Grandstream UCM6300A Audio Series IP PBX
₦286,000.00
- 250 users
- 50 concurrent calls
- 3 Gigabit RJ45 network ports
- Integrated PoE
- Description
- Reviews (0)
- Specifications
The Grandstream UCM6300A is an audio-only IP PBX solution for businesses in Nigeria looking to upgrade their telephone systems. The UCM6300A features a set of tools for enabling high-quality IP voice calling, providing unrivalled support for up to 5000 users, 450 concurrent calls, and up to 200 concurrent SIP conference attendees. Furthermore, the UCM6300A leverages a zero-configuration provisioning system that facilitates straightforward and rapid deployment.
Grandstream UCM6300A Key Features:
This Audio IP PBX also incorporates an embedded Debian 9 Linux OS with Asterisk 16, offering superior compatibility with most SIP devices and offering advanced call routing, network-backed automated attendants, call recording, central control panel for endpoints, IVR, and call queuing functionalities. With a focus on security, the Grandstream UCM6300A boasts an array of security protocols including TLS encryption for SIP signalling and SRTP encryption for voice/data content.
The device supports dual Gigabit network ports with integrated PoE, USB, SD; while providing broad interoperability with the majority of third-party SIP devices and leading SIP/NGN/IMS platforms. To further enhance user experience, this product also incorporates an API for third-party application development, opening up opportunities for further customization to meet specific business needs.
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| Analog Telephone FXS Ports | None All ports have lifeline capability in case of power outage |
| PSTN Line FXO Ports | None 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 color 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 |
| 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 | 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 |
| 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: 705g Package Weight: 1131g |
| 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 Customizable IVR/voice prompts: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, Nederlands Customizable 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/workload, in-queue announcement |
| Customizable Auto Attendant | Up to 5 layers of IVR (Interactive Voice Response) in multiple languages |
| Maximum Call Capacity | Users: 250 Concurrent calls (G.711): 50 Max concurrent SRTP calls (G.711): 50 |
| Maximum Attendees of Conference Bridges | 3 meeting rooms and up to 50 parties |
| Wave Mobile App | Free; Available for desktop (Windows 10+, Mac OS 10+), web (Firefox and Chrome Browsers) and mobile (Android & iOS), allows users to join UCM-hosted meetings, communicate with other users/solutions and make/receive calls using SIP accounts registered to a UCM6300 Audio series IP PBX |
| 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 meeting, event list, 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 |
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