SIPPBX X8200
SIP based IP-PBX

 

  • Standard SIP RFC 3261, TCP, UDP, TLS compliance IP-PBX
  • IPv6/IPv4 Dual Stack
  • Support NAT voice and video calls between WAN and LAN
  • Rich Telephony PBX Features
  • Voice Hacker attack detection and prevention
  • Support Auto-Attendant and Voice Mail
  • Billing Feature built-in
  • Support SIP Trunk and NAT Transparency
  • CPE device Auto Provision
  • IP Recording, CTI/Call Center: Optional Feature

Introduction

Information
X8200 Image

SIPPBX X8200 is an advance version of SIP based IP-PBX which
has different design with existing SIPPBX 6200S and 6200GS.
It provides Linux OS and avoids voice hacker from attacking.
There are rich features and optional features for user to upgrade
once they need them.

Specification

  • Interface:
    • Ethernet ports (RJ-45, 10/100/1000 Base-T)
      • 1-WAN port, for connecting to Internet
      • 1-LAN port for connecting to Private Network
    • AC power Line input outlet
  • System Capability:
    • Max Extensions: 2,000
    • Max Concurrent Calls: 1,000
    • Max NAT Resources: 1,000
    • Max Universal Resources: 256
  • System Services:
    • Multiple SIP Domains
    • Automatic Audio/Video NAT Traversal
    • SIP Proxy / Registrar
    • Permanent & Dynamic Contact
    • SIP Trunk / Voice Router
    • WAN/LAN Simultaneously
    • RADIUS Billing
    • Extension/Device Monitoring
    • Device Allowance Control
    • Session Timer Call Validation
    • INVITE-Initiated Dialog Event (RFC4235)
    • Missed Call Email Notice
    • Multi-Office / Branch
    • CPE Auto Provision
  • Voice Hacker attack and protection:
    • SIP Attack Detection / IP Blocking
    • SIP User Device Restriction
    • Country/IP Network Lock
    • Enhanced Password Option
    • Black Routing List
    • CAPTCHA to protect Web Login
    • Web Access Log
  • IP Protocol:
    • SIP RFC 3261 Compliance with high efficient stack
    • Support IPv4 (RFC791) and IPv6 dual mode
    • UDP, TCP, TLS, RTP, RTCP
  • Flexible Routing Plan:
    • Group Based Routing
    • Time of Day / Week Day Routing
    • Preference Routing
    • Round Robin Routing
    • Load Balancing Routing
    • Broadcast Routing
    • Unavailable Redirect
    • ENUM Routing
    • Black List Reject
  • Auto Attendant (AA) Services:
    • Support Multi-Language
    • Support Multi-Offices
    • Graphic Attendant Flow Editor
    • Incoming Calls Limitation
    • Office Oriented Call Flow
    • Up to 3 Time Segment
    • Working/Off-Time/Holiday Operator
    • Working/Off-Hours Flow
    • Priority/Holiday Flow
    • Black List Filter & Flow
    • Access to Voice Mail
    • Outgoing Call (Password Protect)
  • Voice Mail Services:
    • Message Detail
    • Incoming Call Limitation
    • Personal Greeting
    • Multiple Language Support (Chinese & English)
    • Message Waiting Indication (MWI), RFC 3842
    • Voice mail to Email (MP3)
    • Access Voice Mail via Web
    • Access Voice Mail via Phone
  • Conference Bridge:
    • Up-to 16 parties conference room
    • Multi Language
    • Incoming Call Limitation
    • Ad-Hoc Conference
    • Meet Me Conference
      • Host/Participant Password
      • Join/Quit Announcement
    • Dial out Conference (Option)
      • Host Password
      • Dynamic Participant List calling
      • Predefine Participant List
      • Join/Quit Announcement
      • Unavailable Announcement
      • Add Participant within Conference
  • Broadcasting Services:
    • Up-to 64 parties Broadcasting Target
    • CPE Auto Answer to Speaker by SIP
    • Stat/Stop Tone Notice
  • Billing Feature:
    • Charge Division
    • Top Usage Users report
    • Top Prefix Usage Report
    • Prefix Summaries Report
    • Division Billing Report
    • Division Wide Tariff Plan
      • Charge Unit
      • Charge Amount
    • Call History Detail Report
      • Calling/Called Number
      • Call Duration
      • Call Type
      • Call Connect/Disconnect time
      • SIP Call ID
      • SIP URI
      • Source/Destination IP Address
  • Audio/Video Codecs:
    • G.711 A-law and μ-law
    • G.729A
    • G.723
    • G.722
    • GSM 6.10 (full rate)
    • H.263/H.264 Video Codec Pass-Thru
    • MPEG4 Pass-Thru
  • Telephony PBX features:
    • Call Transfer
    • Call Forward
    • Call Forwarded Notice
    • Call Screening (Call Restriction)
    • Caller ID Privacy
    • Call Waiting
    • Call Hold
    • Call Pickup (Global, Group)
    • Specified Call Pickup
    • Find Me
    • Abbreviate Dialing
    • Do Not Disturb (DND)
    • Missed Call Notify by Email
    • ANI Replacement (Calling Number)
    • Call Return
    • Hide ANI/Show ANI Selection
    • Call Park/Retrieve
    • Call Camp on
    • Display Name Replacement
    • PSTN Number (Caller ID number replacement)
    • Ring PSTN & IP Device Simultaneously
    • Reject Anonymous Call
    • Busy Lamp Filed (RFC 4235)
  • MANAGEMENT:
    • Multi-Language
    • Division Manager
    • Web Provision Access Log
    • Easy Web GUI (Http/Https)
    • On-Line Manual
    • Customize Web Access Rights
    • System Alert by Syslog / Email
    • Real Time System Monitor & Tracing
    • System Statistic reports
    • SOAP Provision Interface
  • Smart Calling Feature: (Option)
    • Smart management via Hand-Held device
    • Support Android and iPhone browser
    • Forward to Smart Phone
    • Click to Call (Call To)
    • Create Outgoing Conference
    • Monitor Meet Me Conference
    • Conference Control
      • Add Participant
      • Remove Participant
      • Speak Request
  • Voice Logging feature: (Option)
    • Max Logging Channels: 512
    • System Service:
      • Dual IPv6/IPv4 Voice Logging
      • Extension Recording
      • Trunk/Gateway Recording
      • Programmable Recording Target
      • Recording on Demand
      • External MYSQL
      • External NAS Storage
    • Voice Codec Decode:
      • G.711
      • G.722
      • GSM
      • iLBC
    • Archiving Format:
      • MP3 Encoded File
      • Separate Caller/Called Tracks
      • CBR Encode (32K – 256K)
      • VBR Encode
      • Optional AES Encryption
    • Voice Logging Report:
      • Logging Target
      • Call Status
      • Call Start Time
      • Call Stop Time
      • SIP Call ID
      • RTP Information
      • Call Forward Information
  • Others Optional features:
    • Dial Out Conference
    • Web Caller
    • CTI / Call Center application
    • Hotel IP-PBX (Other Solution)
  • Environmental:
    • Operating Temp. & Humidity
      • Temp.: 0°C~45°C (32°F~113°F)
      • Humidity: 10%~90% relative humidity non-condensing
  • Hardware Specification:
    • CPU: Intel Pentium G2120, 3.1 GHZ
    • RAM: 2GB
    • Had Disk: SSD 60GB or above
    • OS: Linux CentOS 6 / RHEL 6 (64 bits)
  • Power Consumption:
    • 300 Watts
    • Input Voltage: AC 100V – 240V .
  • Packing:
    • Physical Dimension: 1U, 19-inch chassis.
      • 48.4×4.4×45 cm
    • Gross Weight: 12.5 Kgms
    • Packing Dimension (one unit): 61.5 x 60 x 20 cm
  • Approvals:
    • CE, FCC, LVD and RoHS
  • Country of origin:
    • Made in Taiwan
  • Packing Accessories
    • SIPPBX X8200 x 1 unit
    • AC Power Cable x 1 pcs
  • Warranty:
    • One year
 
 

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