Grandstream UCM6304A Audio VoIP PBX
₦444,750.00
- 1000 users
- 150 concurrent calls
- NAT Router Support
- 3 Gigabit RJ45 network ports
- Description
- Reviews (0)
- Specifications
The Grandstream UCM6304A Audio VoIP PBX is a cutting-edge communication solution engineered to enhance efficiency and connectivity within your business environment. The UCM6304A features four FXO ports and two FXS ports, offering ample capacity for the connectivity of numerous telephone devices. The product’s 2 RJ45 10/100/1000Mbps ports with integrated PoE Plus are designed to provide swift, efficient data transfer and simplified device installation.
At the core of the UCM6304A is its Zero Configuration provisioning, which allows for quick and effortless setup of Grandstream SIP endpoints. It also supports up to 1000 SIP endpoint registrations, 150 concurrent calls, and up to a maximum of 6 conference bridges with up to 32 simultaneous call participants. Another noteworthy aspect of the UCM6304A is its dual Gigabit RJ45 ports with an integrated NAT router for swift, efficient networking and internet access.
It also comes with three conference bridges that allow for five-way voice conferencing sessions – perfect for collaborative work environments. With security as a priority, the Grandstream UCM6304A also comes equipped with advanced defense mechanisms such as secure boot, unique security certificate per device, random default password per device, encrypted data storage, TLS/SRTP/HTTPS encryption and many more.
Key Features of the UCM6304A Audio Series:
- Compatible with GDMS for easy cloud-based setup, monitoring, and management.
- No monthly per-seat fees—an all-inclusive telephone system.
- Supports up to 1,000 users and 150 concurrent calls.
- Advanced security features, including secure boot, unique certificates, and random default passwords to safeguard calls and accounts.
- Built-in platform for instant messaging, audio conferencing, and web meetings, accessible via computers, mobile devices, and SIP endpoints.
- Zero configuration provisioning for Grandstream SIP endpoints simplifies setup.
- Offers three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and NAT router support.
- Supports Full-Band Opus voice codec with jitter resilience of up to 50% packet loss.
- Includes a free Wave App for voice and instant messaging on desktops, web, and Android/iOS devices.
- API available for third-party CRM and PMS integrations.
- Enhanced reliability with Hot Standby High-Availability and local dual deployment support.
- Automated NAT firewall traversal ensures secure remote connections.
- Based on Asterisk* version 16, a powerful open-source telephony operating system.
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| Analog Telephone FXS Ports | 4 RJ11 ports, all with lifeline capability in case of power outage |
| PSTN Line FXO Ports | 4 RJ11 ports, all with 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 | 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, 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, G.722.1 G.722.1C, 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), Layer 3 QoS (ToS, DiffServ, MPLS) |
| 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 | AES encrypted XML configuration file, auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP Option 66, multicast SIP SUBSCRIBE, mDNS) |
| Network Protocols | TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP/HTTPS, PPPoE, STUN, SRTP, TLS, LDAP, HDLC, 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: 775g; Package: 1621g |
| 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, Chinese (Simplified & Traditional), Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish. IVR: customizable in many 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 Features | Multiple configurable call queues, ACD based on agent skills/availability/workload, in-queue announcements |
| Auto Attendant | Up to 5 layers of IVR in multiple languages |
| Max Call Capacity | Users: 1000; Concurrent calls (G.711): 150; Max concurrent SRTP calls (G.711): 120 |
| Conference Bridge | 7 meeting rooms and up to 120 parties |
| Wave Mobile App | Free; Available for desktop (Windows, Mac OS), web (Firefox, Chrome), and mobile (Android & iOS) |
| Call Features | Call park, forward, transfer, waiting, ID, record, history, ringtone, IVR, music on hold, DID, DOD, DND, DISA, ring group, voicemail, etc. |
| Firmware Upgrade | Supported by Grandstream Device Management System (GDMS), a zero-touch cloud provisioning and management system |
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