3232s 32-port FXS SIP IP Gateway

WellGate 3232s is a 32-port FXS gateway with SIP protocol IP device which connects 32 Lines of analog telephone sets or connect to analog CO line of legend PBX to make or receive VoIP call over Internet or VPN network. This device is suitable for office IP-PBX application between office to office or office to branch office to call via PSTN Line and IP network.

  • Dual IP Stack : IPv6 and IPv4 Simultaneously
  • Support up to 32 FXS extensions in one IP address
  • Support up to 16 SIP Trunk Servers
  • Both RJ-21 Female Telecom connector and RJ-45 4-FXS Socket
  • Flexible Routes Plan, Dial Plan, Digit Manipulation
  • Consistent with WellGate 2504, 2808 and 2424S gateway
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Product Description

To select up to 16 SIP service Accounts

WellGate 3232S is appropriate to use to 16 VoIP Service Providers, 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 assigned to WellGate 3232s to make or receive IP Call. It provides up to 16 SIP Trunk numbers for this gateway to use.

Flexible Dial plan and Route Plan Features

WellGate 3232s provides flexible Dial Plan between FXS and SIP IP Trunk (SIP Soft Switch or IP-PBX). 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 want this call to go out or allow to income. For instance, IP incoming call may reach to one FXS port with Priority or Cyclic access. You can also configure IP incoming call byMatched prefix digits, matched dialing number to FXS line and Matched digit length. From FXS outgoing call to SIP IP routes, the hunting type supports Priority or Cyclic or simultaneously select which SIP trunk ( SIP Proxy Server, or IP-PBX ) to go. FXS outgoing call routes also support by Matched prefix digits, Matched outgoing SIP Trunk number and Matched digit length. Both directions support No Answer time out and Backup Routes.

Suit to IP-PBX IP calls

WellGate 3232s is a SIP IP device to connect with analog telephone set and register to IP-PBX (or SIP Server IP Telephony platform) to make or receive IP call without affecting existing cabling, extension number or dial behaviors. It is an easy and quick way to upgrade from existing PBX to VoIP calls by using existing cable layout and dial number plan.

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
    • Telephony port (RJ-11 x 24 pcs)
      • RJ45 x 8 pcs, Each RJ45 socket contain 4 pairs of FXS.
      • RJ-21 Telecom connector (Female) x 2 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 FXS 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
    • Static IP at LAN port for computer to access
    • PPPoE Client
    • DDNS ( DynDNS )
    • DNS Client
    • Firewall
    • 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 16 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 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 :
    • Support Peer to Peer Dialing
    • Up to 8-port FXS connects to analog phone set or analog trunk of PBX.
    • Caller ID generates DTMF (before/after 1st ring) and FSK (before 1st ring ),
      ETSI and Bellcore
    • DTMF Caller ID start and stop BIT configurable
    • Current Drop generation to FXS analog devices
    • Disconnect tone generation
    • 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
    • Outgoing SIP Caller ID Selection
    • Support 16 SIP Trunk
    • Accept desired SIP Proxy incoming calls Only (Avoid hacker’s call)
  • FXS Line Configuration:
    • Activate or deactivate
    • Line ID
    • Line Phone number
    • Polarity Reversal generation for call establish and Billing.
    • Current drop generation to release line
    • Incoming call Handle: Hotline or 2 stage dialing
    • HOT Line to desired phone number
    • Recognize FLASH TIME from analog device
    • T.38 or FAX Relay Type
    • Incoming and outgoing volume in dB configurable
    • Dialing Answer Delay time to establish call path
    • Caller ID generation mode by Country selection
  • 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):
    • FXS 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
  • FXS Analog 2-wire interface:
    • Flash Time Detection: range from 80 to 800 ms
    • ON-HOOK Voltage -48Vdc
    • Support Polarity reversal to start Billing
    • Service Up to 1 Kilo-meter distance to analog telephone set
    • Generate Current Drop Time (Open Loop Disconnect time)
  • 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
  • Environmental :
    • Environmental :
    • Actual Dimension: 48(W)×4.4(H)×40(D) CM
    • Packing Dimension: 60 x 16 x 58 CM
    • Packing Weight: 8.3kg (One unit)
    • 19-inch, 1U chassis with Relay Rack Mount Bracket
    • 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
    • Power Consumption: 195Watts.