Grandstream UCM6308A Audio Series IP PBX

The Grandstream UCM6308A is an innovative audio IP PBX system that offers unparalleled functionality, high-quality audio communications, and excellent scalability. It incorporates an auto-attendant for automatic call handling and a voicemail facility for managing incoming messages effectively. Additionally, it is equipped with state-of-the-art technology to facilitate call recording, call queuing, call routing and call detail record (CDR) for detailed analysis and reporting of call history. The audio only VoIP telephone system ability to support up to 1500 users and 200 concurrent calls, which is beneficial for large-scale organizations that require heavy call traffic management. Additionally, this system includes a dual Gigabit network port with integrated PoE, which provides optimal network connectivity and efficient power usage.

The Grandstream UCM6308A also comes with an integrated NAT router which helps in seamless network routing. It also offers full encryption security using SRTP and TLS protocols. Moreover, it allows integration with third-party SIP devices and major voice-over-IP (VoIP) services making it a versatile option for different business needs.

The system also supports customizable call-routing through its Interactive Voice Response (IVR) system that can be tailored as per the organization’s needs. Its zero-configuration provisioning eliminates the complexity associated with setting up the system while ensuring a quick installation process.

Key Features of the UCM6308A:

• Supports up to 1500 users and 200 concurrent calls.
• No monthly per-seat fees, offering an all-inclusive telephone system.
• Integrated Instant Messaging (IM), audio conferencing, and web meetings, accessible from desktops, mobile devices, and SIP endpoints.
• Zero-configuration provisioning for Grandstream SIP endpoints.
• Free Wave App enables voice and IM communication on desktops, Android, iOS, and web platforms.
• Three Gigabit RJ45 ports with PoE+ support and NAT router functionality.
• Advanced security with secure boot, unique certificates, and randomized default passwords for calls and account protection.
• Hot Standby High-Availability and local dual deployment enhance reliability.
• Automated NAT firewall traversal for secure remote connections.
• Full-Band Opus codec with jitter resilience, handling up to 50% packet loss.
• API supports third-party integrations, including CRM and PMS platforms.
• Compatible with GDMS for cloud-based setup, management, and monitoring.
• Built on Asterisk* version 16 open-source telephony operating system.

Analog Telephone FXS Ports

8 RJ11 ports, all ports have lifeline capability in case of power outage

PSTN Line FXO Ports

8 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

2USB 3.0, 1SD card interface

LED Indicators

Power 1/2, FXS, FXO, LAN, WAN, Heartbeat

LCD Display

128×32 dot matrix graphic LCD with DOWN and OK buttons

Reset Switch

Yes, long press for factory reset and 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, 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

AES encrypted XML config file, auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig, event list 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

2x DC 12V Power Jack, Input: 100~240VAC, 50/60Hz; Output: DC 12V, 2A

Dimensions

485mm(L) x 187.2mm(W) x 46.2mm(H)

Weight

Unit Weight: 2538g, Package Weight: 3463g

Temperature & Humidity

Operating: 32 – 113ºF / 0 ~ 45ºC, Humidity 10 – 90% (non-condensing); Storage: 14 – 140ºF / -10 ~ 60ºC, Humidity 10 – 90% (non-condensing)

Mounting

Rack mount & Desktop

Multi-Language Support

Web UI: English, Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish; Customizable IVR/voice prompts in multiple languages; Language pack support for 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

Customizable Auto Attendant

Up to 5 layers of IVR (Interactive Voice Response) in multiple languages

Maximum Call Capacity

Users: 1500, Concurrent calls (G.711): 200, Max concurrent SRTP calls (G.711): 150

Maximum Attendees of Conference Bridges

9 meeting rooms and up to 150 parties

Wave Mobile App

Free; available for desktop (Windows 10+, Mac OS 10+), web (Firefox and Chrome browsers), and mobile (Android & iOS)

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, paging/intercom, voicemail, speed dial, blacklist/whitelist

Firmware Upgrade

Supported by Grandstream Device Management System (GDMS), a zero-touch cloud provisioning and management system

Datasheet

• UCM6300 Audio Series Datasheet English