Close
You have no items in your shopping cart.
Search
Filters
Παράδοση σε 1-2 μέρες
SKU: UCM6302
$496.75
Ship to
*
*
Shipping Method
Name
Estimated Delivery
Price
No shipping options

Η σειρά UCM6300 επιτρέπει στις επιχειρήσεις να δημιουργήσουν ισχυρές και επεκτάσιμες ολοκληρωμένες λύσεις επικοινωνίας και συνεργασίας.Αυτή η σειρά IP PBX παρέχει μια πλατφόρμα που ενοποιεί όλες τις επιχειρηματικές επικοινωνίες σε ένα κεντρικό δίκτυο, συμπεριλαμβανομένης της φωνής, των βιντεοκλήσεων, της τηλεδιάσκεψης, της παρακολούθησης βίντεο, των συναντήσεων ιστού, των δεδομένων, των αναλυτικών στοιχείων, της κινητικότητας, της πρόσβασης στις εγκαταστάσεις, των ενδοεπικοινωνιών και πολλά άλλα.

Χαρακτηριστικά

Analog Telephone FXS Ports:

2 RJ11 Ports

PSTN Line FXO Ports:

2 RJ11 Ports

Network Interfaces:

Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+

NAT Router:

Yes

Peripheral Ports:

1*USB 2.0, 1*USB 3.0, 1*SD card interface

LED Indicators:

None

LCD Display:

320x240 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

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:

SIP, 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®

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: 715g;
Package Weight: 1211g

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/
work-load, in-queue announcement

Customizable Auto Attendant:

Up to 5 layers of IVR (Interactive Voice Response) in multiple languages

Maximum Call Capacity:

Users: 1000
Concurrent calls (G.711): 150
Max concurrent SRTP calls (G.711): 100

Maximum Attendees of
Conference Bridges:

6 Video Conference rooms and up to 30 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus)
Voice Conference: Up to 150 parties (G.711)

Wave App:

Free

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, post-meeting reports, virtual fax sending/receiving, email to fax

Η σειρά UCM6300 επιτρέπει στις επιχειρήσεις να δημιουργήσουν ισχυρές και επεκτάσιμες ολοκληρωμένες λύσεις επικοινωνίας και συνεργασίας.Αυτή η σειρά IP PBX παρέχει μια πλατφόρμα που ενοποιεί όλες τις επιχειρηματικές επικοινωνίες σε ένα κεντρικό δίκτυο, συμπεριλαμβανομένης της φωνής, των βιντεοκλήσεων, της τηλεδιάσκεψης, της παρακολούθησης βίντεο, των συναντήσεων ιστού, των δεδομένων, των αναλυτικών στοιχείων, της κινητικότητας, της πρόσβασης στις εγκαταστάσεις, των ενδοεπικοινωνιών και πολλά άλλα.

Χαρακτηριστικά

Analog Telephone FXS Ports:

2 RJ11 Ports

PSTN Line FXO Ports:

2 RJ11 Ports

Network Interfaces:

Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+

NAT Router:

Yes

Peripheral Ports:

1*USB 2.0, 1*USB 3.0, 1*SD card interface

LED Indicators:

None

LCD Display:

320x240 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

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:

SIP, 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®

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: 715g;
Package Weight: 1211g

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/
work-load, in-queue announcement

Customizable Auto Attendant:

Up to 5 layers of IVR (Interactive Voice Response) in multiple languages

Maximum Call Capacity:

Users: 1000
Concurrent calls (G.711): 150
Max concurrent SRTP calls (G.711): 100

Maximum Attendees of
Conference Bridges:

6 Video Conference rooms and up to 30 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus)
Voice Conference: Up to 150 parties (G.711)

Wave App:

Free

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, post-meeting reports, virtual fax sending/receiving, email to fax

Write your own review
  • Only registered users can write reviews
*
*
Bad
Excellent
*
*
*