- Grandstream PABX, IP PABX, IP-PBX, PABX, PBX Phones & Systems, PBX System
Grandstream UCM6308A Audio Series IP PBX
Grandstream PABX, IP PABX, IP-PBX, PABX, PBX Phones & Systems, PBX SystemGrandstream UCM6308A Audio Series IP PBX
Got A Question? Contact Us.
We Offer Good Discount Expert Technical Support
Key Features
- Up to 1500 users
- Up to 200 concurrent calls
- 9 meeting rooms
Grandstream UCM6300 Audio Series Datasheet (PDF)
Grandstream UCM6304/8 & UCM6308A Quick Installation Guide (PDF)
Grandstream UCM630xA Series User Manual (PDF)SKU: UCM6308A - Grandstream PABX, IP PABX, IP-PBX, PABX, PBX System
Grandstream UCM6302 IP PBX Phone System
This series of IP PBX provides a platform that can integrate all business communications (including voice, video calls, video conferencing, video surveillance, web conferencing, data and analysis)
Grandstream UCM6302 Key Features
- Supports 150 concurrent calls and about 1000 users
- Ensure zero configuration of Grandstream SIP endpoints
- Built-in conferencing platform and supports Wave app, desktop, and SIP endpoints
- Wave for Android, iOS, Chrome and Firefox browser permit interaction with all UCM6300 users & solutions
- Availability of API for third-party integrations, including CRM and PMS platforms
SKU: UCM6302 - PBX Phones & Systems, IP PABX, PABX, IP-PBX, Grandstream PABX, PBX System
Grandstream UCM6304 IP PBX
This series of IP PBX provides a platform that can integrate all business communications (including voice, video calls, video conferencing, video surveillance, web conferencing, data and analysis)
Grandstream UCM6304 Key Features
- Supports 300 concurrent calls and about 2000 users
- Ensure zero configuration of Grandstream SIP endpoints
- Built-in conferencing platform and supports Wave app, desktop, and SIP endpoints
- Wave for Android, iOS, Chrome and Firefox browser permit interaction with all UCM6300 users & solutions
- Availability of API for third-party integrations, including CRM and PMS platforms
SKU: UCM6304
Grandstream UCM6308A Audio Series IP PBX
Got A Question? Contact Us.
We Offer Good Discount Expert Technical Support
Key Features
- Up to 1500 users
- Up to 200 concurrent calls
- 9 meeting rooms
Grandstream UCM6300 Audio Series Datasheet (PDF)
Grandstream UCM6304/8 & UCM6308A Quick Installation Guide (PDF)
Grandstream UCM630xA Series User Manual (PDF)
$810.59
Grandstream UCM6308A Audio Series IP PBX
Grandstream UCM6308A Audio Series IP PBX
The Grandstream UCM6308A IP PBX offers businesses a scalable solution for unified communications and collaboration. This IP PBX series enables integration of all essential communication channels—voice, video calls, video conferencing, video surveillance, network conferencing, data transmission, analysis, mobility, facility access, and more—into a single, centralised network.
With support for up to 1,500 users, the UCM6300 Audio series includes built-in web and video conferencing capabilities, allowing employees to connect seamlessly via desktop computers, mobile devices, GVC series devices, and IP phones. For added flexibility, it integrates with the UCM6300 ecosystem, providing a hybrid solution that combines local IP PBX management with secure, cloud-based remote access.
Key Features of the Grandstream UCM6308A Audio Series IP PBX:
- Supports up to 200 concurrent calls and 1,500 users
- Zero-configuration setup for Grandstream SIP endpoints
- Built-in conferencing platform with support for Wave app, desktop, and SIP endpoints
- Based on open-source Asterisk Version 16 telephony operating system
- Fully compatible with GDMS for cloud-based configuration, management, and monitoring
- Three gigabit auto-sensing RJ45 network ports with PoE+ support and integrated NAT router
- Automated NAT firewall traversal service for secure remote access
- Advanced security features, including secure boot, unique certificates, and random default passwords for enhanced call and account security
- Wave app available for Android, iOS, Chrome, and Firefox, enabling easy communication with all UCM6300 users and solutions
- API availability for integration with third-party platforms, such as CRM and PMS
The UCM6308A series is an ideal choice for enterprises seeking a powerful, versatile platform with comprehensive communication, security, and collaboration capabilities.
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8 Inch Cordless Smart Phone 4G VoLTE LTE SIM Card Fixed Wireless Desktop Phone
Desk Phone, IP Phone, LTE Desk IP Phone8 Inch Cordless Smart Phone 4G VoLTE LTE SIM Card Fixed Wireless Desktop Phone
- 8 inch touch screen HD 1280*720
- quad core cpu
- android 8.1
- 2GB RAM+16GB ROM
- wireless smart telephone / bluetooth handset/ video telephone /video phone
- 1 SIM NANO card
- 2g GSM:850/900/1800/1900
3g WCDMA:850/900/1900/2100
4g LTE-FDD:B1/2/3/4/7/8/17/20/28A/28B
4g TDD:38/39/40/41M - rear camera 5.0 mp
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Key Features :
- 4.3″ colour LCD screen with swappable face plates
- 2.4″ additional colour display screen
- 48 virtual multipurpose keys (VPKs)
Grandstream GRP2616 Datasheet (PDF)
Grandstream GRP2616 Quick Installation Guide (PDF)
Grandstream GRP2600 series User Guide (PDF)SKU: GRP2616 - IP Phone, SIP Phone, VoIP Phone, Yealink
Enhance Productivity with Yealink SIP-T46U IP Phone: 16 VoIP Accounts, 4.3-Inch Color Display, Dual USB 2.0, Dual-Port Gigabit Ethernet, 802.3af PoE.
IP Phone, SIP Phone, VoIP Phone, YealinkEnhance Productivity with Yealink SIP-T46U IP Phone: 16 VoIP Accounts, 4.3-Inch Color Display, Dual USB 2.0, Dual-Port Gigabit Ethernet, 802.3af PoE.
Features of Yealink SIP-T46U
- 4.3″ colour display
- Dual USB ports
- Up to 16 SIP accounts
SKU:SIP-T46UBrand: YealinkSKU: T46U - Fanvil IP Phone, Fanvil SIP / VoIP, IP Phone, IP Video Phone, SIP Phone, VoIP Phone
Enjoy High Quality Calls with Fanvil H5 Hotel Room IP Phone: Speakerphone with an RJ45 port, a USB port, SIP based VoIP, analog lines, and 7” color LCD display
Fanvil IP Phone, Fanvil SIP / VoIP, IP Phone, IP Video Phone, SIP Phone, VoIP PhoneEnjoy High Quality Calls with Fanvil H5 Hotel Room IP Phone: Speakerphone with an RJ45 port, a USB port, SIP based VoIP, analog lines, and 7” color LCD display
Fanvil H5 VoIP Phone Key Features
- 3.5 inch (480 x 320) colour screen
- 6 Soft keys programmable
- HD voice: HD handset
- 802.3af PoE
- Vxworks OS
SKU: FANVIL H5 - IP Phone
Escene WS330 Wireless IP Phone
Escene WS330 Key Features• Wireless working frequency 5.8GHz.• HD Voice: HD Codec, HD Speaker, HD Handset.• Multi-language, e.g. Chinese, English, Russian, France etc .• SIP accounts and support three-way conference, SMS.• Excellent adjustable bracket and wall-mountable.• 2xLAN, PoE, RJ9Headset.• Support 15 paperless programmable keys.• XML/LDA, BLF/BLA, Search someone in the Phone book.• Auto-provision, HTTP/TFTP/FTP, TR069.SKU: XPK-5487FB49 - IP Phone
Escene WS620 Wireless IP Phone
Got A Question? Contact Us.
We Offer Good Discount Expert Technical Support
Escene WS620 Key Features
- Wireless working frequency 5.8GHz.
- HD Voice: HD Codec, HD Speaker, HD Handset.
- Multi-language, e.g. Chinese, English, Russian, France etc
- 8 SIP accounts and support three-way conference, SMS.
- Excellent adjustable bracket and wall-mountable.
- 2xLAN, PoE, RJ9Headset.
SKU: XPK-5487FB50 - Fanvil SIP / VoIP, IP Phone, IP Video Phone, SIP Phone, VoIP Phone
Experience Clear Communication with Fanvil X1SP Enterprise IP Phone: 2.8” LCD Color Screen, Bluetooth Support, Conference Calling & Voicemail, Automatic Call Distribution (ACD)
Fanvil SIP / VoIP, IP Phone, IP Video Phone, SIP Phone, VoIP PhoneExperience Clear Communication with Fanvil X1SP Enterprise IP Phone: 2.8” LCD Color Screen, Bluetooth Support, Conference Calling & Voicemail, Automatic Call Distribution (ACD)
Main Features
- Dual Ethernet Ports
- 2.8” LCD Color Screen
- HD Voice Quality
- Bluetooth Support
- PoE & WiFi Support
- Programmable Keys
SKU: Fanvil X1SP - IP Phone, Fanvil SIP / VoIP, IP Video Phone, SIP Phone, Video Phones, VoIP Phone
Fanvil A32i Android Console IP Phone
IP Phone, Fanvil SIP / VoIP, IP Video Phone, SIP Phone, Video Phones, VoIP PhoneFanvil A32i Android Console IP Phone
Got A Question? Contact Us.
We Offer Good Discount Expert Technical Support
Fanvil A32i Key Features
- 20 SIP lines, 3-way conference, hotspot
- Support an external Fanvil USB camera (optional).
- 112 one-touch DSS keys on 10.1” capacity colour touch screen
- HD audio on speaker and handset
SKU: A32i - IP Phone, Fanvil IP Phone
Fanvil C600 Android IP Video Phone with Touchscreen & Dual Gigabit Ports
Got A Question? Contact Us.
We Offer Good Discount Expert Technical Support
Key Note
HD Voice
2x 100/1000M Network Port (Power over Ethernet)
Camera- 5 mega pixel & Support Video Call
7” TFT 800×480 Multi touch Screen
HDMI output
DSS Can Support Upto 100 Stations
On Line Recording & Message Forwarding
Call Conference EnhancementDownloads Documents & Manuals
IP-PBX Server CompatibilitySKU: C600
Grandstream UCM6308A – Technical Specifications
Analog Telephone FXS Ports
- 8 RJ11 ports
- All ports have lifeline capability in case of power outage
PSTN Line FXO Ports
- 8 RJ11 ports
- 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
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
- 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: 2538g
- Package Weight: 3463g
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
- 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: 1500
- Concurrent calls (G.711): 200
- Max concurrent SRTP calls (G.711): 150
Maximum Attendees of Conference Bridges
- 9 meeting rooms and up to 150 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
- 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
- H.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
- Customizable IVR/voice prompts: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, Netherlands
- 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/ work-load, in-queue announcement
Customizable 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
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, ETSI 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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Grandstream UCM6304 IP PBX
This series of IP PBX provides a platform that can integrate all business communications (including voice, video calls, video conferencing, video surveillance, web conferencing, data and analysis)
Grandstream UCM6304 Key Features
- Supports 300 concurrent calls and about 2000 users
- Ensure zero configuration of Grandstream SIP endpoints
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- Availability of API for third-party integrations, including CRM and PMS platforms
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Grandstream UCM6302 IP PBX Phone System
This series of IP PBX provides a platform that can integrate all business communications (including voice, video calls, video conferencing, video surveillance, web conferencing, data and analysis)
Grandstream UCM6302 Key Features
- Supports 150 concurrent calls and about 1000 users
- Ensure zero configuration of Grandstream SIP endpoints
- Built-in conferencing platform and supports Wave app, desktop, and SIP endpoints
- Wave for Android, iOS, Chrome and Firefox browser permit interaction with all UCM6300 users & solutions
- Availability of API for third-party integrations, including CRM and PMS platforms
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Grandstream UCM6510 Enterprise IP PBX
Grandstream UCM6510 enterprise IP-PBX
- 2x PSTN FXO Ports and USB Port
- UCM6510 support about 2000 endpoint SIP registrations, 200 concurrent calls and about 64 conference attendees
- fifty SIP Trunk Accounts
- Realize the highest possible security through SRTP, TLS and HTTPS with hardware encryption accelerators
- redundant power supply, hot standby cluster support and high availability to minimize system downtime (to be determined)
- 1GHz quad-core Cortex A9 processor with enough memory size of 1GB DDR3 Ram, 32GB Flash and SD Card port for local recording.
SKU: UCM6510
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