- Interface:
- Ethernet port (RJ-45, 10/100 base-T)
- 1-WAN port, connect to IP Network
- 4-LAN port connect to PC with NAT
- Support Bridge and NAT mode
- Telephony port (RJ-11 x 8 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)
- Configurable WAN HTTP port, 80, 1024 to 65535
- Configurable WAN HTTPS port, 443, 1024 to 65535
- IP/ICMP/ARP/RARP/SNTP
- Static IP
- DHCP Client (RFC 2131), WAN port
- DHCP Server, LAN port
- Specify maximum DHCP Lease Time
- NAT Server (RFC 1631)
- PPPoE Client
- DNS Client
- Auto or Manual configure DNS Server IP address
- Behind NAT, use DMZ for NAT traversal
- Use STUN for NAT Type 1 and 2 for NAT Traversal
- SNTP with time zone setting
- TCP/UDP (RFC 793/768)
- RTP/RTCP (RFC 1889/1890)
- IPV4 ICMP (RFC 792),
- TFTP Client
- QoS : DiffServ (DSCP RFC 2475), ToS (RFC791, 1394)
- Configure DSCP on RTP and SIP signal separately
- Configure ToS on RTP and SIP signal separately
- SIP Protocol :
- RFC3261 compliance
- Support local SIP and RTP port configure from 1 to 65535 at each line and SIP
trunk. - SIP UDP Protocol
- Support RFC 3325 to send “ anonymous “ or not at caller ID
- SIP Session keep mode : Disable, Empty packet, SIP options, SIP register and
SIP Ping ( Nortel ). - 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)
- Generate Ring Back tone or Custom Tone after received SIP message 100
trying - 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 :
- G.711 A-law/μ-law, G.729, G.723.1 (6.3K, 5.3K),G.726-32
- Display IP Bandwidth with selected codec and payload
- Select Voice Codec Order : Local or Remote
- Silence Suppression
- VAD/CNG selection
- LEC : Line Echo Canceller
- Packet Loss Compensation
- Configure INPUT, OUTPUT and DTMF Gain
- In-band/out of band DTMF (RFC4733, RFC2833 / SIP INFO)
- Adaptive/Configurable Jitter Buffer range: 0 to 200ms
- G.168 Acoustic Echo Cancellation
- 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 :
- Caller ID display DTMF (before 1st ring) and FSK (before 1st ring ), ETSI and
Bellcore - DTMF Caller ID start and stop BIT (A,B,C,D,#) configurable
- Polarity Reversal before Caller ID or not
- Tone Generation: Ring Back, Dial, Busy, Call waiting, ROH and Disconnect
tone - Configure Tone Frequency, Cadence, Level and Cycle
- CDR output Server IP address and port number
- SYSLOG output Server IP address and port number
- NAT Traversal support STUN and IP Sharing
- Payload type setting : RFC2833, FAX Bypass and Modem Bypass
- Out-Band DTMF : RFC2833 and SIP Info
- DTMF detection Sensitivity setting : 1 (Lowest signal level) to 5
- DTMF generate configurable Duration and Interval Time
- Ring time limitation : 10 to 600 seconds
- Remote user drop call indication : Polarity Reversal or Loop Current Drop
- Network Connection Detection
- Network Unavailable announce Programmable Tone or Voice
- Message Waiting Indication
- Before dial first digit wait timeout configuration ( 1 to 60 sec )
- Inter Digits Timeout configure : 1 to 5 seconds
- Speed Dialing ( 50 sets )
- Call Waiting/Switching between Calls
- Call Forward (Busy, Unconditional, No Answer)
- No answer forward time setting
- Line service enable or disable
- Sequential Ring or Simultaneously Ring to each line
- Each line Ring Time setting at Sequential Ring
- Display each Line registration status
- Configure each line Ring priority at sequential ring
- Block Anonymous Call
- DND ( Do Not Disturb )
- Call Hold
- Configurable Call HOLD Tone or Music
- Call Transfer
- Flexible Dial Plan
- Dial Plan: Dial out immediately when matched leading digit and total digits
count ( 50 entries ) - Digit Manipulation (Drop and Replace Rule) :
- Apply Rule to FXS dialing out, IP incoming to FXS or pre-program 4 different
groups - Insert pause key (p) by 2 sec at DM group only
- Matched Prefix Code
- Matched minimum digit length
- Replace start and stop digit position
- Replace number
- Call Routes :
- Support Peer to Peer (P2P) Call only
- Matched Called Prefix code
- Matched Minimum Called digit length
- Secondary Backup Routes
- Support additional Digit Manipulation Group rule
- Hot Line
- Outgoing SIP Caller ID Selection
- Flexible Routing Plan
- Prefix Match and Length
- Priority Ring
- Cyclic Ring
- Simultaneous Ring
- Programmable Hunting Cycle
- Backup Routing with Digit Manipulation
- Default Routing
- FAX Transmission mode : T.38, Bypass or Auto
- FAX Bypass Keyword required from SIP Server
- FAX Bypass Codec : G.711 u-Law or G.711A-Law
- Support Peer to Peer Dialing
- Flash Time Detection: range from 60 to 2000 ms
- Flash Key definition: Disable, Transfer or SIP Message
- ON-HOOK Voltage -48Vdc
- Configure Ring Cadence, Frequency and Voltage
- Distinctive Ring Pattern by incoming Caller ID digit length
- Provide world wide country telephone line Impedance
- Ring Frequency range setup : 15 to 100 HZ
- Ring ON duration : 100 to 8000 ms
- Ring OFF duration : 100 to 8000 ms
- Ring AC RMS Level Voltage : 0 to 94 Vrms
- Support Polarity reversal for Billing
- Service Up to 1 Kilo-meter distance to analog telephone set
- Generate Current Drop Time (Open Loop Disconnect time)
- Caller ID display DTMF (before 1st ring) and FSK (before 1st ring ), ETSI and
- MANAGEMENT :
- Administrative Telnet, HTTP, HTTPS
- WAN IP address voice announcement from analog phone by dialing #126#
- LAN IP address voice announcement from analog phone by dialing #120#
- HTTP/FTP provision through MAC address
- 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
- Configuration Backup/Restore
- Upload user recorded voice announced file: Greeting, Hold Music, and Network
Failure - 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: 24(W) × 3.4(H) × 16(D) CM
- Weight: 1.5kg (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:
- INPUT: AC100V~240V, 50/60Hz
- OUTPUT: DC 12V, 3.0A
- Approvals:
- CE, FCC (Part 15, Class B), LVD and RoHS
- Country of origin:
- Made in Taiwan
- Packing Accessories
- WellGate 2608 x 1 pcs
- AC to DC+12V Power adaptor x 1 pcs
- CD User Manual x 1 pcs
- Warranty
- One year
WellGate 2608
Call for Price
Interface:
- Ethernet port (RJ-45, 10/100 base-T)
- 1-WAN port, connect to IP Network
- 4-LAN port connect to PC with NAT
- Support Bridge and NAT mode
- Telephony port (RJ-11 x 8 pcs)
- DC +12V power input Jack
- Reset key to return Factory setting
- LED Indicator for System, SIP and FXS status
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