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WellGate 2522
2-Line FXO, 2-Line FXS SIP IP Gateway
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- Dual IP Stack : IPv6 and IPv4 Simultaneously
- Support up to 4 SIP Trunk Servers
- Enhance existing PABX to VoIP Call
- Auto HTTP Provision feature
- Flexible Routes Plan, Dial Plan, Digit
Manipulation
- Redundant Firmware Image
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Introduction |
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WellGate 2522 is an 2-Line FXO plus 2-Line FXS gateway with
SIP protocol IP device which allows to connect 2 Lines of
analog PSTN telephone line and connect to 2 Lines analog telephone set (for instance, analog trunk line of PABX) to
make
or receive VoIP call over Internet or VPN network. This
device
is suitable for office PABX to enable VoIP call without
changing cabling, dial plan, extension number. |
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To select up to 4 SIP TRUNK Accounts |
WellGate 2522 is appropriate to use four VoIP SIP Trunk or IP Centrex service or IP-PBX within
offices and remote branch offices. One of four SIP Servers ( or ITSP Service provider or
alternative IP-PBX ) can be configured freely at each line ( FXO port or FXS port ) to make or
receive IP Call. It provides 4 service platforms according to your dial number or routes plan. |
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IPv6 VoIP Gateway is ready to Market |
IPv6 address was developed for years, however, it was not practical to our life up to date. More
and more electronic devices are able to link to IP Network, this makes existing IPv4 address
supply in short to global market. Meanwhile, the emerging countries are not able to increase
IPv4 address supply due to strong market demand on broadband services. WellGate 2522 is an
SIP based FXO plus FXS gateway which was built-in both IPv6 and IPv4 IP address. No matter
when you are ready to deploy IPv6 network now, or reserve the future expansion to IPv6 from
existing IPv4 address, WellGate 2522 is ready to grow up with you. Both IPv6 and IPv4 address
are working simultaneously at Voice IP Call. Its flexibility of both IPv6 and IPv4 accept and
interwork both addresses on today and tomorrow whenever you need. |
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Flexible Dial plan and Route Plan Features |
WellGate 2522 provides flexible Dial Plan between FXO, FXS and IP Trunk (SIP Soft
Switch). Dial
Plan is to configure in what condition the digits can be sent out to/from IP network. The dialing
inter-digit time before dialing is configurable to meet local PSTN line or PBX’s extension line. Dial
Rule is able to detect the prefix code and maximum digits reached and then dial automatically.
The Digit Manipulation (DM) allows you to configure
matched prefix code, digits length, start and
stop digit position to be replaced digits as
well.
Route Plan is to configure the incoming and outgoing call routes which you desired this
call to go
out or allow to income. For instance, IP incoming call may Reach to one FXO or
FXS port with
Priority or Cyclic access. You can also configure IP incoming call by Matched
prefix digits, Matched
dialing number to FXO/FXS line and Matched digit length. For
FXO/FXS outgoing call to IP
routes, the hunting type supports Priority or Cyclic or
Simultaneously to select which SIP trunk
( SIP Proxy Server ) to go. FXO/FXS outgoing
call routes also support by Matched prefix digits,
Matched outgoing SIP Trunk number
and Matched digit length. Both direction supports No
Answer time out and Backup
Routes. |
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Upgrade existing PABX to enable VoIP call |
WellGate 2522 is a SIP IP device to connect with existing PABX and enable to make or receive
VoIP call without affecting existing cabling, extension number or dial behaviors. It is an easy and
quick way to upgrade existing PABX to VoIP calls. Do not expand your PABX’s extension lines
and analog trunk lines. Bypass PSTN line to PABX automatically if this device power is failure or
IP network disconnection. It is an dependable solution to keep your PABX works as before.
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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 local PSTN line (RJ-11 x 4 pcs)
- DC +12V power input Jack
- Reset key to return Factory setting
- LED Indicator for System, SIP, FXS 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 :
- Support Peer to Peer Dialing
- 2-Line FXO connects to PSTN Line
- 2-Line FXS connects to analog phone set or PABX.
- 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/FXS Line Configuration:
- Activate or deactivate
- Line ID
- Line Phone number
- Polarity Reversal detection or generation for call establish and Billing.
- Current drop recognition or 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
- 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 4 SIP Trunk
- Accept desired 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
- FXS 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
- FXS Analog 2-wire interface :
- 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)
- 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 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 SIPPBX6200x IP-PBX
- Environmental :
- Actual Dimension: 17.5(W) × 3.2(H) × 12.6(D) CM
- Weight: 0.5kg (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, 1.5A
- Approvals:
- CE, FCC (Part 15, Class B), LVD and RoHS
- Country of origin:
- Packing Accessories
- WellGate 2522 x 1 pcs
- AC to DC+12V Power adaptor x 1 pcs
- CD User Manual x 1 pcs
- Warranty
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Comparison Table between WellGate 3702B and WellGate 2522 |
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WellGate 2522 |
WellGate 3702B |
| IPv6 |
YES |
NO |
| SIP |
YES |
YES |
| H.323 |
NO |
YES |
| NAT for WAN/LAN |
YES |
NO |
| Routes and Dial Plans |
YES |
NO |
| Rich Telephony Features |
YES |
NO |
| SIP Trunk |
4 sets |
NO (Only SIP Proxy Server) |
| Enclosure |
Plastic |
Metal |
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