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    Corecasys FXO-4
Corecasys FXO-4 รุ่นนี้ใช้รองรับการเชื่อมต่อคู่สายภายในระบบ PABX มีคุณสมบัติดังนี้
- รองรับการยิงตรงระหว่างอุปกรณ์ 2 ตัว (Peer to Peer) โดยไม่จำเป็นต้องมี Sip Server หรือ IPPBX
- สามารถเชื่อมโยง PABX ระหว่างสาขาถึงกันได้ ผ่านคู่สายภายใน
- มีฟังก์ชั่น Hot Line Number สามารถโทรไปยังปลายทางอัตโนมัติเมื่อมีเสียงเรียกเข้า
- รองรับการทำงานแบบ 24x7
Corecasys เราให้บริการจัดจำหน่ายผลิตภัณฑ์ต่างจากผลิตภัณฑ์ยี่ห้ออื่นเป็นอย่างมาก เพราะเราขายอุปกรณ์พร้อมบริการหลังการขาย และความรู้ต่างๆ ที่คุณจำเป็นต้องใช้งานเกี่ยวกับอุปกรณ์ของเรา ด้วยความที่เราเป็นผู้พัฒนาระบบ (SI)  เราจึงสามารถผลักดันคุณให้ใช้งานอุปกรณ์ได้จนจบความต้องการของคุณ โดยเฉพาะอย่างยิ่ง นโยบาย การให้ทดลองใช้งานผลิตภัณฑ์ยาวนานถึง 3 สัปดาห์ ยังเป็นข้อพิสูจนได้ว่า เมื่อคุณซื้ออุปกรณ์ไปแล้ว อุปกรณ์จะต้องช่วยงานคุณได้อย่างมีเสถียรภาพ ไม่ใช่เอาอุปกรณ์เข้าไปสร้างปัญหา และความวุ่นวายใจให้กับคุณ
Specification
  • Interface:
    • Ethernet port (RJ-45, 10/100 base-T)
      • 1-WAN port, connect to IP Network
      • 1-LAN port connect to PC with NAT
    • Support Bridge, NAT and Gateway mode
    • Telephony port connect to local PSTN line (RJ-11 x 4 pcs)
    • DC +12V power input Jack
    • Reset key to return Factory setting
    • LED Indicator for System, SIP and FXO status
  • IP Network connection:
    • IPv4 (RFC 791) and IPv6 Simultaneously
    • IPv6 Auto Configuration (RFC 4862)
    • IPv6 Only, IPv4 Only or dual stack
    • MAC Address (IEEE 802.3)
    • MAC Clone Setting
    • Vendor Class ID
    • IP/ICMP/ARP/RARP/SNTP
    • Static IP
    • DHCP Client (RFC 2131), WAN port
    • DHCP Server, LAN port
    • NAT Server (RFC 1631)
    • PPPoE Client
    • DDNS ( DynDNS )
    • DNS Client
    • Firewall
    • URL Filter
    • IP Filter
    • MAC Address Filter
    • Application program Filter
    • Port Filter
    • Port Forwarding (TCP, UDP or both)
    • Bandwidth Control (Download and Upload), Maximum Bandwidth priority
      setting
    • UPnP Server at LAN port
    • Behind NAT, use DMZ for NAT traversal
    • SNTP with time zone and Daylight Saving
    • TCP/UDP (RFC 793/768)
    • RTP/RTCP (RFC 1889/1890)
    • IPV4 ICMP (RFC 792),
    • TFTP Client
    • VoIP VLAN Support 802.1Q, 802.1P
    • VLAN ID Range : 2 to 4094
    • VLAN Priority : 0 to 7 (Highest Priority)
    • QoS : DiffServ (RFC 2475), TOS (RFC791, 1394)
  • SIP Protocol :
    • RFC3261 compliance
    • Support up-to 4 SIP Trunk to Register
    • SIP UDP Protocol
    • Support SIP compact Form
    • Support SIP HOLD Type: Send Only, 0.0.0.0 or inactive
    • SIP Session Timer (RFC 4028)
    • SIP Session Refresher: UAC or UAS
    • SIP Encryption
    • MD5 Digest Authentication (RFC2069/RFC2617)
    • Reliability of provision response PRACK (RFC3262)
    • Early/Delay Media support
    • Offer/Answer (RFC3264)
    • Message Waiting Indication (RFC3842)
    • Event Notification (RFC3265)
    • REFER (RFC3515)
    • Support Outbound Proxy
    • Support Primary and Backup SIP Server
    • Support STUN NAT Traversal
    • Support “rport” parameter (RFC 3581)
    • Configure SIP local Port
    • SIP QoS Type: DiffServe or QoS
    • Accept Proxy Only : YES or NO
  • Audio Codec :
    • G.711 A-law/μ-law, G.729A, G.723.1 (6.3K, 5.3K)
    • Select voice codec priority : Local or Remote
    • Voice Payload size (ms) configuration
    • Silence Suppression
    • VAD/CNG
    • LEC : Line Echo Canceller
    • Max Echo Tail Length (G.168): 32, 64 and 128ms
    • Packet Loss Compensation
    • Automatic Gain Control
    • In-band/out of band DTMF (RFC4733, RFC2833 / SIP INFO)
    • Adaptive/Configurable Jitter Buffer
    • G.168 Acoustic Echo Cancellation
    • Configure RTP basic Port
    • RTP QoS Type : DiffServ or TOS
    • Phone Book ( 50 records ) for peer to peer calls
    • Dialing Plan with drop, replace, Insert dialing digits
    • Select First digit and Inter digit timeout duration (Sec)
    • Selectable Call Progress Tone
    • Support Specified Line Calling
  • Call Features :
    • 4-Line FXO connect to PSTN or PBX simultaneously
    • Caller ID recognition DTMF (before/after 1st ring) and FSK (before 1st ring ),
      ETSI and Bellcore
    • DTMF Caller ID start and stop BIT configurable
    • Current Drop Detection to release FXO port
    • Disconnect tone recognition to release FXO port
    • Tone Generation: Ring Back, Dial, Busy, call waiting, ROH, Warning, Holding,
      Stutter dial tone and disconnect tone
    • Configure Tone Frequency, Cadence, Level and Cycle
    • Select Tone specification by Country name List
    • Global Country Based Tone Specification
    • NAT Traversal support STUN, UPNP and Behind NAT
    • Out-Band DTMF : RFC2833 and SIP Info
    • RFC2833 Payload type : 101 or 96
    • DTMF send out ON and OFF Time configure
    • DTMF incoming recognition Minimum ON and OFF time
    • DTMF Relay Volume configuration
    • T.38 FAX Volume configuration
    • Flash Time transmit via SIP Info (Enable or Disable)
    • Message Waiting Indication (Stutter Tone Notice)
    • Block Anonymous Call
    • Call Hold
    • Call Transfer
  • FXO Line Configuration:
    • Activate or deactivate
    • Line ID
    • FXO Line Phone number
    • Polarity Reversal detection for call establish and Billing
    • Current drop recognition to release port
    • Incoming call Handle: Hotline or 2 stage dialing
    • HOT Line to desired phone number
    • Play voice file to incoming call
    • Repeat playing voice file counts
    • Self-recorded voice files to upload
    • Generate FLASH TIME to PSTN network
    • T.38 or FAX Relay Type
    • Incoming and outgoing dB value configurable
    • Dialing Answer Delay time to establish call path
    • Answer PSTN incoming call after how many ring cycles
    • Caller ID detection mode by Country selection
    • VoIP dial to FXO/PSTN Line: 1 stage dialing and 2 stage dialing
    • Outgoing SIP Caller ID Selection
    • Support 4 SIP Trunk
    • Accept desired SIP Proxy incoming calls Only
  • Flexible Routing Plan :
    • Prefix Match and Length
    • Priority Ring
    • Cyclic Ring
    • Simultaneous Ring
    • Programmable Hunting Cycle
    • Backup Routes with Digit Manipulation
    • Default Routes
  • Flexible Dial Plans :
    • Retrieve transfer call from 3rd party by dial Code (default: *#)
    • Inter digit time out setting
    • First digit dial out delay time setting
    • End of dial keypad number
    • Dial Rule : Match dial Prefix and Maximum digits length ( 1-15 )
    • Phone Book can be Exported or Imported
  • Digit Manipulation (Drop and Replace Rule):
    • FXO DM Group
    • VoIP DM Group
    • DM 1 Group
    • DM 2 Group
    • DM 3 Group
    • DM 4 Group
    • Matched Prefix
    • Matched digit length
    • Replace digit start position
    • Replace digit stop position
    • Replace number
    • Incoming Ring frequency recognition range: 10 to 70 Hz
    • Incoming Ring ON time recognition range: 0 to 8000ms
    • Incoming Ring OFF time recognition range: 0 to 8000ms
    • Incoming Ring Level recognition range: 10 to 95Vrms
    • Support Peer to Peer Dialing
    • Flash Time Detection: range from 80 to 800 ms
    • Configure Ring Cadence, Frequency and Voltage
  • MANAGEMENT :
    • Administrative Telnet CLI and HTTP, HTTPS
    • HTTP provision through MAC address
    • Multilingual Web User Interface
    • 3 Levels of User Access Right with Password protection with different Web
      Language (Administrator, Supervisor and User)
    • HTTP/HTTPS Service Access limitation from WAN port
    • Configure Service ports at HTTP, HTTPS and telnet Services
    • Phone Debug Module: Device Control, Call Control, DB, Verbose
    • SIP Debug Module: Register, Call, SIP Message, Others
    • SNTP Debug Module
    • Device Debug Module
    • DSP Debug
    • Provide 8 Debug Levels :
      • Emergency
      • Alert
      • Critical
      • Error
      • Warning
      • Notice
      • Information
      • Debug
    • Provides System Status Logs
    • Connect to external SYSLOG Server
    • Status display: Network, Line, SIP Trunk status
    • Diagnostics (debug through Syslog Event Notice)
    • Debug in real time by Telnet
    • Auto Provision via HTTP Server
    • SNMP V2/Trap
    • Configuration Backup/Restore
    • Dual Firmware Image Backup
    • Reset to factory Default

    ** Support Welltech proprietary encryption protocol at SIP Signal and Voice codec
    during transmitting to IP network in order to Anti-ISP block of VoIP call. This
    feature only be available with Welltech SIP server or SIPPBX6200 IP-PBX

  • Environmental :
    • Actual Dimension: 17.5(W) × 3.2(H) × 12.6(D) CM
    • Weight: 0.5kg (One unit with packing)
    • Operating Temp. & Humidity
      • Temp.: 0°C~45°C (32°F~113°F)
      • Humidity: 10%~90% relative humidity, non-condensing
    • Power Adaptor:
      • INPUT: AC100V~240V, 50/60Hz
      • OUTPUT: DC 12V, 1.5A
  • Approvals:
    • CE, FCC (Part 15, Class B), LVD and RoHS
  • Country of origin:
    • Made in Taiwan
  • Packing Accessories
    • WellGate 2540 x 1 pcs
    • AC to DC+12V Power adaptor x 1 pcs
    • CD User Manual x 1 pcs
  • Warranty
    • One year
Datasheet Download Click here