Grandstream UCM6301

Αρχική/Phones & Tablets/Τηλεφωνία/Τηλεφωνικά Κέντρα & Συσκευές/Τηλεφωνικά Κέντρα/Grandstream UCM6301

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ΚΩΔΙΚΟΣ: UCM6301

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Σε Απόθεμα

300.60 €

Ποσότητα:

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Online διαθεσιμοτητα προϊοντων ανα αποθηκη

Αθήνα

Τζώρτζ 30-32

Κεντρικές Αποθήκες

Περιγραφή

UCM6301 supports up to 500 users and 75 concurrent calls. Integrated 1 PSTN trunk FXO ports, 1 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)
Βάρος: 715g

Επιπρόσθετα:

4 Video Conference rooms and up to 20 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus)
Analog Telephone FXS Ports: 1 RJ11 Port
API: Full API available for third-party platform and application integration
Call Center: Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/ work-load, in-queue announcement
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/interc
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
Caller ID: Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT
Customizable Auto Attendant: Up to 5 layers of IVR (Interactive Voice Response) in multiple languages
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
Disconnect Methods: Busy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect
DTMF Methods: In-band audio, RFC2833, and SIP INFO
LCD Display: 320x240 color LCD with touch screen for Shortcut Keys and Scroll Bar
Maximum Call Capacity: Users: 500 Concurrent calls (G.711): 75 Max concurrent SRTP calls (G.711): 50
Media Encryption: SRTP, TLS, HTTPS, SSH, 802.1X
Mounting: Wall mount & Desktop
NAT Router: Yes
Network Interfaces: Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+
Network 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®
Operating: 32 - 113ºF / 0 ~ 45ºC, Humidity 10 - 90% (non-condensing)
Peripheral Ports: 1*USB 3.0, 1*SD card interface
PSTN Line FXO Ports: 1 RJ11 Port
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
Video conference, eventlist, feature codes, busy camp-on/ call completion, voice control, post-meeting reports, virtual fax sending/receiving, email to fax
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
Voice Conference: Up to 75 parties (G.711)
Web UI: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish

S100

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Yeastar S100 VoIP Τηλεφωνικό Κέντρο

788.60 €

ΚατασκευαστήςYeastar Διαστάσεις προϊόντοςΒάρος: 2.5 kg44 x 25.2 x 4.4cm (L x W x H) ΕπιπρόσθεταMounting: 1U Rack-mountVoIP Trunks: 100T.38 Fax/FirewallExternal Storage: SD CardMax FXS Port: 16Users: 100 (up to 200)Range: -20°C to 65°C, -4°F to 149°F, Humidity: 10-90% non-condensingEthernet Interfaces: 2 × 10/100/1000 MbpsUSB: 1Max FXO/BRI Ports: 16Power Consumption : 5.5-41.2 WDTMF: In-band, RFC4733, RFC2833, SIP INFOTransport Protocol: UDP, TCP, TLS, SRTPMax Concurrent Calls: 30 (up to 60)Power: AC 100-240V 50/60HZ 1.8A maxCodec: 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 IntegrationVoicemail : 10000 min (expandable)IP Services: Static IP, DHCP, VPN, Firewall, VLAN, DDNS, PPPoE, QoS, Static NAT, STUNMax GSM/3G/4G Ports: 6 Max E1/T1/J1 Ports: 2Hot Standby

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Σε Απόθεμα

Grandstream UCM6302

395.80 €

ΚατασκευαστήςGrandstream Διαστάσεις προϊόντοςUnit Weight: 715g270mm(L) x 175mm(W) x 36mm(H) ΕπιπρόσθεταQoS: Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoSNetwork 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®Analog Telephone FXS Ports: 2 RJ11 PortsMounting: Wall mount & DesktopReset Switch: Yes, long press for factory reset and short press for rebootCustomizable language pack to support any other languages6 Video Conference rooms and up to 30 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus)Maximum Call Capacity: Users: 1000/Concurrent calls (G.711): 150/Max concurrent SRTP calls (G.711): 100Telephony Operating System: Based on Asterisk version 16Customizable IVR/voice prompts: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, Nederlandspaging/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/whitelistLEC 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 lossNetwork Interfaces: Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+Universal Power Supply: Input: 100 ~ 240VAC, 50/60Hz; Output: DC 12V, 1.5ACaller ID: Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTTPeripheral Ports: 1*USB 2.0, 1*USB 3.0, 1*SD card interfaceLCD Display: 320x240 color LCD with touch screen for Shortcut Keys and Scroll BarOperating: 32 - 113ºF / 0 ~ 45ºC, Humidity 10 - 90% (non-condensing)Video Codecs: H.264, H.263, H263+, H.265, VP8Call 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 groupPolarity Reversal/Wink: Yes, with enable/disable option upon call establishment and terminationNAT Router: Yesvoice conference, video conference, eventlist, feature codes, busy camp-on/ call completion, voice control, post-meeting reports, virtual fax sending/receiving, email to faxAPI: Full API available for third-party platform and application integrationPSTN Line FXO Ports: 2 RJ11 PortsMultiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/ work-load, in-queue announcementWeb UI: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish

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Yeastar P570 Τηλεφωνικό Κέντρο

2,242.40 €

ΚατασκευαστήςYeastar Διαστάσεις προϊόντος44 x 25.2 x 4.4cm (L x W x H) Βάρος: 2.38 KG ΕπιπρόσθεταEthernet Interfaces: 2 x (10/100/1000 Mbps)Video Codec: H263, H263P, H264, MPEG4, VP8Power Supply: AC 100-240V 50/60HZ 1.5A maxMax FXS Ports: 16Expandable D30: 2Voice Codec: G711 (alaw/ulaw), G722, G726, G729A, GSM, Speex, ADPCM, iLBCTransport Protocol: UDP, TCP, TLS, SRTPMax GSM/3G/4G Ports: 6Form Factor: 1U RackmountBase / Max Call Center Agents: 300 / 500NFC Read/WriteHard Disk: 1 SATA (Up to 2TB)Base Users / Max Users: 300 / 500Max Concurrent Calls: 60 / 120Company Contacts: 500,000Max E1/T1/J1 Ports: 2DTMF: In-band, RFC4733, RFC2833, SIP INFOUSB: 1 (Up to 2TB)Operation Range: 0°C to 40°C, 32°F to 104°F StorageMax FXO/BRI Ports: 16