Grandstream UCM6302A IP PBX Audio Only

Grandstream UCM6302A Audio Series IP PBX The Grandstream UCM6302A 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. Grandstream UCM6302A Key Features Supports up to 1500 users and up to 200 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 Grandstream UCM6302A – Technical Specifications Analog Telephone FXS Ports 2 RJ11 ports All ports have lifeline capability in case of power outage PSTN Line FXO Ports 2 RJ11 ports All ports have lifeline capability in case of power outage 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 2.0 1*USB 3.0 1*SDcard interface LED Indicators None LCD Display 320×240 colour LCD with touch screen for Shortcut Keys and Scroll Bar 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 Dimensions 270mm(L) x 175mm(W) x 36mm(H) Weight Unit Weight: 725g Package Weight: 1221g 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 Multi-Language Support Web UI: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish Customisable IVR/voice prompts: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, Nederlands Customisable 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 Customisable Auto Attendant Up to 5 layers of IVR (Interactive Voice Response) in multiple languages Maximum Call Capacity Users: 500 Concurrent calls (G.711): 75 Max concurrent SRTP calls(G.711): 75 Maximum Attendees of Conference Bridges 5 meeting rooms and up to 75 parties Wave Mobile App Free; Available for desktop (Windows 10 , Mac OS 10 ), web (Firefox and Chrome Browsers) and mobile (Android & iOS), allows usersto 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, callwakeup, 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 centralised interface to provision, manage, monitor and troubleshoot Grandstream products SKU: UCM6302A Categories: Grandstream UCM Series, IPBx Brand: Grandstream