Grandstream UCM6301 IP PBX

The Grandstream UCM6301 is an advanced IP PBX solution designed to provide a unified communication platform for small and medium-sized businesses. This phone system supports up to 500 users and 75 concurrent calls, making it an ideal choice for businesses looking for a powerful yet affordable IPPBX. The system provides two FXO analog PSTN lines, two FXS analog telephony ports and integrated PoE (Power over Ethernet) – optimizing operational efficiency and reducing the necessity for excess cables or adapters.

In terms of connectivity options, the UCM6301 offers extensive options including Gigabit network ports with integrated PoE, USB, SD card interfaces for recording calls or importing/exporting data and an HDMI output for connection to a video display. Embedded with advanced security features such as secure boot, data encryption, TLS/SRTP/HTTPS for advanced privacy, and firewall functionality, the UCM6301 ensures secure and reliable communication.

The automated NAT firewall traversal service facilitates easy installation behind enterprise firewalls. Additionally, the Grandstream UCM6301 PBX is equipped with API (Application Programming Interface) support which allows third-party platform integration for custom application development, providing flexibility and increased utility. It also comes with a built-in conference bridge that supports up to 25 attendees, facilitating seamless business communication.

Key Features of the UCM6301 IP PBX:

• Accommodates up to 500 users and handles 75 concurrent calls seamlessly.
• Simplifies setup with zero-configuration provisioning for Grandstream SIP devices.
• Includes a built-in conferencing and meetings platform compatible with desktops, the Wave app, and SIP endpoints.
• Wave app available for Android, iOS, Chrome, and Firefox browsers, enabling communication across all UCM6300 users and solutions.
• Offers API support for integrating third-party systems such as CRM and PMS platforms.
• Delivers advanced security with secure boot, unique certificates, and randomized default passwords to safeguard calls and accounts.
• Features three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and NAT router support.
• Provides an automated NAT firewall traversal service for secure and reliable remote connections.
• Supports Full-Band Opus voice codec and H.264/H.263/H.263+/H.265/VP8 video codecs with jitter resilience of up to 50% packet loss.
• Fully compatible with GDMS for streamlined cloud-based setup, management, and monitoring.
• Built on the flexible Asterisk* version 16 open-source telephony operating system.

Analog Telephone FXS Ports

• 1 RJ11 Port
• All ports have lifeline capability in case of power outage

PSTN Line FXO Ports

• 1 RJ11 Port
• 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 3.0
• 1*SD card 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

Video Codecs

• H.264, H.263, H263+, H.265, VP8

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, RFC2833, 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: 715g
• Package Weight: 1211g

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/ work-load, 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): 50

Maximum Attendees of Conference Bridges

• 2 Video Conference rooms and up to 12 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & G.711)
• Voice Conference: Up to 75 parties (G.711)

Wave Mobile App

• Allows Android & iOS users to join UCM-hosted meetings & communicate with other users/solutions registered to the UCM6301

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, 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

Firmware Upgrade

• Supported 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

Compliance

• FCC: Part 15 (CFR 47) Class B, Part 68
• CE: EN 55032, EN 55035, EN 61000-3-2, EN 61000-3-3, EN 62368-1, ETSI 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

Datasheet

• Datasheet_UCM6300_Series_English