説明

• U120 VoIP PBX U120 is a VoIP PBX telephone formula designed for small and medium sized enterprises to boost productivity, improve performance and decrease telephony and operational charge As a converged platform offering varied connectivity to all networks like FXO (CO), and VoIP/SIP, supporting as much as 60 users, U120 permits businesses benefit from state-of-art technology and enterprise class features with small investments, delivers excessive performance and superior quality to satisfy the communique needs of in the present day and day after today Notice: Current mannequin don&;t assist 4G/GSM module! VoIP Solution for SMEs 60 SIP users, 15 concurrent calls 2 FXO ports IP/SIP Failover Multiple SIP trunks Fax over IP (T and Pass-through) Built-in VPN TLS / SRTP security WiFi hot spots Low cost yet excessive effective to construct your call formula Suitable for many of small office, apartment, hotel and many others Full VoIP Features Call Recording Voicemail Call forking Auto CLIP Fax to Email Black/White list Auto Attendant Conference Call Reliable Quality SIP PBX, VoIP Server, Call Center Easy Management Configuration Backup & Restore Advanced Debug tools on cyber web interface Intuitive Web interface Multiple language support Multiple language assist Cloud Management System Specification: PBX Services Voice Capabilities Maintenance SIP Trunks and Extensions VoIP Protocols: SIP over Web GUI Configuration Fax over IP (T and Pass-through) UDP/TCP,RTP/RTCP Telnet Management Call Forward G/μ law,G, G/B Codecs Configuration Restore/Backup Call Waiting / Hold Silence Suppression Multiple Languages Supported Call Transfer Comfort Noise Generator(CNG) HTTP/HTTPS/FTP Firmware Upgrade Call Pickup Voice Activity Detection(VAD) Auto Provision Call Monitor / Bargin in G Echo Cancellation CDR Report and Export Hotline Dynamic Jitter Buffer Syslog Do-not-disturb Adjustable Gain Control Ping and Tracert three Way Conference Automatic Gain Control (AGC) Traffic Statistics: TCP,UDP,RTP Ring Group/Route Group Call Progress Tones: Dial Tone, Ring Back Network Capture Call Queue Tone, Busy Tone NTP Paging NAT: STUN/Rport 2G/VoLTE Caller/Called Number Manipulation DTMF: RFC2833/Signal/Inband 1* SIM Slot Routing Based on Time Period SIP TLS/SRTP/VOS encrypt 1,3V SIM Interface Routing Based on Number FXS / FXO SMS/USSD Prefixes(Support Regex) Connector type: RJ11 Antenna: SMA (Male) Connector, Routing Based on Multi Source Caller ID: DTMF, FSK ITU-T V and 3 Failover Routing FSK Bellcore GSM or VoLTE on Demand SMS Route Polarity Reversal Frequency Bands: Multi-level IVR Busy Tone Detection ( FXO) 2G: 850/900/1800/1900MHz(GSM) Auto-attendant Function Environmental LTE(VoLTE and Data): CDRS/Signal Trace Loop Back Power Supply: 12VDC, 1A Back forward Compatible with 2G/3G Voicemail/Voice Recording Power Consumption: <10W Type E Frequency Bands: Up to 60 SIP Extensions Operating Temperature 0 °C ~ 40 °C FDD:B1/B3/B5/B7/B8/B20 Up to fifteen Concurrent Calls Storage Temperature: -20 °C ~80 °C TDD: B38/B40/B41 Open NATS Cloud Service Humidity: 10%-90% Non-Condensing WCDMA: B1/B5/B8 Event Report via HTTP/NATS Dimensions (W/D/H): GSM:900/1800MHz Voicemail to Email 178×98×28mm Type AU Frequency Bands: Physical Interfaces Unit Weight: 0 FDD:B1/B2/B3/B4/B5/B7/B8/B28 FXS/FXO Ports: TDD:B40 WCDMA: B1/B2/B5/B8 2* FXO Hybrid GSM: 850/900/1800/1900 1* GSM or VoLTE(Optional to buy) Network USB Interface: 1* USB2 Access Rules Ethernet Interfaces: DHCP Client/ DHCP Server 2* 10/100M, Base-T, RJ45 Port Forward, DMZ, Static Routes Wi-Fi: 802 b/g/n ,2 ,1T1R PPTP/L2TP/OpenVPN Server/Client 1* SD card slot Indicators: FXS, FXO, USB,SD,PWR/RUN

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