Yeastar S20 VoIP Τηλεφωνικό Κέντρο

Αρχική/Phones & Tablets/Τηλεφωνία/Τηλεφωνικά Κέντρα & Συσκευές/Τηλεφωνικά Κέντρα/Yeastar S20 VoIP Τηλεφωνικό Κέντρο

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ΚΩΔΙΚΟΣ: 0170600545

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Σε Απόθεμα

424.00 €

Ποσότητα:

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Online διαθεσιμοτητα προϊοντων ανα αποθηκη

Αθήνα

Τζώρτζ 30-32

Κεντρικές Αποθήκες

Περιγραφή

Yeastar S20 VoIP Τηλεφωνικό Κέντρο. Σχεδιασμένο για μικρές και μεσαίες επιχειρήσεις, το S20 VoIP PBX διακρίνεται για την ευελιξία, την ποιότητα και την αξιοπιστία του, αλλά και για τα πλούσια χαρακτηριστικά τηλεφωνίας που διαθέτει. Υποστηρίζει έως 20 χρήστες και 10 ταυτόχρονες κλήσεις.

Το μοντέλο S20 της Yeastar είναι ένα επεκτάσιμο και ευέλικτο VoIP PBX, ιδανικό για μικρές επιχειρήσεις. Υποστηρίζει έως 20 χρήστες και 10 ταυτόχρονες κλήσεις. Διαθέτει δύο υποδοχές για Voice Modules και υποστηρίζει έως 4 FXS/FXO/BRI ports, έως 1 x GSM port και έως 20 VoIP trunks. Όπως όλα τα μοντέλα της σειράς S-VoIP PBX, το S20 ενσωματώνει όλες τις λειτουργίες IP Τηλεφωνίας, όπως Προαπάντηση (IVR), Αυτόματο Τηλεφωνητή (Voice mail), Αυτόματη Καταγραφή Κλήσεων (Call Recording), Απομακρυσμένα ή/και Φορητά εσωτερικά, Αναγνώριση Κλήσεων, Τηλεδιασκέψεις, Fax to email, Ring Group, Δρομολόγηση και Μεταφορά Κλήσης, Queuing κ.α.

Υποστηριζόμενες εφαρμογές: Linkus Mobile Client, Phone Provisioning, LDAP Server, Conference Panel, VPN Server, Billing App, Hotel App, ενσωμάτωση char utile h+, ενσωμάτωση QueueMetrics.

Χαρακτηριστικά

Κατασκευαστής:

Yeastar

Διαστάσεις προϊόντος:

16 x 16 x 3cm (L x W x H)
Βάρος: 0.3 kg

Επιπρόσθετα:

App Center: Linkus Mobile Client, Phone Provisioning, LDAP Server, Conference Panel, VPN Server, Billing App, Hotel App, char utile h+ Integration, QueueMetrics Integration
Codec: G711 (alaw/ulaw), G722, G726, G729A, GSM, Speex, ADPCM, iLBC, H263, H263P, H264, MPEG4
DTMF: In-band, RFC4733, RFC2833, SIP INFO
Environment: Operation Range: 0°C to 40°C, 32°F to 104°F, Storage Range: -20°C to 65°C, -4°F to 149°F, Humidity: 10-90% non-condensing
Ethernet Interfaces: 2 × 10/100 Mbps
External Storage: TF Card
Firewall/T.38 Fax/Hot Standby
IP Services: Static IP, DHCP, VPN, Firewall, VLAN, DDNS, PPPoE, QoS, Static NAT, STUN
Max 4G Port (Data): 1
Max Concurrent Calls: 10
Max FXO/BRI Ports: 4
Max FXS Ports: 4
Max GSM/3G/4G Ports: 1
Mounting: Desktop & Wall-mount
Power Consumption: 1.8-10.6 W
Power: DC 12V 1A
Protocol: SIP (RFC3261), IAX2
Transport Protocol: UDP, TCP, TLS, SRTP
Users: 20
Voicemail: 5000 min (expandable)
VoIP Trunks: 20

Εγγύηση:

2 Έτη

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)Power: AC 100-240V 50/60HZ 1.8A maxEthernet Interfaces: 2 × 10/100/1000 MbpsMax FXO/BRI Ports: 16Operation Range: 0°C to 40°C, 32°F to 104°F; StorageDTMF: In-band, RFC4733, RFC2833, SIP INFOVoicemail : 10000 min (expandable)Max Concurrent Calls: 30 (up to 60)Codec: G711 (alaw/ulaw), G722, G726, G729A, GSM, Speex, ADPCM, iLBC, H263, H263P, H264, MPEG4Max 4G Port (Data): 1App Center: Linkus Mobile Client, Phone Provisioning, LDAP Server, Conference Panel, VPN Server, Billing App, Hotel App, char utile h+ Integration, QueueMetrics IntegrationT.38 Fax/FirewallIP Services: Static IP, DHCP, VPN, Firewall, VLAN, DDNS, PPPoE, QoS, Static NAT, STUNMax GSM/3G/4G Ports: 6 Max E1/T1/J1 Ports: 2Range: -20°C to 65°C, -4°F to 149°F, Humidity: 10-90% non-condensingUSB: 1Hot StandbyMounting: 1U Rack-mountVoIP Trunks: 100Power Consumption : 5.5-41.2 WTransport Protocol: UDP, TCP, TLS, SRTPExternal Storage: SD Card

0170600262

Σε Απόθεμα

Grandstream UCM6302

395.80 €

ΚατασκευαστήςGrandstream Διαστάσεις προϊόντοςUnit Weight: 715g270mm(L) x 175mm(W) x 36mm(H) ΕπιπρόσθεταCustomizable 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 lossPeripheral Ports: 1*USB 2.0, 1*USB 3.0, 1*SD card interfaceNetwork Interfaces: Three self-adaptive Gigabit ports (switched, routed or dual mode) with PoE+Caller 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 terminationOperating: 32 - 113ºF / 0 ~ 45ºC, Humidity 10 - 90% (non-condensing)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 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 PortsNAT Router: YesAPI: 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) QoSMultiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/ work-load, in-queue announcementAnalog Telephone FXS Ports: 2 RJ11 PortsNetwork 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 rebootMounting: Wall mount & Desktop6 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 16Maximum Call Capacity: Users: 1000/Concurrent calls (G.711): 150/Max concurrent SRTP calls (G.711): 100paging/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