Specification

• Interface:
  • Ethernet port (RJ-45, 10/100 base-T)
  • 1-WAN port, connect to IP Network
  • 4-LAN port connect to PC with NAT
  • Support Bridge and NAT mode
  • Telephony port (RJ-11 x 8 pcs)
  • DC +12V power input Jack
  • Reset key to return Factory setting
  • LED Indicator for System, SIP and FXS status
• IP Network connection
  • IPv4 (RFC 791)
  • Configurable WAN HTTP port, 80, 1024 to 65535
  • Configurable WAN HTTPS port, 443, 1024 to 65535
  • IP/ICMP/ARP/RARP/SNTP
  • Static IP
  • DHCP Client (RFC 2131), WAN port
  • DHCP Server, LAN port
  • Specify maximum DHCP Lease Time
  • NAT Server (RFC 1631)
  • PPPoE Client
  • DNS Client
  • Auto or Manual configure DNS Server IP address
  • Behind NAT, use DMZ for NAT traversal
  • Use STUN for NAT Type 1 and 2 for NAT Traversal
  • SNTP with time zone setting
  • TCP/UDP (RFC 793/768)
  • RTP/RTCP (RFC 1889/1890)
  • IPV4 ICMP (RFC 792),
  • TFTP Client
  • QoS : DiffServ (DSCP RFC 2475), ToS (RFC791, 1394)
  • Configure DSCP on RTP and SIP signal separately
  • Configure ToS on RTP and SIP signal separately
• SIP Protocol :
  • RFC3261 compliance
  • Support local SIP and RTP port configure from 1 to 65535 at each line and SIP 
    trunk.
  • SIP UDP Protocol
  • Support RFC 3325 to send “ anonymous “ or not at caller ID
  • SIP Session keep mode : Disable, Empty packet, SIP options, SIP register and
    SIP Ping ( Nortel ).
  • Support SIP HOLD Type
  • SIP Session Timer (RFC 4028)
  • MD5 Digest Authentication (RFC2069/RFC2617)
  • Reliability of provisional response PRACK (RFC3262)
  • Early/Delay Media support
  • Offer/Answer (RFC3265)
  • Message Waiting Indication (RFC3842)
  • Generate Ring Back tone or Custom Tone after received SIP message 100 
    trying
  • Event Notification (RFC3265)
  • REFER (RFC3515)
  • Support Outbound Proxy
  • Support Primary and Backup SIP Server
  • Support STUN NAT Traversal
  • Support “rport” parameter (RFC 3581)
• Audio Codec :
  • G.711 A-law/μ-law, G.729, G.723.1 (6.3K, 5.3K),G.726-32
  • Display IP Bandwidth with selected codec and payload
  • Select Voice Codec Order : Local or Remote
  • Silence Suppression
  • VAD/CNG selection
  • LEC : Line Echo Canceller
  • Packet Loss Compensation
  • Configure INPUT, OUTPUT and DTMF Gain
  • In-band/out of band DTMF (RFC4733, RFC2833 / SIP INFO)
  • Adaptive/Configurable Jitter Buffer range: 0 to 200ms
  • G.168 Acoustic Echo Cancellation
  • 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 :
  • Caller ID display DTMF (before 1st ring) and FSK (before 1st ring ), ETSI and 
    Bellcore
  • DTMF Caller ID start and stop BIT (A,B,C,D,#) configurable
  • Polarity Reversal before Caller ID or not
  • Tone Generation: Ring Back, Dial, Busy, Call waiting, ROH and Disconnect 
    tone
  • Configure Tone Frequency, Cadence, Level and Cycle
  • CDR output Server IP address and port number
  • SYSLOG output Server IP address and port number
  • NAT Traversal support STUN and IP Sharing
  • Payload type setting : RFC2833, FAX Bypass and Modem Bypass
  • Out-Band DTMF : RFC2833 and SIP Info
  • DTMF detection Sensitivity setting : 1 (Lowest signal level) to 5
  • DTMF generate configurable Duration and Interval Time
  • Ring time limitation : 10 to 600 seconds
  • Remote user drop call indication : Polarity Reversal or Loop Current Drop
  • Network Connection Detection
  • Network Unavailable announce Programmable Tone or Voice
  • Message Waiting Indication
  • Before dial first digit wait timeout configuration ( 1 to 60 sec )
  • Inter Digits Timeout configure : 1 to 5 seconds
  • Speed Dialing ( 50 sets )
  • Call Waiting/Switching between Calls
  • Call Forward (Busy, Unconditional, No Answer)
  • No answer forward time setting
  • Line service enable or disable
  • Sequential Ring or Simultaneously Ring to each line
  • Each line Ring Time setting at Sequential Ring
  • Display each Line registration status
  • Configure each line Ring priority at sequential ring
  • Block Anonymous Call
  • DND ( Do Not Disturb )
  • Call Hold
  • Configurable Call HOLD Tone or Music
  • Call Transfer
  • Flexible Dial Plan
  • Dial Plan: Dial out immediately when matched leading digit and total digits 
    count ( 50 entries )
  • Digit Manipulation (Drop and Replace Rule) :
  • Apply Rule to FXS dialing out, IP incoming to FXS or pre-program 4 different
    groups
  • Insert pause key (p) by 2 sec at DM group only
  • Matched Prefix Code
  • Matched minimum digit length
  • Replace start and stop digit position
  • Replace number
  • Call Routes :
  • Support Peer to Peer (P2P) Call only
  • Matched Called Prefix code
  • Matched Minimum Called digit length
  • Secondary Backup Routes
  • Support additional Digit Manipulation Group rule
  • Hot Line
  • Outgoing SIP Caller ID Selection
  • Flexible Routing Plan
  • Prefix Match and Length
  • Priority Ring
  • Cyclic Ring
  • Simultaneous Ring
  • Programmable Hunting Cycle
  • Backup Routing with Digit Manipulation
  • Default Routing
  • FAX Transmission mode : T.38, Bypass or Auto
  • FAX Bypass Keyword required from SIP Server
  • FAX Bypass Codec : G.711 u-Law or G.711A-Law
  • Support Peer to Peer Dialing
  • Flash Time Detection: range from 60 to 2000 ms
  • Flash Key definition: Disable, Transfer or SIP Message
  • ON-HOOK Voltage -48Vdc
  • Configure Ring Cadence, Frequency and Voltage
  • Distinctive Ring Pattern by incoming Caller ID digit length
  • Provide world wide country telephone line Impedance
  • Ring Frequency range setup : 15 to 100 HZ
  • Ring ON duration : 100 to 8000 ms
  • Ring OFF duration : 100 to 8000 ms
  • Ring AC RMS Level Voltage : 0 to 94 Vrms
  • Support Polarity reversal for Billing
  • Service Up to 1 Kilo-meter distance to analog telephone set
  • Generate Current Drop Time (Open Loop Disconnect time)
• MANAGEMENT :
  • Administrative Telnet, HTTP, HTTPS
  • WAN IP address voice announcement from analog phone by dialing #126#
  • LAN IP address voice announcement from analog phone by dialing #120#
  • HTTP/FTP provision through MAC address
  • 3 Levels of User Access Right with Password protection (Administrator, 
    Supervisor and User)
  • HTTP/HTTPS Service Access limitation from WAN port
  • Provides System Status Logs
  • Status display: Network, Line, SIP Trunk status
  • Diagnostics (debug through Syslog Event Notice)
  • Debug in real time by Telnet
  • Configuration Backup/Restore
  • Upload user recorded voice announced file: Greeting, Hold Music, and Network 
    Failure
  • 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: 24(W) × 3.4(H) × 16(D) CM
  • Weight: 1.5kg (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, 3.0A
• Approvals:
  • CE, FCC (Part 15, Class B), LVD and RoHS
• Country of origin:
  • Made in Taiwan
• Packing Accessories
  • WellGate 2608 x 1 pcs
  • AC to DC+12V Power adaptor x 1 pcs
  • CD User Manual x 1 pcs
• Warranty
  • One year