OpenVox UC120P IPPBX



OpenVox UC120P Series IPPBX is a new generation Unified Communication terminal equipment designed for SME and SOHO users. It is compact and lightweight, and provides following voice interfaces: FXS, FXO, GSM/LTE.

It integrates functions such as IP telephony, voice and recording. It is compatible with multiple service platforms and terminals and can seamlessly connect to VoIP networks, traditional telephone networks (PSTN) and mobile networks(PLMN), provide diverse converged communications solutions. UC120P IPPBX also features fast installation, easy deployment, and high reliability, which brings a new experience of mobile office and communication to the enterprise.


  • 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


Physical Specification

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


  • Interface Type: RJ11
  • Caller ID Signaling: BELL, V23, V23_JP, DTMF
  • Hang Up Detection: Off-hook, On-hook, Busy Tone
  • Polarity Reverse
  • Hooking Detection
  • FXS Interface High Vlotage Spotlight


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

Mobile Feature

  • GSM:900/1800MHz
  • LTE FDD: B1/B3/B5/B8
  • LTE TDD:B34/B38/B39/B40/B41

PBX Features

  • Ring Group/Routes Group
  • Calling/Called Number Transform
  •  Call Duration Limitation/Call Failure Rerouting
  • Caller ID Number Acquisition/DID Acquisition
  •  Remote Party ID/Remote Management
  •  P-Asserted-Identity/P-Preferred-Identity
  • Routing based on user privilege level/time condition/caller id number
  • Time Condition
  • Based on Destination Routing/Source Routing
  • Dial Plan
  • Failover Routing
  • FXO Impedance Matching
  • Customizable Multi-language IVR
  • Auto Attendant Function
  • Local CDR Storage
  • SIP forking (multiple SIP device registration with same sip account) / Customized SIP Fields
  • WebPhone (WebRTC)

Voice Features

  • VoIP Protocols: SIP over UDP/TCP/TLS, SDP, RTP/SRTP
  • Supported Codecs: G.711a/μ law, G.729A, GSM, G.726, G.722, iLBC, OPUS, VP8, H264
  • VPN: N2N and OpenVPN
  • 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
  • AutoCLIP Routing
  • Auto Announcement with outgoing call
  • OPUS/VP8 HD Voice/Video Call
  • TOS/QOS support for SIP and RTP
  • Call Proceeding Tone: Dial Tone, Ring-back Tone, Busy Tone
  • Support NAT Traversal
  • Supports HD broadcasting (automatic broadcasting, timed broadcasting)
  • DTMF Mode: RFC2833/Signal/Inband
  • Customized Signal Tones
  • Intra-group Pickup
  • Hotline
  • Do Not Disturb (DND)
  • Tripartite Meeting
  • T38 fax
  • Call Forwarding (Unconditional/No Reply/Busy/Not Reachable)
  • Call Waiting/Holding/Transfer/Queue/Spy
  • Permission Control/Broadcast control
  • Secretary Extension
  • High Availability (Hot Standby)
  • Phone Book/Announcement/Morning Call (Wake up)
  • Hotel interface (minibar, alarm, room cleaning, extension privilege setting)
  • PBX License Control

Management & Maintenance

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