• 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 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