Corecasys ATA is designed specifically to solve NAT problems in VOIP systems — a major challenge in VOIP deployments. Voice traffic is not like data traffic: the more layers of NAT in the network, the greater the impact on voice quality. In these situations, the VPN function built into the ATA is essential. Corecasys ATA supports up to 5 SIP accounts and router mode, with 1× WAN RJ45, 1× LAN RJ45 and 1× RJ11 telephone port. Status LEDs show WAN, LAN, SIP Server and phone activity. Audio is loud and clear, and the unit can run 24×7 without issues.

Unlike many other brands, Corecasys sells products together with after-sales service and the knowledge you need to use our equipment. As a systems integrator (SI), we help you use the product until your requirements are fully met. Our three-week trial policy also proves that after you buy, the equipment must work reliably for you — not create problems and frustration.

• Interface:
  • Ethernet port (RJ-45, 10/100 base-T)
    • 1-WAN port, connect to IP Network
    • 1-LAN port connect to PC with NAT
  • DC +12V power input Jack
  • Reset key to return Factory setting
• IP Network connection
  • IPv4 (RFC 791),
  • MAC Address (IEEE 802.3)
  • MAC Clone Setting
  • IP/ICMP/ARP/RARP/SNTP
  • Static IP
  • DHCP Client (RFC 2131), WAN port
  • DHCP Server, LAN port
  • Wire line speed more than 85MB at Bridge mode
  • PPPoE
  • DDNS
  • DMZ
  • VLAN : 802.1Q/1P
  • Virtual Server (DHCP Server IP range)
  • DNS Client
  • PPTP VPN Client tunnel 64 bits without compression
  • SNTP support Daylight Saving Time (DST) configuration
  • SNTP with time zone
  • TCP/UDP (RFC 793/768)
  • RTP/RTCP (RFC 1889/1890)
  • IPV4 ICMP (RFC 792),
  • TFTP Client
  • QoS Support : ToS
• SIP Protocol :
  • RFC3261 compliance
  • Support up-to 5 SIP Register Accounts
  • SIP Proxy compatible with brand name : Asterisk and Nortel
  • SIP UDP Protocol
  • Support SIP compact Form
  • SIP Session Timer (RFC 4028)
  • MD5 Digest Authentication (RFC2069/RFC2617)
  • Message Waiting Indication (RFC3842)
  • Event Notification (RFC3265)
  • REFER (RFC3515)
  • Support Outbound Proxy
  • Support DNS SRV to locate SIP Server (RFC 3263)
  • Support STUN NAT Traversal
  • Support "rport" parameter (RFC 3581)
• Audio Codec :
  • G.711 A-law/μ-law, G.729, iLBC, G.726
  • Silence Suppression
  • VAD/CNG
  • Jitter Buffer : Up to 32 packets
  • LEC : Line Echo Canceller
  • Packet Loss Compensation
  • Automatic Gain Control
  • In-band/out of band DTMF (RFC4733, RFC2833 / SIP INFO)
  • Adaptive/Configurable Jitter Buffer
  • Acoustic Echo Cancellation
  • Speed Dial
  • Phone Book ( up to 140 records )
  • Clock, Call-Duration display
  • Call History of Missed, Received and Dialed
  • Dialing Plan with drop, replace, Insert dialing digits
  • Selectable Call Progress Tone
  • Support Personal Melody Ring
  • Auto Answer Mode
  • Support Specified Line Calling
• Call Features :
  • Caller ID display DTMF (before/after 1st ring) and FSK (before 1st ring )
  • Tone Generation: Ring, Ring Back, Dial, Busy, call waiting and congestion tone
  • Out-Band DTMF : RFC2833 and SIP Info
  • Voice Mail with Indication
  • Speed Dialing
  • Call Waiting/Switching between Calls
  • Call Forward (Busy, Unconditional, No Answer)
  • DND : Always ON or configurable period
  • Call Hold
  • Call Mute
  • Call Transfer
  • Flexible Dial Plan: Drop and Replace Rule
  • T.38 FAX : Enable or G.711 Codec A-law/u-Law Pass trough Codec
  • Alarm Ring Reminder
  • 3-way conference call
  • Music-on-hold support (via IPPBX or local)
  • Redial
  • Hot Line
  • Support Peer to Peer Dialing
  • Volume Adjustment: Handset Volume (receiver) and Handset Gain (Transmitter) 
    selection
  • Flash Time Detection: range from 70 to 2550 ms
  • ON-HOOK Voltage -48Vdc
  • Support 12/16Khz metering signal or Polarity reversal for Billing
  • Service Up to 1 Kilo-meter distance from ATA to analog telephone set
  • Global Country Impedance setting
  • CPC Delay : 2 to 5 seconds (Open Loop Disconnect time)
  • CPC duration: 10 to 1200ms  
• MANAGEMENT :
  • Administrative Telnet CLI and HTTP
  • 2 Levels of User Access Right with Password protection
  • Management from WAN enable or disable
  • Provides System Status Logs
  • Network Status Display : WAN and LAN port Status
  • Diagnostics (debug through syslog)
  • Configuration Backup/Restore
  • Firmware configurable updated
  • Reset to factory Default
  • Support Auto Provision through MAC address
  • Voice configuration from analog telephone set with DTMF tone and voice 
    announcement
  • ** 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 :
  • Dimension: 9.9(H) × 9.9(W) × 3.2(T) CM
  • Weight: 0.35kg (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,
• Approvals:
  • CE, FCC, LVD and RoHS
• Country of origin:
  • Made in Taiwan
• Packing Accessories
  • ATA Box
  • AC to DC+12V Power adaptor x 1 pcs
  • 1 meter Ethernet cable x 1 pcs
  • CD User Manual x 1 pcs
• Warranty
  • One year

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