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UC 120 IP PBX
- Model: UC 120 an advanced all-in-one IP PBX system for modern business communication.
- Design: Engineered for seamless integration of voice, video, and data into a unified platform.
- Functionality: Ensures efficient call management with multi-layer security and HD voice quality.
- Connectivity: Fully SIP-compatible, enabling smooth integration with VoIP networks.
- Performance: Delivers crystal-clear communication and robust system reliability.
GSM
Voice
FXO/FXS
SIP Trunk
SIP
HD Voice Quality
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Specifications
UC120 Series IPPBX is a new generation Unified Communication terminal equipment designed for SME and SOHO users. It is compact and lightweight, providing the following voice interfaces: FXS, FXO, GSM/LTE, and Φ3.5 audio interface.
The UC120 Series integrates functions such as IP telephony, voice, and recording. It is compatible with multiple service platforms and terminals, enabling seamless connections to VoIP networks, traditional telephone networks (PSTN), and mobile networks (PLMN), offering diverse converged communication solutions.
UC120 IPPBX also features fast installation, easy deployment, and high reliability, providing a new experience of mobile office and communication to the enterprise.
UC120 Series IPPBX Product Model UC120-1S1O1G, UC120-2S1G FXO 1 FXS 1 GSM 1 LTE N SIP Register License 32 SIP Concurrency (G.711A/U) 15 SIP Concurrency (G.729) 5 Audio Interface (Pair) 1 Weight 0.5 kg Power Supply 12VDC, ≥1.5A Maximum Power Consumption 12W Dimension (W/D/H) 160 x 105 x 24 mm WAN (LAN1 Port) 10/100 Base-T RJ45 LAN (LAN2 Port) 10/100 Base-T RJ45 Micro SD Card Slot 1 Operating Temperature 0℃ ~ 45℃ Operating Humidity 10% ~ 90% Non-condensing Storage Temperature Range -20℃ ~ 80℃
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Features
VoIP Features
- VoIP Protocols: SIP over UDP/TCP/TLS, SDP, RTP/SRTP, PPTP VPN
- Supported Codecs: G.711a/μ law, G.723.1, G.729A/B, GSM, G.726, G.722, SPEEX, ADPCM, iLBC
- Silence Suppression
- Comfort Noise Generator (CNG)
- Voice Activity Detection (VAD)
- Echo canceller (G.168), Maximum 128ms
- Adaptive Dynamic Buffering
- Adjustable Gain Control
- Automatic Gain Control
- Call Proceeding Tone: Dial Tone, Ring-back Tone, Busy Tone
- Support NAT Traversal
- DTMF Mode: RFC2833/Signal/Inband
- VPN clients: PPTP/OpenVPN/Zerotier/L2TP/N2N (Not available for EAEU)
- GSM: 850/900/1800/1900MHz
- LTE: LTE FDD: B1/B3/B5/B8, LTE TDD: B38/B38/B40/B41
- SIM/UIM: Each channel supports 1 SIM/UIM
- SIM Card Voltage: 1.8V, 3.0V
- Antenna: 3.0dB, SMA interface
Additional Services
- Call Forwarding (Unconditional/No Reply/Busy/Not Reachable)
- Call Waiting/Holding
- Call Transfer
- Intra-group Pickup
- Hotline
- Do Not Disturb (DND)
- Tripartite Meeting
- Interface Type: RJ11
- Caller ID Signaling: BELL, V23, V23_JP, DTMF
- Hang Up Detection: Off-hook, On-hook, Busy Tone
- Polarity Reverse
- Hooking Detection
- Ring Group
- Routes Group
- Calling/Called Number Transform
- Time Condition
- Based on Destination Routing
- Based on Source Routing
- Dial Plan
- Failover Routing
- FXO Impedance Matching
- Customizable Mult-language IVR
- Auto Attendant Function
- Local CDR Storage
