Skip to Content
  • Delivery in 4–10 working days across Saudi Arabia
  • A tax invoice with every order
  • Technical support before and after you buy
Request a project solution
  1. All products
  2. IP Telephony & Communications
  3. IP PBX
  4. Grandstream UCM6302A Audio IP PBX, 500 Users, 75 Calls, 2 FXS + 2 FXO
  5. IP PBX
[UCM6302A] Grandstream UCM6302A Audio IP PBX, 500 Users, 75 Calls, 2 FXS + 2 FXO

Grandstream UCM6302A Audio IP PBX, 500 Users, 75 Calls, 2 FXS + 2 FXO

Part number UCM6302A
  • Supports up to 500 users and up to 75 concurrent calls (G.711)
  • 2 FXS and 2 FXO RJ11 ports; all ports have lifeline capability in case of power outage
  • Zero configuration provisioning of Grandstream SIP endpoints
  • Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform with access from computers, mobile devices and SIP endpoints
  • Free Wave App allows easy voice & IM communications using desktops, Web, and Android/iOS devices
  • API available for third-party integrations, including CRM and PMS platforms
  • Three Gigabit auto-sensing RJ45 ports with integrated PoE+ and NAT router; automated NAT firewall traversal (UCM RemoteConnect) for secure remote connections
  • Secure boot, unique certificate and random default password; Hot Standby High-Availability and local dual deployment
SIP PoE Gigabit Color Screen Touch Screen FXO FXS Grandstream IP PBX
Price
⃁ 1,357.03 ⃁ 1,357.03 ⃁ 1,766.98 Save 23%
⃁ 1,357.03 ⃁ 1,766.98
(Tax excluded)
Ask on WhatsApp Quote for installers and contractors
  • Delivery within 4–10 business days across Saudi Arabia
  • Warranty according to the manufacturer
  • Tax invoice with every order
  • Technical support before and after purchase

Not sure this model fits your setup? Read the buying guide

Official datasheetGrandstream_UCM6302A_Datasheet.pdf Download

Overview of Grandstream UCM6302A

The Grandstream UCM6302A is part of the UCM6300 Audio series of IP PBXs, a platform that unifies business communications including voice, instant messaging (IM), voice meetings, audio web meetings, data, analytics, mobility, facility access and intercoms. It supports up to 500 users and 75 concurrent calls, with 2 FXS and 2 FXO ports for analog phones and PSTN lines. The built-in IM and voice/web conferencing platform, the free Wave app for desktop, web and mobile, UCM RemoteConnect for remote users, Hot Standby high availability, GDMS cloud management and an API for CRM and PMS integration make it a practical voice platform for offices, hotels, clinics, schools and retail chains.

Technical Specifications – Grandstream UCM6302A

Hardware

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 mode) with PoE+
NAT RouterYes (supports router mode and switch mode)
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 & Telephony

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, G.722.1, G.722.1C, G.723.1 5.3K/6.3K, G.726-32, G.729A/B, iLBC, GSM; T.38
Telephony Operating SystemBased on Asterisk version 16
DTMF MethodsIn-band audio, RFC4733, and SIP INFO
Disconnect MethodsBusy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect
Caller IDBellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT
Polarity Reversal/WinkYes, with enable/disable option upon call establishment and termination

Network & Security

QoSLayer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
APIFull API available for third-party platform and application integration
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, IPv6, OpenVPN
Media EncryptionSRTP, TLS, HTTPS, SSH, 802.1X
Internet Protocol StandardsRFC 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
Firmware UpgradeSupported by Grandstream Device Management System (GDMS), a zero-touch cloud provisioning and management system that provides a centralized interface to provision, manage, monitor and troubleshoot Grandstream products

Capacity & Features

Maximum Call CapacityUsers: 500; Concurrent calls (G.711): 75; Max concurrent SRTP calls (G.711): 75
Maximum Meeting CapacityUnlimited meeting rooms (up to 5 simultaneous public rooms); maximum of 75 simultaneous participants across all rooms combined
Call CenterMultiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/workload, in-queue announcement
Customizable Auto AttendantUp to 64 layers of IVR (Interactive Voice Response) in multiple languages
Wave AppFree; 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 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
Multi-Language SupportWeb 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

Physical

Universal Power SupplyInput: 100 ~ 240VAC, 50/60Hz; Output: DC 12V, 1.5A
Dimensions270mm(L) x 175mm(W) x 36mm(H)
WeightUnit Weight: 725g; Package Weight: 1221g
Temperature & HumidityOperating: 32 - 113ºF / 0 ~ 45ºC, Humidity 10 - 90% (non-condensing); Storage: 14 - 140ºF / -10 ~ 60ºC, Humidity 10 - 90% (non-condensing)
MountingWall mount & Desktop
ComplianceFCC: 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

Source: Grandstream official website and datasheet. Specifications are subject to change by the manufacturer.

Equipping a project with several devices?We recommend the right models and prepare a formal quotation.
Request a quote

PRACTICAL ANSWERS

Business PBX questions

How do I estimate the size of a business phone system?

Separate the number of extensions from the number of simultaneous external calls. Also list sites, users, call routing and growth needs so the proposed system and licenses can be checked against the actual workload.

Should the PBX be on-site or hosted?

Compare administration responsibilities, connectivity, continuity needs and the available support arrangement. Ask for a clear description of what is hosted, what remains at your site, and any recurring licenses or service charges.

Can existing business phones be reused?

Provide the exact phone models and firmware details where available. Check the proposed PBX’s supported device list and required features, then test a representative device before assuming every existing phone can be retained.

Are telephone numbers and call service included with the PBX?

Treat the PBX equipment or software and the telephone service as separate scope items unless the quotation explicitly combines them. Confirm responsibility for numbers, external calling capacity, provider coordination and ongoing charges.

How should business-hours and after-hours calls be handled?

Write the intended call flow before configuration: greetings, departments, fallback destinations and closing-time behavior. Review the wording in the required languages and test each route with a real representative call.

What should be included in a PBX handover?

Agree on the extension list, call-flow diagram, authorized administrator access and configuration backup process. Test the required incoming and outgoing calls, transfers and fallback paths, and document any functions excluded from the scope.

Explore this topic