Grandstream UCM6301 IP PBX Dubai
The Grandstream UCM6301 IP PBX Dubai System allows for pairing with the Grandstream UCM6300 series to achieve a hybrid platform that combines the control of an on-premise IP PBX with the remote access of a cloud solution. It is packed with many advanced features which enhance the communication experience and maximize productivity in businesses. The Grandstream UCM6301 IP PBX Dubai 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 UCM6301 Dubai offers cloud set up and management which allows for easy integration with 3rd 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.
Grandstream UCM6301 IP PBX Features
High Performance & Stability
The Grandstream UCM 6301 IP – PBX Dubai 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.
Flexible Deployment, Easy Operation & Simple Maintenance
The Grandstream UCM 6301 IP PBX system Dubai 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 6204 IP PBX system uses a graphical configuration interface to accord you with an easy user experience even without any technical expertise.
Supports 1 FXO Port and 1 FXS Port
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 UCM6301 IP PBX Dubai has been equipped with 1 FXS port to maximize communication while using your telephone to make calls, thus maximizing work productivity. It also comes with 1 FXO port and allows for the making of up to 75 concurrent VoIP calls. It supports up to 500 SIP devices or users, proving to be a highly efficient device in busy offices.
Automated – Provisioning – TFTP/HTTP/HTTPS
The Grandstream UCM 6301 IP PBX Dubai 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 6301 IP PBX Dubai whenever there are high volumes of traffic, therefore eliminating any delays while working.
Excellent Audio Quality
The high performing Grandstream UCM 6301 IP PBX Dubai 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 6301 IP PBX Dubai to your IP PBX System or direct to a VoIP service provider in order to enjoy seamless conference calls. The Grandstream UCM 6301 IP PBX Dubai is based on SIP v2 (RFC 3261) 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.
You can easily update the operating system on your Grandstream UCM 6301 IP PBX Dubai 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 UCM 6301 IP PBX system’s manufacturer’s website. Finally, upload the update and reboot the telephony system.
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.
The Grandstream UCM6301 IP PBX Dubai pushes versatility to a new high, thanks to its ability to complement already existing fixtures for enhanced web and audio conferencing, as well as other communication needs. This makes it easy to integrate seamlessly into any office setup without disorienting earlier structures and work ethic. Furthermore, the device is easy to redeploy as the business grows and needs to transform.
- Call Transfer: This advanced feature of the Grandstream UCM 6301 IP PBX system 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 UCM6301 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 UCM6301 IP PBX.
Grandstream UCM6301 Key Features
- Supports up to 500 users and up to 75 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
Grandstream UCM6301 IP PBX Specifications
Analog Telephone FXS Ports
- 1 RJ11 Port
- All ports have lifeline capability in case of a power outage
PSTN Line FXO Ports
- 1 RJ11 Port
- All ports have lifeline capability in case of a power outage
- Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+
- Yes (supports router mode and switch mode)
- 1*USB 3.0
- 1*SD card interface
- 320×240 colour LCD with touch screen for Shortcut Keys and Scroll Bar
- Yes, long press for factory reset and short press for the reboot
- 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
- 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
- 264, H.263, H263+, H.265, VP8
- Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
- Full API available for third-party platform and application integration
Telephony Operating System
- Based on Asterisk version 16
- 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), event list between local and remote trunk
- 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®
- Busy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect
- SRTP, TLS, HTTPS, SSH, 802.1X
Universal Power Supply
- Input: 100 ~ 240VAC, 50/60Hz; Output: DC 12V, 1.5A
- 270mm(L) x 175mm(W) x 36mm(H)
- 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)
- Wall mount & Desktop
- 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
- Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT
- Yes, with enable/disable option upon call establishment and termination
- 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 UCM6300
- 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, event list, feature codes, busy camp-on/ call completion, voice control.