WellGate 2424O/24FXO

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