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WellGate 2608

8-line FXS SIP IP Gateway

wellgate 2608 1WellGate 2608 is an 8-Line FXS gateway with SIP protocol IP device which allows to connect 8 sets of analog telephone to make or receive VoIP call over Internet or VPN network through Internet Telephony service provider. This device is suitable for office user through ITSP service provider to install at office or branch office to call between different offices. It also can install at trunk line in front of digital PBX to migrate digital PBX to IP Telephony call without changing PBX’s dial rule and cable

Rich Telephony Features

WellGate 2608 has 8-Line FXS ports. Each Line can be configured to meet different countries Telephony requirements. For Instance, AC Impedance, Ring Frequency, Ring Voltage, Distinctive Ring, Ring Cadence, different Tone (Level, Frequency, Cadence), Hook Flash detection time, DTMF/FSK Caller ID generation, Polarity Reversal Billing signal and Generate Current Drop Time (Open Loop Disconnect). FXS interface has highly compatibility with different countries Telephony interface, but also support existing digital PBX analog interface. You don’t need to worry the interoperability with existing PBX or Telco equipment.

Rich Digit Manipulation Features

Flexible digit manipulation processes Matched Prefix code, Minimum digit length, replace digits at start and stop digit position. This digit manipulation feature applies to FXS phone dial out or IP incoming call or pre-program up to 4 groups of digit manipulation processors. Each group can define up to 50 indexes to meet your variety dial purpose.

Flexible Dial plan

WellGate 2608 provides flexible Dial Plan from FXS to IP Trunk (SIP Softswitch). Dial Plan is to configure in what condition the digits can be sent out to IP network during dialing. The dial inter digit time before dialing is configurable to meet busy users or home user. Dial Rule is able to detect the prefix code and maximum digits reached and then dial automatically.

Call Routes Plan

Route Plan is to configure receiver IP address at peer to peer ( P2P ) mode when dialing from FXS phone. When dialing prefix code and Minimum length was matched at this plan, the IP address was sent to receiver site directly. Besides, there are four groups of digit manipulation can be applied at Call Routes plan before sending as well. Each Routes plan has an Backup Routes P2P IP address in case of original one was failure. Up to 50 entries of Call Routes List are supported to meet your application.

Suit to IP Telephony Service Provider

WellGate 2608 is a SIP IP device to connect with existing analog telephone set to make IP call. Its Simple, Slim and Compact design and easy installation allow office users or SOHO users to make or receive call just like an legend telephone call but less cost. It is compatible with broadband internet service devices such as ADSL/Cable Modem and WiMax/3G Modem. It helps ISP provider to provide Telephony service to existing customers without additional cable.

wellgate 2504a 3

Additional Info

  • PoE on WAN port
  • Dual IP Stack : IPv6 and IPv4
  • Simultaneously
  • Support up to 4 SIP proxy
  • Servers
  • Support different SIP Trunk to
  • each FXS line
  • Auto HTTP Provision feature
  • Flexible Routes Plan and Dial Plan
  • 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
Last modified on Saturday, 23 November 2013 13:53

Contacts

Contact Info

Netregy Systems Sdn Bhd.
26 Pusat Perdagangan One Puchong,
Jalan OP 1/3 Off Jalan Puchong,
47160, Puchong, Selangor,
Malaysia

Telephone:
+603 80707770

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