- GXV Series, IP Video Phone, SIP Phone, IP Video Door Phone
Grandstream GXV3380 IP Video Desktop Phone
GXV Series, IP Video Phone, SIP Phone, IP Video Door PhoneGrandstream GXV3380 IP Video Desktop Phone
Grandstream GXV3380 IP Video Desktop Phone is such a wonderful quality high-end model for Android, with inbuilt Bluetooth and WIFI, wrapped up in all-in-one communication solution with ideal range of features that include multi-video platform collaboration, 8-inch 1280×800 capacitive touch screen and a 2-megapixel camera.Grandstream GXV3380 Key Features
- Android 7.X version
- Integrated 2MP video phone camera with private shutter
- Integrated dual-band WiFi (2.4 and 5GHz)
- Inbuilt Wi-Fi and Bluetooth
- IN and OUT HDMI ports
- Inbuild PoE / PoE +
SKU: GXV3380.
Grandstream UCM6300A IP PBX: The Future of Voice Communication | Supports up to 3000 Users and up to 450 Concurrent Calls | Advanced Security Protection with Secure Boot
- Supports Full-Band Opus voice codec
- Built-in conferencing & meetings platform
- Advanced security protection with secure boot
- Supports desktop, Wave app, and SIP endpoints
- Supports up to 3000 users and up to 450 concurrent calls
- Zero configuration provisioning of Grandstream SIP endpoints
- API available for third-party integrations, CRM and PMS platforms
- Compatible with GDMS for cloud setup, management and monitoring
- Based on Asterisk* version 16 open source telephony operating system
$257.00
Grandstream UCM6300A: The Future of Voice Communication | Supports up to 3000 Users and up to 450 Concurrent Calls | Advanced Security Protection with Secure Boot
Grandstream Launches UCM6300A the Next-Gen Audio Series IP PBX for Seamless Communication, Dive into the future of communication with Grandstream’s latest innovation, the UCM6300A. With its amazing features you will experience unparalleled connectivity and functionality, this IP PBX system is designed to streamline communication processes within your organization.
You can now Say goodbye to communication barriers and hello to seamless collaboration. This Voice Communication will elevate your business communication.
Features of Grandstream UCM6300A:
- Supports Full-Band Opus voice codec
- Built-in conferencing & meetings platform
- Advanced security protection with secure boot
- Supports desktop, Wave app, and SIP endpoints
- Supports up to 3000 users and up to 450 concurrent calls
- Zero configuration provisioning of Grandstream SIP endpoints
- API available for third-party integrations, CRM and PMS platforms
- Compatible with GDMS for cloud setup, management and monitoring
- Based on Asterisk* version 16 open source telephony operating system
- Automated NAT firewall traversal service facilitates secure remote connections
- H.264/H.263/ H.263+/H.265/VP8 video codec, jitter resilience up to 50% packet loss
- Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
Brands | |
---|---|
Color | |
Dimension | 270mm(L) x 175mm(W) x 36mm(H) |
Video Codecs | H.264, H.263, H263+, VP8 |
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 colour 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
Compliance
- FCC: Part 15 (CFR 47) Class B, Part 68
- CE: EN 55032, EN 55035, EN 61000-3-2, EN 61000-3-3, EN 62368.1, ES 203 021, ITU-T K.21
- IC: ICES-003, CS-03 Part I Issue 9
- RCM: AS/NZS CISPR 32, AS/NZS 62368.1, AS/CA S002, AS/CA S003.1/.2
- Power adapter: UL 60950-1 or UL 62368-1
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