Description

Grandstream UCM6308 IP PBX Oman

The Grandstream UCM6308 Oman features the Wave app for web and mobile which offers a platform, enabling you to collaborate remotely. UCM Remote Connect which is a cloud NAT traversal service ensures secure remote connections. Through GDMS and API, the Grandstream UCM6308 Oman offers cloud set up and management which allows for easy integration with 3^rd party platforms. By offering a high-end unified communications and collaboration solution packed with a suite of mobility, security, meeting and collaboration tools, the UCM6300 series provides a powerful platform for any organization.

Features

High Performance & Stability

The Grandstream UCM 6308 Oman comes with a distributed architecture that configures an integrated media resource card. It also not only features a highly responsive concurrent call processing, but also a smart auto load balancing function which results in an excellent performance, thus giving you a maximum communication experience.

Supports Power over Ethernet

The Grandstream UCM6308 IP PBX Oman is equipped with 3 self-adaptive 10/100Mbps ports which are integrated with Power over Ethernet to allow for convenient installation and a variety of remote features including provisioning, status monitoring and handset firmware upgrades.

Cloud Management

Grandstream provides a reliable and secure cloud service to deliver connectivity to its users at cheaper maintenance costs while enhancing operation and maintenance efficiency. The cloud service on the internet enables customers to gain direct access to Grandstream devices available in private networks such as IP-PBXs, audio gateways, and IP phones, therefore allowing simple remote maintenance and management of Grandstream devices. This kind of cloud service is specifically designed to meet the needs of large-scale installation, configuration, and maintenance and operation. Auto-provisioning and configuration backup, online update, real-time monitoring and alert enable users to achieve efficient operation and maintenance, thus maximizing productivity in organizations.

Flexible Deployment, Easy Operation & Simple Maintenance

The Grandstream UCM 6308 Oman comes with a centralized single-site deployment and multi-site distributed network. Its centralized equipment management ensures a highly efficient operation and maintenance. The highly efficient Grandstream UCM 6308 IP PBX system Oman uses a graphical configuration interface to accord you with an easy user experience even without any technical expertise.

Supports 8 FXO Ports and 8 FXS Ports

Foreign Exchange Subscribers is the port that actually delivers the analog line to the subscriber. It is simply the “plug in the wall” that delivers a dial tone, battery current and ring voltage. This is the jack or interface to the phone system to which FXO devices can be connected to. The Grandstream UCM6308 IP PBX Oman has been equipped with 8 FXS ports to maximize communication while using your telephone to make calls, thus maximizing work productivity. It also comes with 8 FXO ports and allows for the making of up to 450 concurrent VoIP calls. It supports up to 3000 SIP devices or users, proving to be a highly efficient device in busy offices.

Automated – Provisioning – TFTP/HTTP/HTTPS

The Grandstream UCM 6308 Oman has the ability to deploy an information technology or telecommunications service by using embedded pre-deﬁned procedures that are carried out electronically without requiring any human intervention, thus making it very reliable and convenient.

Rich QoS Capabilities

Quality of Service (QoS) is a feature of routers and switches which prioritizes traffic so that more important traffic can pass first, resulting in an overall performance improvement for critical network traffic. This feature is very useful in the Grandstream UCM 6308 IP PBX whenever there are high volumes of traffic, therefore eliminating any delays while working.

Excellent Audio Quality

The high performing Grandstream UCM6308 Oman offers users distraction-free communication courtesy of the inbuilt industry-leading Acoustic Echo Canceling Technology, Adaptive Jitter Buffer, Voice Activity Detection, and Comfort Noise Generation that deliver superior High Definition sound quality devoid of extraneous noises. This makes it possible to enjoy fluent phone conversations without any distractions.

SIP – Based Connections

You can easily connect your Grandstream UCM 6308 IP PBX to your IP PBX System or direct to a VoIP service provider in order to enjoy seamless conference calls. The Grandstream UCM 6304 IP PBX Oman is based on SIP and works perfectly with most of the SIP platforms such as Cisco CallManager/CUCM, Broadsoft, Microsoft Skype for Business (Lync), – Huawei IMS, and Asterisk/Elastix that currently exist in the market.

Firmware Upgradeable

You can easily update the operating system on your Grandstream UCM 6308 IP PBX Oman to improve its functionality and enhance user experience. Upgrading firmware also fixes any existing bugs and protects you from any kind of software malfunctions and security threats. To update your gadget’s firmware, type your gateway’s IP address into your web browser and enter your login information. Then locate the Firmware or Update section and download the latest firmware update on your Grandstream UCM6308 IP PBX system’s manufacturer’s website. Finally, upload the update and reboot the telephony system.

Call Features

Call Transfer : This advanced feature of the Grandstream UCM6308 Oman allows the user to relocate an inbound call to another phone or messaging system by using a dedicated call transfer button, or software that has been configured for use on the Gateway.
Call Waiting : You can hear another incoming call when you are already on an active phone call (beep). With the Grandstream UCM6308 IP PBX you can also turn off call waiting so that incoming calls are directly sent to voicemail during moments you are active on another phone call.
Call Holding : You can easily place an active phone call on hold in order to make or pick another incoming call using the premium Grandstream UCM6308 Oman IP PBX.

Key Features

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

Specifications

Analog Telephone FXS Ports

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

PSTN Line FXO Ports

8 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

2*USB 3.0
1*SD 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 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

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

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 : 2550g
Package Weight : 3320g

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, 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 : 3000
Concurrent calls (G.711) : 450
Max concurrent SRTP calls (G.711) : 300

Maximum Attendees of Conference Bridges

8 Video Conference rooms and up to 60 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & G.711)
Voice Conference : Up to 300 parties (G.711)

Wave Mobile App

Allows Android & iOS users to join UCM-hosted meetings & communicate with other users/solutions registered to the UCM6300

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.