Grandstream UCM6302

Αρχική/Phones & Tablets/Τηλεφωνία/Τηλεφωνικά Κέντρα & Συσκευές/Τηλεφωνικά Κέντρα/Grandstream UCM6302

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ΚΩΔΙΚΟΣ: 0170600262

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Σε Απόθεμα

395.80 €

Ποσότητα:

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Online διαθεσιμοτητα προϊοντων ανα αποθηκη

Αθήνα

Τζώρτζ 30-32

Κεντρικές Αποθήκες

Περιγραφή

UCM6302 supports up to 1000 users and 150 concurrent calls. Integrated 2 PSTN trunk FXO ports, 2 analog telephone FXS ports.

Η σειρά UCM6300 επιτρέπει στις επιχειρήσεις να δημιουργήσουν ισχυρές και επεκτάσιμες ενοποιημένες 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.

Η σειρά UCM6300 υποστηρίζει έως και 3000 χρήστες και περιλαμβάνει μια ενσωματωμένη λύση διαδικτυακών συσκέψεων και τηλεδιάσκεψης που επιτρέπει στους υπαλλήλους να συνδέονται από επιτραπέζιους, κινητούς, συσκευές της σειράς GVC και τηλέφωνα IP. Μπορεί να συνδυαστεί με το οικοσύστημα UCM6300 για να προσφέρει μια υβριδική πλατφόρμα που συνδυάζει τον έλεγχο ενός PBX IP εσωτερικής εγκατάστασης με την απομακρυσμένη πρόσβαση μιας λύσης cloud. Το οικοσύστημα UCM6300 αποτελείται από την εφαρμογή Wave για επιτραπέζιους υπολογιστές, ιστό και κινητά, η οποία παρέχει έναν κόμβο για απομακρυσμένη συνεργασία, και το UCM RemoteConnect, μια υπηρεσία διέλευσης NAT cloud για τη διασφάλιση ασφαλών απομακρυσμένων συνδέσεων. Η σειρά UCM6300 προσφέρει επίσης ρύθμιση και διαχείριση cloud μέσω GDMS και API για ενοποίηση με πλατφόρμες τρίτων. Προσφέροντας μια ενοποιημένη λύση επικοινωνίας και συνεργασίας υψηλών προδιαγραφών γεμάτη με μια σειρά εργαλείων φορητότητας, ασφάλειας, συναντήσεων και συνεργασίας, η σειρά UCM6300 παρέχει μια ισχυρή πλατφόρμα για κάθε οργανισμό.

-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

Χαρακτηριστικά

Κατασκευαστής:

Grandstream

Διαστάσεις προϊόντος:

270mm(L) x 175mm(W) x 36mm(H)
Unit Weight: 715g

Επιπρόσθετα:

6 Video Conference rooms and up to 30 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus)
Analog Telephone FXS Ports: 2 RJ11 Ports
API: Full API available for third-party platform and application integration
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
Caller ID: Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT
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
LCD Display: 320x240 color LCD with touch screen for Shortcut Keys and Scroll Bar
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
Maximum Call Capacity: Users: 1000/Concurrent calls (G.711): 150/Max concurrent SRTP calls (G.711): 100
Mounting: Wall mount & Desktop
Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/ work-load, in-queue announcement
NAT Router: Yes
Network Interfaces: Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+
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®
Operating: 32 - 113ºF / 0 ~ 45ºC, Humidity 10 - 90% (non-condensing)
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
Peripheral Ports: 1*USB 2.0, 1*USB 3.0, 1*SD card interface
Polarity Reversal/Wink: Yes, with enable/disable option upon call establishment and termination
PSTN Line FXO Ports: 2 RJ11 Ports
QoS: Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
Reset Switch: Yes, long press for factory reset and short press for reboot
Telephony Operating System: Based on Asterisk version 16
Universal Power Supply: Input: 100 ~ 240VAC, 50/60Hz; Output: DC 12V, 1.5A
Video Codecs: H.264, H.263, H263+, H.265, VP8
voice conference, video conference, eventlist, feature codes, busy camp-on/ call completion, voice control, post-meeting reports, virtual fax sending/receiving, email to fax
Web UI: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish

0170600269

Σε Απόθεμα

Grandstream UCM6302A

339.20 €

ΚατασκευαστήςGrandstream ΕπιπρόσθεταEnhanced reliability with support for Hot Standby High-Availability and local dual deploymentAdvanced security protection with secure boot, unique certificate and random default password to protect calls and accountsFree Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devicesAPI available for third-party integrations, including CRM and PMS platformsZero configuration provisioning of Grandstream SIP endpointsAutomated NAT firewall traversal service facilitates secure remote connectionsSupports Full-Band Opus voice codec, jitter resilience up to 50% packet lossBased on Asterisk* version 16 open source telephony operating systemSupports up to 500 users and up to 75 concurrent callsBuilt-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpointsThree Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT routerCompatible with GDMS for cloud setup, management, and monitoring

S100

Σε Απόθεμα

Yeastar S100 VoIP Τηλεφωνικό Κέντρο

796.00 €

ΚατασκευαστήςYeastar Διαστάσεις προϊόντοςΒάρος: 2.5 kg44 x 25.2 x 4.4cm (L x W x H) ΕπιπρόσθεταMax FXS Port: 16Users: 100 (up to 200)Range: -20°C to 65°C, -4°F to 149°F, Humidity: 10-90% non-condensingTransport Protocol: UDP, TCP, TLS, SRTPEthernet Interfaces: 2 × 10/100/1000 MbpsMax FXO/BRI Ports: 16Power Consumption : 5.5-41.2 WDTMF: In-band, RFC4733, RFC2833, SIP INFOMax Concurrent Calls: 30 (up to 60)Power: AC 100-240V 50/60HZ 1.8A maxVoicemail : 10000 min (expandable)Codec: G711 (alaw/ulaw), G722, G726, G729A, GSM, Speex, ADPCM, iLBC, H263, H263P, H264, MPEG4Max 4G Port (Data): 1Operation Range: 0°C to 40°C, 32°F to 104°F; StorageApp Center: Linkus Mobile Client, Phone Provisioning, LDAP Server, Conference Panel, VPN Server, Billing App, Hotel App, char utile h+ Integration, QueueMetrics IntegrationIP Services: Static IP, DHCP, VPN, Firewall, VLAN, DDNS, PPPoE, QoS, Static NAT, STUNMax GSM/3G/4G Ports: 6T.38 Fax/Firewall Max E1/T1/J1 Ports: 2Hot StandbyMounting: 1U Rack-mountVoIP Trunks: 100USB: 1External Storage: SD Card

UCM6301

Σε Απόθεμα

Grandstream UCM6301

300.60 €

ΚατασκευαστήςGrandstream Διαστάσεις προϊόντοςΒάρος: 715g270mm(L) x 175mm(W) x 36mm(H) ΕπιπρόσθεταCall Center: Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/ work-load, in-queue announcementPeripheral Ports: 1*USB 3.0, 1*SD card interfaceCustomizable language pack to support any other languagesVideo Codecs: H.264, H.263, H263+, H.265, VP8Media Encryption: SRTP, TLS, HTTPS, SSH, 802.1XAPI: Full API available for third-party platform and application integrationOperating: 32 - 113ºF / 0 ~ 45ºC, Humidity 10 - 90% (non-condensing)Customizable IVR/voice prompts: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, NederlandsUniversal Power Supply: Input: 100 ~ 240VAC, 50/60Hz; Output: DC 12V, 1.5AMaximum Call Capacity: Users: 500 Concurrent calls (G.711): 75 Max concurrent SRTP calls (G.711): 50Analog Telephone FXS Ports: 1 RJ11 PortNetwork Pro/os: 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®Customizable Auto Attendant: Up to 5 layers of IVR (Interactive Voice Response) in multiple languagesTelephony Operating System: Based on Asterisk version 16LCD Display: 320x240 color LCD with touch screen for Shortcut Keys and Scroll Bar4 Video Conference rooms and up to 20 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus)Network Interfaces: Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+Caller ID: Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTTReset Switch: Yes, long press for factory reset and short press for rebootWeb UI: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, TurkishVideo conference, eventlist, feature codes, busy camp-on/ call completion, voice control, post-meeting reports, virtual fax sending/receiving, email to faxNAT Router: YesCall wakeup, SCA, BLF, voicemail to email, fax to email, speed dial, call back, dial by name, emergency call, call follow me, blacklist/whitelist, voice conferenceQoS: Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoSDTMF Methods: In-band audio, RFC2833, and SIP INFOVoice Conference: Up to 75 parties (G.711)Mounting: Wall mount & DesktopCall 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/intercPSTN Line FXO Ports: 1 RJ11 PortDisconnect Methods: Busy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current DisconnectVoice 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