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Overview

The UC Series IPPBX comes with an asterisk-based system, the Tirasoft IPPBX software, offering not only full PBX functionality, but also a new feature that enables new stability for your unified communication systems with built-in Uninterruptible Power Supply (UPS). It can seamlessly integrate VoIP trunks and your existing PSTN lines with maximum 8 port analog connections and 2 Ethernet ports. They are developed with a wide selection of codecs and signaling protocols, including G711 (alaw/ulaw), G722, OPUS, AMR-NB/WB, SILK, G723.1 G726, G729, GSM, ADPCM, iLBC, H263, H263P, H264, VP8. Taking full advantages of open source platform, the UC Series appliances support industry standard SIP trunks, IAX2 trunks, analog PSTN trunks, and analog station trunks. Moreover, with Tirasoft up-to-date IPPBX system software, the fully-featured UC Series IPPBX is an ideal solution for the SMBs and small enterprises.


Performance

With Quad Core processors, the Nistel IPPBX can easily handle 60 concurrent calls and up to 100 extension registers in such a tiny box. To bring you clear, high fidelity and high definition audio/video calls, Nistel is built in with abundant HD voice codecs such as OPUS, AMR-NB/WB, G.722, SILK and VP8.

Features

Physical Specification

  • FXS: Up to 8, flexible selection
  • FXO: Up to 8, flexible selection
  • Network Interface: 2 10/100 Base-T RJ45

FXS

  • 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

FXO

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

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
  • AutoCLIP Routing
  • Auto Announcement with outgoing call
  • OPUS/VP8 HD Voice/Video Call
  • 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
  • High Avaibility (Hot Standby)
  • Phone Book/Announcement/Morning Call (Wake up)

Management & Maintenance

  • Simple and convenient configuration via Web GUI
  • Support configuration files backup and upload
  • 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