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