4-PORT FXS SIP GATEWAY

4-PORT FXS SIP GATWEAY (Compatible for any Asterisk IP-PBX)

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
Redundant Firmware Image
4-line FXS gateway with SIP protocol IP device which allows to connect 4 sets of
analog telephone sets 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. Highly compatible with Asterisk IP-PBX (Elastix, freepbx, trixbox and other famous sip server software.)

SPECIFICATION

Interface:
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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 (RJ-11 x 4 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) 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
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
VLAN Support 802.1Q, 802.1P
VLAN ID Range : 2 to 4094
VLAN Priority : 0 to 7
QoS : DiffServ (RFC 2475), TOS (RFC791, 1394)

SIP Protocol :
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RFC3261 compliance
Support up-to 4 SIP Server Register Accounts
SIP UDP Protocol
Support SIP compact Form
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)
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:
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G.711 A-law/μ-law, G.729A, G.723.1 (6.3K, 5.3K)
Select voice codec priority : Local or Remote
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
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 :
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Caller ID display DTMF (before/after 1st ring) and FSK (before 1st
ring ), ETSI and Bellcore
DTMF Caller ID start and stop BIT configurable
Polarity Reversal before Caller ID or not
Tone Generation: Ring, Ring Back, Dial, Busy, call waiting, ROH,
Warning, Holding, Stutter dial tone and disconnect tone
Configure Tone Frequency, Cadence, Level and Cycle
Global Country Based Tone Specification
NAT Traversal support STUN, UPnP and Behind NAT
Out-Band DTMF : RFC2833 and SIP Info
Flash Time transmit via SIP Info (Enable or Disable)
Message Waiting Indication (Stutter Tone Notice)
Speed Dialing
Call Waiting/Switching between Calls
Call Forward (Busy, Unconditional, No Answer)
Block Anonymous Call
Call Hold
Call Transfer
Retrieve transfer call from 3rd party by dial Code (default: *#)
Flexible Dial Plan
Dial Rule : Prefix and Maximum digits, digit position to replace
Digit Manipulation (Drop and Replace Rule)
Extension Hunting
Hot Line
Support SIP Trunk up 4 SIP Servers
Outgoing SIP Caller ID Selection
Accept desired SIP Proxy incoming calls Only
Flexible Routing Plan
Prefix Match and Length
Priority Ring
Cyclic Ring
Simultaneous Ring
Programmable Hunting Cycle
Backup Routing with Digit Manipulation
Default Routing
T.38 FAX
Support Peer to Peer Dialing
Flash Time Detection: range from 80 to 800 ms
ON-HOOK Voltage -48Vdc
Configure Ring Cadence, Frequency and Voltage
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 CLI and HTTP, HTTPS
http provision through MAC address
Multilingual Web User Interface
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
Auto Provision via HTTP Server
Support SNMP V2/Trap
Configuration Backup/Restore
Dual Firmware Image Backup
Reset to factory Default

Environmental :
---------------------
Dimension: 17.5(W) × 3.2(H) × 12.6(D) CM
Weight: 0.9kg (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:
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- INPUT: AC100V~240V, 50/60Hz
- OUTPUT: DC 12V, 1.5A
- 15 Watts
Approvals:
CE, FCC (Part 15, Class B), LVD and RoHS

CONTACT US

NORWELL ENGINEERING
Telephone: (632) 292.76.58
Mobile ; (0908) 814.72.47
Email: [email protected]