• Designed for SME and SOHO users;
• Compact and lightweight;
• Provides FXS, FXO, GSM/LTE and Φ3.5 audio interface;
• Support up to 15 concurrent calls;
• Features fast installation, easy deployment, and high reliability.

OpenVox UC120 Series IPPBX is a new generation Unified Communication terminal equipment that combines voice and data. It is compact and lightweight, and provides following voice interfaces: FXS, FXO, GSM/LTE and Φ3.5 audio interface. It is compatible with multiple service platforms and terminals and can seamlessly connect to VoIP networks, traditional telephone networks (PSTN) and mobile networks (PLMN), and provide diverse converged communications solutions.

On the one hand, the UC120 connects to traditional telephones, faxes and traditional analog PBXs via a standard voice interface; on the other hand, it uses the standard SIP protocol and is compatible with most IPPBX, soft switch and SIP-based network platforms. At the same time, the UC120 supports four GSM bands to meet the requirements of global mobile communication networks.

As one of the product features, the product combines a Φ3.5 audio port, which can be directly connected to a computer headset for voice calls.

In addition, it has high-speed data capabilities, allowing users to high-speed Internet access via the WAN (LAN1) / LAN (LAN2) interface.

UC120 can be used as a personal communication product, and can also be used as a centralized communication product for SME, providing high-speed Internet access, enterprise voice communication, and enterprise short message transmission and reception.

• Convergence of telephone calls, recordings, messages and instant messages
• Plug and Play IP Phone;
• Corporate headquarters and branch voice networking
• Commercial communication between mobile phone and extension anytime, anywhere
• Open Application Programming Interface(API)
• Compatible with IMS and Asterisk service platforms

• FXS 1
• FXO: 1
• GSM/LTE: 1
• Network Interface: 2 10/100 Base-T RJ45

• 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

• GSM: 850/900/1800/1900MHz
• LTE: LTE FDD: B1/B3/B5/B8LTE 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

• 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

• Interface Type:RJ11
• Caller ID Detection: FSK, DTMF
• Reversed-Polarity Detection
• Delayed Response Off-hook
• Busy Tone Detection
• No Current Hang-up Detection

• Interface Type: RJ11
• 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

• Simple and convenient configuration via Web GUI
• CLI Management Config
• Support configuration flies backup and upload
• Support Chinese and English page
• Firmware Update by HTTP/TFTP
• Auto Provision Update
• Modify Password via Web & Telnet
• CDR Query & Export
• Syslog Query & Export
• Ping and Tracer Test
• Traffic Statistics: TCP, UDP, RTP
• Network Capture/Network Quality Test
• Automatic Time synchronization