8-PORT FXO SIP GATEWAY

8-PORT FXO SIP GATEWAY (COMPATIBLE WITH ASTERISK IP-PBX)

Dual IP Stack : IPv6 and IPv4 Simultaneously
Support up to 16 SIP Trunks to register
Enhance existing PABX to VoIP Call
Flexible Routes Plan, Dial Plan, Digit Manipulation
Redundant Firmware Image

8-port FXO gateway with SIP protocol IP device which connects up to 8 Lines of analog PSTN telephone line (or
extensions of PABX), to make or receive VoIP call over Internet or VPN network. This device is not only suitable for legend PABX to enable VoIP call without changing existing cabling, dial plan, and extension number, but also work with IP-PBX and SIP Server directly. Its design, firmware, auto provision and webpage are

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 PSTN line (RJ-11 x 8 pcs)
DC +12V power input Jack
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 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, GSMFR, G.723.1(6.3K, 5.3K)
Select voice codec priority: According to 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 :
Support Peer to Peer Dialing
Up to 8-port FXO connects to PSTN Line or extension 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 FXO analog devices
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
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
Play voice file to incoming call
Repeat playing voice file counts
Generate FLASH TIME to PSTN network
T.38 or FAX Relay Type
Incoming and outgoing volume in dB configurable
Dialing Answer Delay time to establish call path
Answer PSTN incoming call after how many ring cycles
Caller ID generation mode by Country selection
VoIP dial to FXO/PSTN Line: 1 stage dialing and 2 stage dialing
Outgoing SIP Caller ID Selection
Support 16 SIP Trunk
Accept desired SIP Proxy incoming calls Only (Avoid hacker’s call)
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
FXO Analog 2-wire interface :
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
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 Authority 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
Environmental :
Physical Dimension: 44(w) x 4.5(h) x 11(d) CM.
Weight: 2.3kg (One unit with packing)
Operating Temp. & Humidity
Temp.: 0°C~45°C (32°F~113°F)
Humidity: 10%~90% relative humidity, non-condensing
Maximum Power Consumption: 17 Watts (1.5A at 12Vdc)
Power Adaptor:
INPUT: AC100V~240V, 50/60Hz
OUTPUT: DC 12V, 3A
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]