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Specifications

Developed by Realtone Technologies,designed for multi-purpose applications,supporting SIP and MGCP,high quality,best service.


VoIP SIP Gateway with 24, 48,72,96 FXS/FXO ports
Compatible with Asterisk and FreeSwitch
FXO PSTN gateway for call termination
 
VoIP SIP Gateway with 48 ,72 to 96 FXS/FXO ports, nice VoIP SIP access gateway for IP-PBX, IPcentrex, Call Center, VoIP Service provider;  48FXO PSTN ports for calling card, call termination;
 
WSS120 series VoIP SIP Gateway are a telecom carrier class VoIP Gateway designed for regular phone service, Fax service and IP Centrex service,WSS120 can be used by carrier and enterprise as part of the cost effective IP-PBX solutions;
 
 
Main application:
     1).  The ideal VoIP SIP access gateway for IP PBX like Asterisk, 3CX, Trixbox, Elastix, FreeSwitch for call center, hotel, Office IP-PBX system;
     2).  IPCentrex or residential applictions with IP PBX or Softswitch
     3). 16, 32,48 FXO ports gateway work with PSTN lines or GSM/CDMA terminal for calling card, and call termination
     4). Intercom PBX functions, IVR, Call transfer, 3-way-conference no need SIP Server;
 
 
WSS120 Series SIP VoIP Gateway including below models:
Model Name
FXS Ports
FXO Ports
WSS120-96S 
96
0
WSS120-72S
72
0
WSS120-48S
48
0
WSS120-96O
0
96
WSS120-72O
0
72
WSS120-48O
0
48
WSS120-88S/8
88
8
WSS120-64S/8
64
8
WSS120-40S/8
40
8
WSS120-80S/16
80
16
WSS120-56S/16
56
16
WSS120-32S/16
32
16
WSS120-72S/24
72
24
WSS120-48S/24
48
24
WSS120-24S/24
24
24
WSS120-36S/36
36
36
WSS120-48S/48 
48
48
 
 
Key Features 
  • MGCP or SIP access gateways 
  • SIP-PSTN analog gateway 
  • Routing (based on called number or other criteria) 
  • Radius based CDR for accounting 
  • STUN client for NAT transversal 
  • Flexible interface for analog phones, fax, traditional PBX with analog trunks. 
  • Redundant power supply option 
  • Hot swappable modules 
  • Fax tone detection and codec bypass (T.30) and fax relay (T.38) function 
  • G.711, G.723.1, G.729A, iLBC, GSM codec 
  • G.168 echo cancellation 
  • DTMF transportation: transparent mode, RFC 2833 mode, SIP/INFO mode 
  • Fashion Color ringing, MWI by Tone or Neon light, Routing table, Speed dial
Supported Protocols
  • Session Initiated Protocol (SIP) 
  • Media Gateway Control Protocol (MGCP) 
  • Real Time Transport Protocol(RTP/RTCP) 
  • TFTP for remote software upgrade 
  • HTTP for system and interface configuration and monitoring 
  • SNMP v.1, v.2, v.3 
  • Telnet for remote support 
  • DHCP for dynamic IP address allocation 
  • DTMF 
  • STUN Client for NAT/firewall traverse
Call Features
  • Caller ID 
  • Message waiting indication 
  • Call waiting 
  • Call forwarding 
  • Caller id on call waiting 
  • Call transfer (on busy, no answer, all) 
  • Special purpose calling codes 
  • Speed dial 
  • Redials 
Other Specifications 
  • Power Consumption : 70 Watt(Max) 
  • Internal Memory: 128MB 
  • System Flash Memory: 16MB 
  • Ethernet Interface: RJ-45 
  • Talk Battery: -24 V 
  • Ringing Voltage: 50V RMS 
  • Loop Current: 21 mA 
  • Max Line Length: 1500m or 415 ohm 
  • Power supply: 110-250V AC 50-60Hz
  • REN Equivalence: 5 for short loop (1000 feet) ,3 for long loop (5000 feet) 
  • Analog Interface: RJ-45
  • Off-hook Detection: Loop start 
  • Dialing: DTMF 
  • Size(L x W x H): 4.4 x 44 x 30 cm
  • Weight: 11Kg