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WG-2424O 24 Ports -Lines FXO SIP IP Gateway

WG-2424O 24 Ports -Lines FXO SIP IP Gateway
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Price: $850.00
Availability: In Stock
Model: WellGate 2424O
Manufacturer: WellTech
Average Rating: Not Rated

Quantity Discount:
Order Quantity Price Per Item
5 $830.00
10 $810.00

WG- 2424O  24-Line FXO SIP IP Gateway


   
  • Dual IP Stack : IPv6 and IPv4 Simultaneously
  • Support up to 16 SIP Trunk Servers
  • Support different SIP Trunk to each FXO line
  • Auto HTTP Provision feature
  • Flexible Routes Plan, Dial Plan, Digit Manipulation
  • Redundant Firmware Image
 

Introduction

WellGate 2424O is an 24-Line FXO gateway with SIP protocol IP device which connects 24
Lines of analog PSTN telephone network or connect to analog extension of PABX to make
or receive VoIP call over Internet or VPN network. This device is suitable for office IP-PBX
application at office to office or office to branch office to call between PSTN Line and IP
Call.

To select up to 16 SIP TRUNK Accounts

WellGate 2424O is appropriate to use 16 VoIP SIP Trunk or IP Centrex service or IP-PBX
within offices and remote branch offices. One of 16 SIP Servers ( or ITSP Service provider
or alternative IP-PBX ) can be configured freely at each line ( FXO port ) to make or
receive IP Call. It provides up to 16 service platforms according to your dial number or
routes plan.

Flexible Dial plan and Route Plan Features

WellGate 2424O provides flexible Dial Plan between FXO and IP Trunk (SIP Soft Switch).
Dial Plan is to configure in what condition the digits can be sent out to/from IP network.
The dialing inter-digit time before dialing is configurable to meet local PSTN line or PBX’s
extension line. Dial Rule is able to detect the prefix code and maximum digits reached
and then dial automatically. The Digit Manipulation (DM) allows you to configure matched
prefix code, digits length, start and stop digit position to be replaced digits as well.
Route Plan is to configure the incoming and outgoing call routes which you desired this
call to go out or allow to income. For instance, IP incoming call may Reach to one FXO
port with Priority or Cyclic access. You can also configure IP incoming call by Matched
prefix digits, Matched dialing number to FXO line and Matched digit length. For FXO
outgoing call to IP routes, the hunting type supports Priority or Cyclic or Simultaneously
to select which SIP trunk ( SIP Proxy Server ) to go. FXO outgoing call routes also
support by Matched prefix digits, Matched outgoing SIP Trunk number and Matched digit
length. Both direction supports No Answer time out and Backup Routes.

VoIP Point to Point Call

In order to connect two different buildings or remotely offices with existing PABX and
analog telephone set, The 24 Lines FXO and FXS gateway were configured to peer to
peer mode to extend voice call via IP network. The following diagram illustrates its
application.

Suit to IP-PBX to access local PSTN Line

WellGate 2424O is a SIP IP device to connect with IP-PBX to access local PSTN network
with FXO interface. Its telephony features, for instance, Caller ID detection and Releasing
FXO port after call was dropped, are easy to integrate with Legend Telephony Line with
IP-PBX in office and branch office IP call application. It is compatible with local Telecom
network regulation and your office IP network to transmit analog voice between them.

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 24 pcs)
    • RS-232 Console port, DB9 Male, 115200 bps
    • AC power input Jack
    • AC Power ON/OFF Switch
    • 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 :
    • 24-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 up to 16 SIP Trunk
    • Accept desire 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: 44(W)×4.4(H)×26.2(D) CM
    • 19-inch, 1U chassis with Relay Rack Mount Bracket
    • Weight: 4.3kg (One unit with packing)
    • Operating Temp. & Humidity
      • Temp.: 0°C~45°C (32°F~113°F)
      • Humidity: 10%~90% relative humidity, non-condensing
    • Power Input: AC100V to 240V, 50/60Hz
  • Country of origin:
    • Made in Taiwan
  • Packing Accessories
    • WellGate 2424O gateway x 1 pcs
    • Relay Rack Mount Bracket x 2 pcs
    • AC Power cable x 1 pcs
    • CD User Manual x 1 pcs
  • Warranty
    • One year

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