with PoE+
NAT RouterYes (supports router mode and switch mode)
Peripheral Ports1*USB 2.0, 1*USB 3.0, 1*SD card interface
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
Video CodecsH.264, H.263, H263+, H.265, VP8
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, RFC2833, 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
Network ProtocolsTCP/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 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
Dimensions270mm(L) x 175mm(W) x 36mm(H)
WeightUnit Weight: 705g
Package Weight: 1190g
Temperature & HumidityOperating: 32 - 113ºF / 0 ~ 45ºC, Humidity 10 - 90% (non-condensing)
Storage: 14 - 140ºF / -10 ~ 60ºC, Humidity 10 - 90% (non-condensing)
MountingWall 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 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/ work-load, in-queue announcement
Customizable Auto AttendantUp to 5 layers of IVR (Interactive Voice Response) in multiple languages
Maximum Call CapacityUsers: 1000
Concurrent calls: 200
Max concurrent SRTP calls: 120
Maximum Attendees of Conference BridgesVideo Conference: Up tp 20
Voice Conference: Up to 80
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 conference, video conference, eventlist, feature codes, busy camp-on/ call completion, voice control


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