Grandstream UCM6301 Unified Communications IP PBX 500 Users, 75 Concurrent Calls, Built-in Conferencing, 3 Gigabit Ports, PoE+, Wave App, RemoteConnect, Asterisk-Based
The Grandstream UCM6301 delivers enterprise-grade Unified Communications & Collaboration (UCC) for small-to-medium businesses. Grandstream UCM6301 is designed as an affordable, secure, and scalable IP PBX, it supports up to 500 users and 75 concurrent calls, featuring integrated voice, video calls, video conferencing, audio conferencing, mobility tools, facility access, and intercom systems.
Grandstream UCM6301 comes with built-in audio and video conferencing, desktop and mobile Wave app support, and cloud-managed remote connections via UCM RemoteConnect, the UCM6301 provides a flexible, hybrid UC solution. It also features secure boot, TLS, SRTP encryption, and random default passwords for enhanced protection.
Based on Asterisk 16, this system supports advanced call handling, customizable multi-layer IVRs, call queues, and integrates easily with CRM and PMS platforms via its open API. Management and provisioning are simplified with Grandstream’s GDMS cloud platform and ZeroConfig auto-provisioning for SIP endpoints.
Grandstream UCM6301 is ideal for growing businesses looking for a reliable, full-featured UC solution, the UCM6301 offers enterprise-level features in a compact, easy-to-deploy form factor.
🎯 Key Features:
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Supports up to 500 users and 75 concurrent calls
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1 FXS and 1 FXO analog port (with lifeline capability)
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3 Gigabit network ports with PoE+
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Built-in audio/video conferencing with up to 4 video rooms and 75 participants
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Wave App for Android, iOS, Chrome, and Firefox for calling, messaging, and conferencing
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Secure remote access via UCM RemoteConnect NAT traversal
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ZeroConfig provisioning for Grandstream SIP endpoints
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Advanced call center features: call queues, ACD, skill-based routing
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Supports Full-Band Opus, G.711, G.722, G.729, GSM, T.38, H.264, H.263, VP8 codecs
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Asterisk 16-based telephony OS
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Advanced security: Secure boot, SRTP, TLS, HTTPS, random passwords, 802.1X
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API for CRM, PMS, and third-party platform integration
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Cloud device management with Grandstream Device Management System (GDMS)
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Customizable IVR with up to 5 layers and multi-language prompts
- Supports up to 3000 users and up to 450 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
Feature | UCM6301 |
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Analog Telephone FXS Ports | 1 x RJ11 port (with lifeline capability in case of power outage) |
PSTN Line FXO Ports | 1 x RJ11 port (with 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 x USB 3.0, 1 x SD card interface |
LED Indicators | None |
LCD Display | 320×240 color LCD with touch screen for shortcut keys and scroll bar |
Reset Switch | Yes, long press for factory reset, short press for reboot |
Voice-over-Packet Capabilities | LEC with NLP, 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, G.722.1, G.722.1C, G.723.1 (5.3K/6.3K), G.726-32, G.729A/B, iLBC, GSM, T.38 |
Video Codecs | H.264, H.263, H.263+, VP8 |
QoS | Layer 2 QoS (802.1Q, 802.1p), 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, RFC2833, SIP INFO |
Provisioning Protocols & Plug-and-Play | Mass provisioning via AES encrypted XML, 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 | SIP, TCP/UDP/IP, RTP/RTCP, IAX, 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, ZRTP |
Power Supply | Input: 100–240VAC, 50/60Hz; Output: DC 12V, 1.5A |
Dimensions | 270mm (L) x 175mm (W) x 36mm (H) |
Weight | Unit: 715g; Package: 1211g |
Temperature & Humidity | Operating: 0–45ºC, 10–90% (non-condensing); Storage: -10–60ºC, 10–90% (non-condensing) |
Mounting | Wall mount & desktop |
Multi-Language Support | Web UI: English, Simplified/Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish Customizable IVR: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, Nederlands |
Caller ID | Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT |
Polarity Reversal/Wink | Yes, with enable/disable option |
Call Center | Multiple configurable call queues, ACD based on agent skills/availability/workload, in-queue announcement |
Customizable Auto Attendant | Up to 5 layers of IVR in multiple languages |
Maximum Call Capacity | Up to 500 users, 75 concurrent calls (G.711), 50 max concurrent SRTP calls |
Conference Bridges | Up to 4 simultaneous video conference rooms, up to 75 participants in all rooms combined, up to 9 video feeds in all conference rooms Up to 75 voice conference parties |
Wave App | Free app for desktop, web, and mobile (Android/iOS) for meetings and SIP calls |
Call Features | Call park, forward, transfer, waiting, caller ID, record, history, ringtone, IVR, music on hold, call routes, DID, DOD, DND, DISA, ring group, time schedule, PIN groups, call queue, pickup group, paging/intercom, voicemail, SCA, BLF, voicemail to email, fax to email, speed dial, call back, dial by name, emergency call, follow me, blacklist/whitelist, voice conference, video conference, eventlist, feature codes, busy camp-on/call completion, voice control, post-meeting reports, virtual fax sending/receiving, email to fax |
Firmware Upgrade | Supported by Grandstream Device Management System (GDMS) for centralized provisioning and management |
Compliance | FCC, CE, IC, RCM, UL, RoHS |
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