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Grandstream UCM6301 IP PBX
Grandstream UCM6301 IP PBX
Grandstream UCM6301 IP PBX
manufacturer

Grandstream UCM6301 IP PBX

UCM6301
4204
In Stock
(0)Stock
Initial Price

348.60€

Price without VAT : 281.13€

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AVAILABILITY
For bulk availability inquiries please call 210-2209582

  • Description

    Grandstream UCM6301 IP PBX

    The UCM6300 series allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifies all business communication on one centralized network, including voice, video calling, video conferencing, video surveillance, web meetings, data, analytics, mobility, facility access, intercoms and more. The UCM6300 series supports up to 3000 users and includes a built-in web meetings and video conferencing solution that allows employees to connect from the desktop, mobile, GVC series devices and IP phones. It can be paired with the UCM6300 ecosystem to offer a hybrid platform that combines the control of an on-premise IP PBX with the remote access of a cloud solution. The UCM6300 ecosystem consists of the Wave app for desktop, web, and mobile, which provides a hub for collaborting remotely, and UCM RemoteConnect, a cloud NAT traversal service for ensuring secure remote connections. The UCM6300 series also offers cloud setup and management through GDMS and an API for integration with third-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.

    Features:

    • Supports up to 3000 users and up to 450 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

  • Specification
    Specifications
    Analog Telephone FXS Ports1 RJ11 Port
    All ports have lifeline capability in case of power outage
    PSTN Line FXO Ports1 RJ11 Port
    All ports have lifeline capability in case of power outage
    Network InterfacesThree self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+
    NAT RouterYes (supports router mode and switch mode)
    Peripheral Ports1*USB 3.0, 1*SD card interface
    LED IndicatorsNone
    LCD Display320x240 color LCD with touch screen for Shortcut Keys and Scroll Bar
    Reset SwitchYes, long press for factory reset and short press for reboot
    Voice-over-Packet CapabilitiesLEC 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 CodecsOpus, 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 CodecsH.264, H.263, H263+, H.265, VP8
    QoSLayer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
    APIFull API available for third-party platform and application integration
    Telephony Operating SystemBased on Asterisk version 16
    DTMF MethodsIn-band audio, RFC2833, and SIP INFO
    Provisioning Protocol & Plug-and-PlayMass 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 ProtocolsSIP, TCP/UDP/IP, RTP/RTCP, IAX, 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 MethodsBusy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect
    Media EncryptionSRTP, TLS, HTTPS, SSH, 802.1X
    Universal Power SupplyInput: 100 ~ 240VAC, 50/60Hz
    Output: DC 12V, 1.5A
    Dimensions(LxWxH)270mm x 175mm x 36mm
    WeightUnit Weight: 715g
    Package Weight: 1211g
    Temperature & HumidityOperating:
    32 - 113ºF / 0 ~ 45ºC,
    Humidity 10 - 90% (non-condensing)
    Storage:
    14 - 140ºF / -10 ~ 60ºC,
    Humidity 10 - 90% (non-condensing)
    MountingWall 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 IDBellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT
    Polarity Reversal/WinkYes, with enable/disable option upon call establishment and termination
    Call CenterMultiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/work-load, in-queue announcement
    Customizable Auto AttendantUp to 5 layers of IVR (Interactive Voice Response) in multiple languages
    Maximum Call CapacityUsers: 500
    Concurrent calls (G.711): 75
    Max concurrent SRTP calls (G.711): 50
    Maximum Attendees of Conference Bridges4 Video Conference rooms and up to 20 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus)
    Voice Conference: Up to 75 parties (G.711)
    Wave AppFree; Available for desktop (Windows 10+, Mac OS 10+), web (Firefox and Chrome Browsers) and mobile (Android & iOS), allows users to join UCM-hosted meetings/conferences, communicate with other users/solutions and make/receive calls using SIP accounts registered to a UCM6300 series IP PBX
    Call FeaturesCall 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, post-meeting reports, virtual fax sending/receiving, email to fax
    Firmware UpgradeSupported 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

  • Downloads

    Datasheet

    User manual

    Quick installation guide

Grandstream UCM6301 IP PBX

The UCM6300 series allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifies all business communication on one centralized network, including voice, video calling, video conferencing, video surveillance, web meetings, data, analytics, mobility, facility access, intercoms and more. The UCM6300 series supports up to 3000 users and includes a built-in web meetings and video conferencing solution that allows employees to connect from the desktop, mobile, GVC series devices and IP phones. It can be paired with the UCM6300 ecosystem to offer a hybrid platform that combines the control of an on-premise IP PBX with the remote access of a cloud solution. The UCM6300 ecosystem consists of the Wave app for desktop, web, and mobile, which provides a hub for collaborting remotely, and UCM RemoteConnect, a cloud NAT traversal service for ensuring secure remote connections. The UCM6300 series also offers cloud setup and management through GDMS and an API for integration with third-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.

Features:

• Supports up to 3000 users and up to 450 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

Specifications
Analog Telephone FXS Ports1 RJ11 Port
All ports have lifeline capability in case of power outage
PSTN Line FXO Ports1 RJ11 Port
All ports have lifeline capability in case of power outage
Network InterfacesThree self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+
NAT RouterYes (supports router mode and switch mode)
Peripheral Ports1*USB 3.0, 1*SD card interface
LED IndicatorsNone
LCD Display320x240 color LCD with touch screen for Shortcut Keys and Scroll Bar
Reset SwitchYes, long press for factory reset and short press for reboot
Voice-over-Packet CapabilitiesLEC 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 CodecsOpus, 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 CodecsH.264, H.263, H263+, H.265, VP8
QoSLayer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
APIFull API available for third-party platform and application integration
Telephony Operating SystemBased on Asterisk version 16
DTMF MethodsIn-band audio, RFC2833, and SIP INFO
Provisioning Protocol & Plug-and-PlayMass 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 ProtocolsSIP, TCP/UDP/IP, RTP/RTCP, IAX, 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 MethodsBusy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect
Media EncryptionSRTP, TLS, HTTPS, SSH, 802.1X
Universal Power SupplyInput: 100 ~ 240VAC, 50/60Hz
Output: DC 12V, 1.5A
Dimensions(LxWxH)270mm x 175mm x 36mm
WeightUnit Weight: 715g
Package Weight: 1211g
Temperature & HumidityOperating:
32 - 113ºF / 0 ~ 45ºC,
Humidity 10 - 90% (non-condensing)
Storage:
14 - 140ºF / -10 ~ 60ºC,
Humidity 10 - 90% (non-condensing)
MountingWall 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 IDBellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT
Polarity Reversal/WinkYes, with enable/disable option upon call establishment and termination
Call CenterMultiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/work-load, in-queue announcement
Customizable Auto AttendantUp to 5 layers of IVR (Interactive Voice Response) in multiple languages
Maximum Call CapacityUsers: 500
Concurrent calls (G.711): 75
Max concurrent SRTP calls (G.711): 50
Maximum Attendees of Conference Bridges4 Video Conference rooms and up to 20 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus)
Voice Conference: Up to 75 parties (G.711)
Wave AppFree; Available for desktop (Windows 10+, Mac OS 10+), web (Firefox and Chrome Browsers) and mobile (Android & iOS), allows users to join UCM-hosted meetings/conferences, communicate with other users/solutions and make/receive calls using SIP accounts registered to a UCM6300 series IP PBX
Call FeaturesCall 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, post-meeting reports, virtual fax sending/receiving, email to fax
Firmware UpgradeSupported 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

Ο καταναλωτής έχει την δυνατότητα να επιλέξει με ποια συνεργαζόμενη εταιρεία courier επιθυμεί να λάβει την παραγγελία του.

Η εταιρεία μας συνεργάζεται με τις ακόλουθες εταιρείες courier:

  • FIS courier
  • ACS courier
  • ELTA courier

* Για τον υπολογισμό του βάρους μιας παραγγελίας προσμετράται το βάρος ή ο όγκος. Ο τελικός υπολογισμός βάρους γίνεται από τον αριθμό που υπερτερεί (βάρος ή ογκομετρικό βάρος αντίστοιχα)

** Ο χρόνος παράδοσης εξαρτάται από την συνεργαζόμενη εταιρεία courier καθώς και από τον τελικό προορισμό της παραγγελίας.

  • Για παραγγελίες εντός Αττικής εως 14:00 μ.μ. εκτιμώμενος χρόνος παράδοσης: επόμενη εργάσιμη ημέρα.
  • Για παραγγελίες προς χερσαίους ή νησιώτικούς προορισμούς: 2 εργάσιμες ημέρες.
  • Για παραγγελίες προς δυσπρόσιτες ή απομακρυσμένες περιοχές: 3 – 4 εργάσιμες ημέρες

(Οι παραγγελίες δεν πραγματοποιούνται Σάββατα – Κυριακές και αργίες)

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