Grandstream UCM6302A

Αρχική/Phones & Tablets/Τηλεφωνία/Τηλεφωνικά Κέντρα & Συσκευές/Τηλεφωνικά Κέντρα/Grandstream UCM6302A

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ΚΩΔΙΚΟΣ: 0170600269

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Σε Απόθεμα

339.20 €

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Online διαθεσιμοτητα προϊοντων ανα αποθηκη

Αθήνα

Τζώρτζ 30-32

Κεντρικές Αποθήκες

Περιγραφή

UCM6302Α supports up to 500 users and 75 concurrent calls. Integrated 2 PSTN trunk FXO ports, 2 analog telephone FXS ports. Audio only.

Χαρακτηριστικά

Κατασκευαστής:

Grandstream

Επιπρόσθετα:

Advanced security protection with secure boot, unique certificate and random default password to protect calls and accounts
API available for third-party integrations, including CRM and PMS platforms
Automated NAT firewall traversal service facilitates secure remote connections
Based on Asterisk* version 16 open source telephony operating system
Built-in Instant Messaging (IM), Audio Conferencing & Web Meetings platform that supports access from computers, mobile devices, and SIP endpoints
Compatible with GDMS for cloud setup, management, and monitoring
Enhanced reliability with support for Hot Standby High-Availability and local dual deployment
Free Wave App allows easy voice & Instant Messaging (IM) communications using desktops, Web, and Android/ iOS devices
Supports Full-Band Opus voice codec, jitter resilience up to 50% packet loss
Supports up to 500 users and up to 75 concurrent calls
Three Gigabit auto-sensing RJ45 network ports with integrated PoE+ and support NAT router
Zero configuration provisioning of Grandstream SIP endpoints

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) ΕπιπρόσθεταVoIP Trunks: 100Max FXS Port: 16External Storage: SD CardRange: -20°C to 65°C, -4°F to 149°F, Humidity: 10-90% non-condensingUSB: 1Users: 100 (up to 200)Max FXO/BRI Ports: 16Ethernet Interfaces: 2 × 10/100/1000 MbpsPower Consumption : 5.5-41.2 WTransport Protocol: UDP, TCP, TLS, SRTPMax Concurrent Calls: 30 (up to 60)DTMF: In-band, RFC4733, RFC2833, SIP INFOPower: AC 100-240V 50/60HZ 1.8A maxMax 4G Port (Data): 1Codec: G711 (alaw/ulaw), G722, G726, G729A, GSM, Speex, ADPCM, iLBC, H263, H263P, H264, MPEG4Operation Range: 0°C to 40°C, 32°F to 104°F; StorageVoicemail : 10000 min (expandable)IP Services: Static IP, DHCP, VPN, Firewall, VLAN, DDNS, PPPoE, QoS, Static NAT, STUNApp Center: Linkus Mobile Client, Phone Provisioning, LDAP Server, Conference Panel, VPN Server, Billing App, Hotel App, char utile h+ Integration, QueueMetrics IntegrationMax GSM/3G/4G Ports: 6 Max E1/T1/J1 Ports: 2Mounting: 1U Rack-mountHot StandbyT.38 Fax/Firewall

0170600262

Σε Απόθεμα

Grandstream UCM6302

395.80 €

ΚατασκευαστήςGrandstream Διαστάσεις προϊόντοςUnit Weight: 715g270mm(L) x 175mm(W) x 36mm(H) ΕπιπρόσθεταNetwork 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®Reset Switch: Yes, long press for factory reset and short press for rebootNetwork Interfaces: Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+6 Video Conference rooms and up to 30 parties with 1080p, assuming 4 video feeds + 1 screen sharing (H.264 & Opus)Customizable language pack to support any other languagesTelephony Operating System: Based on Asterisk version 16Operating: 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/whitelistCustomizable 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, 1.5ALEC 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 lossNAT Router: YesPeripheral Ports: 1*USB 2.0, 1*USB 3.0, 1*SD card interfaceCaller ID: Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTTVideo Codecs: H.264, H.263, H263+, H.265, VP8LCD Display: 320x240 color LCD with touch screen for Shortcut Keys and Scroll BarPolarity Reversal/Wink: Yes, with enable/disable option upon call establishment and terminationMultiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/ work-load, in-queue announcementCall 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 groupvoice conference, video conference, eventlist, feature codes, busy camp-on/ call completion, voice control, post-meeting reports, virtual fax sending/receiving, email to faxPSTN Line FXO Ports: 2 RJ11 PortsMounting: Wall mount & DesktopAPI: Full API available for third-party platform and application integrationWeb UI: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, TurkishQoS: Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoSMaximum Call Capacity: Users: 1000/Concurrent calls (G.711): 150/Max concurrent SRTP calls (G.711): 100Analog Telephone FXS Ports: 2 RJ11 Ports

UCM6301

Σε Απόθεμα

Grandstream UCM6301

300.60 €

ΚατασκευαστήςGrandstream Διαστάσεις προϊόντος270mm(L) x 175mm(W) x 36mm(H)Βάρος: 715g ΕπιπρόσθεταMounting: 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.38Call 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)