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					   VoIP Server Center Solution




                                               WELLTECH
WellBilling 6600          WellSIP 6500
Billing Server            Redundant SIP
                          Telephony Server




                          WellBG 5800
WellRTP 5100
                          Session Border
RTP Server
                          Controller




WellRec 5600              SIPIVR 6800
IP Recorder               Application Server




IPCentx 6850
IP Centrex                WellXfer 6300
                          Conference Server
Application




                          PRO 9510
WellGate 5250             Element
                          Provisional Server
Universal Trunk
Gateway



Wellsc 5900
Universal SIP/H.323
Converter
              WellSIP 6500                                     SIP Telephony Server
VOIP SERVER


              The Welltech WellSIP 6500 series SIP Telephony Server is the best choice to
              your convergence VOIP network which covert the requirements from enterprise
              to service provider. With built-in rich telephony services, WellSIP 6500 enables
              traditional PABX features to your VOIP convergence platform. Also you can
              easily upgrade the license or provide the high available service according to the
              growth of your business without any hardware changes.



              Intelligent Call Routing
              WellSIP 6500 provides multiple service routing polices to meet different service providers’ requirements
              (e.g. load balancing, priority, most idle etc.) It enables service provider to tell how to route the call depending
              on the call results or predefined rules. The incoming prefix match and outgoing prefix insert provides a very
              easy way to manage your VOIP exchange service.


              Easy to Configure and Management
              Full web management interface make you to manage your WellSIP 6500 anywhere of the world. You don’t
              need remember the command lines or operate it on the specified console. Also the system event notice features
              keep you the system status updated remotely.


              NAT On-Demand Traversal
              Due to the lack of IPV4 address, a lot of customer is using NAT for their network. WellSIP 6500 provides the
              NAT on-demand traversal which will only route the voice when needed. It saves the bandwidth and provides
              better voice quality compare to route each call voice back to server. No CPE modification is required.


              Voice NAT/Firewall Router
              With built-in SIP and voice routing features, WellSIP 6500 provides a secure and easy way to migrate your
              voice IP PBX solution. It acts as a NAT router and firewall role which voice RTP port is only opened when SIP
              signaling is established successfully.


              Rich Telephony Service
              The WellSIP 6500 provides build-in rich set of telephony service which enables the service provider quick time
              to market to delivery their service to their customers. By cooperating to Welltech IPCentrex 6850, the service
              provider can provide Announcement, Auto Attendant, VMS, CRBT etc. immediately.


              Multiple Access to Receive Calls Anywhere
              With provided SIP TCP and UDP protocol, WellSIP 6500 can accept both type of signals and do the conversion
              when needed. For each protocol, WellSIP 6500 can support up-to 3 service ports which enabling to receive
              SIP service anywhere of world. Also a proprietary voice and SIP encryption can break through all ISP
              blockings.


              High Availability Redundant
              WellSIP 6500 provides high availability VOIP service by using active and stand-by redundant technologic which
              provides hot standby and hitless fail-over for stable call to reach mission-critical service requirement. It keeps
              your service continues running.


              Microsoft Unified Communication Server Integration
              WellSIP 6500 can work with Microsoft Live Communication Server as a total solution to meet the enterprise
              communication requirement. Without any extra settings in Live Communication Server, WellSIP 6500 connecting
              your Office Communicator to PSTN and VOIP world. Also the WellSIP 6500 can becomes a perfect connecting
              in between Exchange 2007 and PSTN/VOIP.
                                                                                                                                                            VOIP SERVER
             7x24 Running
               Redundant




Selected Features                                      Application Examples
●   Call Transfer                                      ●   VOIP Core Service
●   Call Forward                                       ●   VOIP Class 5 Subscriber Service
●   Call Forwarded Notice                              ●   Internet Exchange Service Center
●   Call Screening                                     ●   Hosted IP-PBX
●   Caller ID Privacy
                                                       ●   Medium to Large Enterprise IP-PBX
●   Call Waiting
                                                       ●   VOIP Call Center
●   Call Hold
                                                       ●   Microsoft LCS 2005 Integration
●   Call Pickup (Global, Group)
                                                       ●   Exchange 2007 Integration
●   Specified Call Pickup
●   Find Me
●   Short Code
                                                       ●   Service Provider Application
●   Do Not Disturb
                                                                                                                    SIP Proxy Telephony Server
●   Miss Call Notify by Email                                                                                       Hitless HA Redundant

●   ANI Replacement                                                             E1/T1 Trunk Gateway
                                                                                ISDN/CAS
●   Call Return                                            IP Phone
                                                                                SS7
                                                                                                                        WS6500
                                                                                                                                         RTP Resource Server
●   Hide ANI/Show ANI Selection                                       PSTN                                             WS6500
●   Call Park/Retrieve
●   Call Camp on                                                                     WG5250              SIP VOIP Network
                                                                                                                                                    WelRTP 5100
●   Display Name Replacement
●   PSTN Number                                              Private Network
                                                                   User
●   Ring PSTN & IP Device Simultaneously                                                     SIPBG 5800                                         IP Phone




Ready-to-Run Value Added Service                           IP Phone
                                                                       IAD                 SIP Session Boarder Controller
                                                                                           Freeing NAT Users                IAD
●   System Announcement Service
●   Multi-Company Auto Attendant
●   Voice Mail Service
                                                              ●   Large Enterprise IP-PBX Application
●   Coloring Ring Back Tone Service
●   Number Change Notice                                              IP-PBX Core
                                                                                                                              AA/VMS/CRBT
●   Call Forward Notice
                                                                                            WS6500
●   Call Forward Notice and Forward
                                                                                             WS6500
●   Call Interception                                                                                                       IP Centrex 6850

●   Call Recording Service
●   IP Centrex                                                                       SIP VOIP Network
                                                                                                                                    WellXfer 6300

                                                                       WG5250



    WellRTP 5100
                                                                                                                                  Conference Bridge


    External RTP Resource                                                       IP Phone    IP Phone      IP Phone

    ●   Up-to 384 RTP channels extension for WellSIP 6500
    ●   Subscriber Selectable
    ●   Automatically load balancing by WellSIP 6500
WellSIP 6500B                                   SIP Telephony Server

The Welltech WellSIP 6500B, a combination version of SIP Proxy Telephony
Server and RADIUS Billing Server together, is a very cost effective solution to
start your convergence VOIP network business. You can start your service
immediately to provide prepaid, post paid and a lot of enhanced service, no
interoperability issues any more.

SIP Proxy Telephony Features
●   Intelligent Call Routing                      ●   Immediate Response
●   Easy to Configure and Management              ●   Share Secret with MD5 Protection
●   NAT On-Demand Traversal                       ●   Wellgate 5250 and WellSIP 6500B Full
●   Voice NAT/Firewall Router                         Interoperability
●   Multiple Access to Receive Calls Anywhere     ●   Automatic 6500 provision
●   High Availability Redundant                   ●   Detail Access Log
●   Microsoft Unified Communication Server        ●   Provide Basic Reports
    Integration                                   ●   Support Coin Phone Service
●   Support SIP & RTP Encryption/Decryption       ●   Support Calling ID (ANI) Validation
●   Selected Telephony Features                   ●   Support Charge Account
      Call Transfer                               ●   Auto Monthly Charge Deduction
      Call Forward                                ●   Up-to 5 level User Management
      Call Forwarded Notice                             Administrator
      Call Screening                                    Distributor
      Caller ID Privacy                                 Group Reseller
      Call Waiting                                      Reseller
      Call Hold                                         Subscriber
      Call Pickup (Global, Group)                 ●   Prepaid Service
      Specified Call Pickup                             Real Time Balance Deduct
      Find Me                                           Subscriber/Reseller Recharge & Rollback
      Short Code                                        Recharge Log
      Do Not Disturb                                    Effective Date/Expired Date
      Miss Call Notify by Email                         PIN Code Generate and Consume
      ANI Replacement                             ●   Postpaid Service
      Call Return                                       Call Detail Record Storage
      Hide ANI/Show ANI Selection                       Effective Date/Expired Date
      Call Park/Retrieve                                CDR Report
      Call Camp on                                ●   Flexible Rate Plan Support
      Display Name Replacement                          Up-to 5 Charge Segments per Rate Prefix
      PSTN Number                                       Effective Date/Expired Date
      Ring PSTN & IP Device Simultaneously              Longest Prefix Match
RADIUS Billing Server Features                          Programmable Charge Unit, Amount and Cycle
●   RADIUS AAA Support                                  Support Per Call Charge
      Authentication Message                            Call Screening
      Authorization Message                             Holiday & Night Time Charge
      Billing Start/Stop Message                        Free Monthly Minutes based on Prefix
●   RFC 2865, 2866 Compliance with Selected             Monthly Free Charge based on Prefix
    Attributes                                          Deductible Monthly Fee
●   Up-to 1000 subscribers support                ●   External Database support
●   Fully Web Management Interface                      MS-SQL
●   Support Prepaid/Postpaid User                       Built-in DB Connection Pool management
●   Max Call Duration Protection
WellBG 5800




                                                                                                                                              VOIP SERVER
Session Border Controller
WellBG 5800 is a SIP-aware Session Border
Controller which provides a cost-effective solution to
deploy VOIP services. It offers automatic NAT detection & traversal,
SIP signaling conversion, network topological protect and greatly decreased
the soft-switch loading.

Features
●   Free NAT Connections from SIP Proxy or Soft-Switch

●   Easy Plug and Play without Pre-setting Subscriber

●   Up-to 3 UDP service ports                                                       NAT/Firewall
●   Auto NAT Detect and Traversal

●   Support Audio and Video NAT Traversal
                                                                                                   Audio
●   Authentication Cryptography Bypass

●   Global NAT Group/IP Definition                                                                 Video
●   Black List/White List of ANI and IP

●   Call Protection for non-Registered User

●   Register Protection for Authentication-Failed User

●   Call Keep Live Validation

●   Provides Call Detail Record

●   Call Statistic Report and Subscriber

    Status Monitor

●   Real Time RADIUS Billing Message

●   RFC 3261 Comply
                                                                            Class 5 Soft-Switch
●   Support SIP Call Hold and Call Transfer

●   Support external RTP resource server
                                                                                   VOIP                   Enable NAT Users
                                                                                                          Auto Audio/Video proxy
●   Support VOIP Recorder                                                          Core                   Plug and Play
                                                                                   Network                Soft-Switch Protection
●   Support 2 Ethernet Legs for Voice

    Router/Firewall

●   Support Network Failure Backup                                   WellBG 5800                  WellBG 5800

●   Support N+1 Redundant                                                                                                          IP Phone
                                                     NATed
                                                     Private   NATed                                       Internet
                                                                                                                            Private
Application Example                                  Network   Private                       Private
                                                                                                                            Network
                                                               Network                       Network
●   Hosted PABX Customer Side SBC
                                                                                         Soft-phone              IP Phone
                                               IAD                                                                                 IP Phone
●   Hosted Call Center Customer                                IAD




    Side VOIP Recording

●   Soft-Switch Server Side SBC

●   Lawful Interception for Soft-Switch
              SIPIVR 6800/6800S
VOIP SERVER

              Service Creation Application Server

              SIPIVR 6800 brought you a fully web user interfaced Value Added Service
              Creation Application Server. By using easy drag and drop web interface,
              you can create your owned VOIP application or value added service very quick
              time to market. With built-in rich pre-designed components, the developer can
              create their service without paying attention to the complexities of programming.
              Also the real time debugger makes developer very easy to debug and trace
              the call flow status.


                                                                                                              SIPIVR 6800S

                                                                  SIPIVR 6800




                                                    SIPIVR 6800                                                     SIPIVR 6800S




               Selected Features
               ●   Up-to 120 Universal VOIP Channels
               ●   SIP RFC 3261 Compliance
               ●   Fully Web Management Interface
               ●   Audio Codec G.711, G.729A, G.723.1 *
               ●   Drag and Drop Call Flow Editor                                                                                  DB Query
                                                                                USA PSTN
               ●   Real Time Status/Variable Debugger
               ●   Rich-set of Predefined Components:
                                                                                                                                    SIPIVR 6800
                                                                                             WG5250   SIP VOIP Network
                      Basic Flow Components
                      IVR Components                                                                                                   VOIP Application
                                                                                        WG 38xx
                                                                                                                                       Server
                      Database Components
                      Flow Control Components                                   China PSTN
                                                                                                             WG5250
                      RADIUS Components
                      Channel Components
                                                                                                        TWN PSTN
                      HTTP Access Components
                      External Customized Components
               ●   Support Call Hold and Transfer
               ●   Support in-Band and out-of-Band DTMF relay
               ●   Support Database Connection Pools
               ●   Support Internal/External Job Push & Retrieve
               ●   Support Internal/External Hook Function Calls
               ●   Free Text Math Expression with rich functions
               ●   Hitless Call Flow Update
               ●   Optimized Developing Platform


               Application Examples
               ●   Audio Broadcasting/Announcement Service
               ●   Hosted IVR
               ●   Prepaid Calling Card Service
               ●   Universal Call Back System
               ●   IP Centrex Service
               ●   Voice Mail System
               ●   Outbound Dialer System
               ●   VOIP Call Center

               * G.723.1 is an optional for 6800S
IPCentx 6850




                                                                                                                                 VOIP SERVER
The first ready to run enhanced service
application, SIP IP Centrex, is based on powerful
Welltech SIPIVR 6800 to provide the customer a quick time
to market and easy to customize their needs’ solution. With built-in
pre-designed Auto Attendant, Voice Mail, Coloring Ring Back Tone and
Announcement service, IP Centrex fulfils the requirements of service provider.
It supports multi-company operation and each company can have their own
administrator to manage.

Features
●   Auto-attendant
      Multiple Time-based Greeting
      Multi-language Support
                                                                                              Auto Attendant
      Special Greeting                                                                        Voice Mail
                                                                                              Coloring Ring Back Tone
      Multi-company                                                                           Announcement
                                                                                                 -
                                                                                              Multi Company
      Customizable Greeting
●   Voice Mail System                                                                 WS 6500

      Customization Personal Greeting                                                 WS 6500              IPCentx 6850
                                                                           SIP VOIP Network
      Distinct Busy or no Answer Voice
      Mail Process
                                                                  Company A              Company B
      Multi-language support                     WG 3806         Private Network        Private Network       WG5250
                                                                                                                          PSTN
      IVR and Web Voice Mail Access
      Automatic Email Forward
                                                 PSTN
      Message Waiting Indication                           IP Phone   IP Phone     IP Phone    IP Phone IP Phone


●   Color Ring Back Tone
                                                           Company A                                  Company B
      Caller based & default CRBT
      Up-to 10 Callers Setting per Subscriber
●   Announcement
      Reason-Code Mapped Announcement
      Number Change Notice
●   Enhanced Service
      Subscriber Balance Inquiry
      Call Me Testing Line
      Call Back Based on ANI
      Calling Card Service
●   SIP RFC 3261 Compliance
●   G.711, G.723.1, G.729A Support
●   System Administrator/Company Administrator Management
●   Support subscriber login and access their voice mail and CRBT library
●   Support Customizable Call Flow by Easy Drag & Drop UI
●   Fully Web Management Interface
●   Stackable by using External NAS and DB
●   Up-to 120 Channels/10000 Subscriber per System
              WellRec 5600
VOIP SERVER
                                                                 VoIP Recorder
              WellRec 5600 is a dedicated VoIP recorder for Welltech SIP network
              architecture. Just simply setting the subscriber or the target list to be logged
              in WellSIP 6500, WellRec 5600 will do the recording on demand no matter
              incoming call or outgoing call. Through the web interface, administrators can
              easily search and play the recorded VOIP call anywhere. It provides a very
              cost/effective solution to meet your VOIP logging service.


              WellRec 5600 can also be a lawful interception or voice monitor system.
              Administrator can monitor the conversation real time either by recording channel
              or selected subscriber without any delay.

              Features
              ●   Full Integrated with WellSIP 6500                     ●     Encrypted or mal-formatted RTP voice stream
              ●   Subscriber Based Voice Recording                            can be kept for off-line process
              ●   Target Based Voice Recording                          ●     Recorded Call History for Subscriber, Supervisor
              ●   Multiple SIP proxy server support                           or Administrator
              ●   Fully Web Management Interface                        ●     Support external DB and NAS
              ●   Support G.723, G.729, G.711 codec                     ●     Stackable to extend
              ●   MP3 Compression for Storage and Play Back             ●     Support CTI information Attachment
              ●   Separate Caller and Called voice tracks               ●     Up-to 120 concurrent recording channels
              ●   Easy Search by Calling Number, Called number,         ●     Up-to 30,000 hours storage with RAID 5
                  Date or Time
              ●   Public Caller and Called Signal IP Address and        Application Examples
                  Port Tracking                                         ●     VOIP Call Center Recorder
              ●   Lawful Interception or Call Monitoring                ●     Lawful Interception
                     Channel based Interception                         ●     Enterprise Call Recording
                     Subscriber based Interception
                     Selectable both, Caller or Called voice
                     Supervisor based interception list
                     Up-to 32 Interception Target for each Supervisor
                     Support Behind NAT Call Monitoring
                                                                                           Target based Recorder
                     Support IETF HI2/HI3 ftp Features                                     Subscriber Based Recorder
                                                                                           Call Interception
                                                                                           No Extra Setting or Wiring
                                                                                           Caller/Called Number & IP
                                                                                           Easy to Retrieve

                                                                                        WS6500
                                                                                        WS6500



                                                                                                        WellRec 5600

                                     PSTN

                                                                        SIP VOIP Network
                                                            WG5250




                                                                                                               IP Phone

                                                                        IAD
WellBilling 6600




                                                                                                                    VOIP SERVER
WellBilling 6600 is a high performance,
reliable and scalable RADIUS billing server.
Its flexible rate plan and reseller/subscriber features
fulfill the requirements of service provider. With built-in prepaid
and postpaid subscriber service, service provider can provide their time to
market VOIP service. It is the best shoot of a cost/effective solution.

Features
●   RADIUS AAA Support                                          ●    Prepaid Service
      Authentication Message                                           Real Time Balance Deduct
      Authorization Message                                            Subscriber/Reseller Recharge & Rollback
      Billing Start/Stop Message                                       Recharge Log
●   RFC 2865, 2866 Compliance with Selected Attributes                 Effective Date/Expired Date
●   Up-to 500k subscribers support                                     PIN Code Generate and Consume
●   Fully Web Management Interface                              ●    Postpaid Service
●   Support Prepaid/Postpaid User                                      Call Detail Record Storage
●   Max Call Duration Protection                                       Effective Date/Expired Date
●   Immediate Response                                                 CDR Report
●   Share Secret with MD5 Protection                            ●    Flexible Rate Plan Support
●   Wellgate 5250 and WellSIP 6500 Full Interoperability               Up-to 5 Charge Segments per Rate Prefix
●   Automatic 6500 provision                                           Effective Date/Expired Date
●   Detail Access Log                                                  Longest Prefix Match
●   Provide Basic Reports                                              Programmable charge unit, amount and cycle
●   Support Coin Phone Service                                         Support Per Call Charge
●   Support Calling ID (ANI) Validation                                Call Screening
●   Support Charge Account                                             Holiday & Night Time Charge
●   Auto Monthly Charge Deduction                                      Free Monthly Minutes based on Prefix
●   Up-to 5 level User Management                                      Monthly Free Charge based on Prefix
      Administrator                                                    Deductible Monthly Fee
      Distributor                                               ●    External Database support
      Group Reseller                                                   MSSQL
      Reseller                                                         Built-in DB Connection Pool management
      Subscriber
                                                                           RADIUS Server
                                                                           Prepaid/Postpaid
                                                                           Reseller Management
                                                                           Rate Plan


                                                            WS6500
                                                            WS6500                             CDR Storage
                                                                                               Subscriber Storage
                                                                           Wellbilling 6600


                        PSTN
                                                        SIP VOIP Network

                                       WG5250                                                  Database Server




                                                  IAD                                      IP Phone
              WellXfer 6300
VOIP SERVER

              Conference Server

              Up-to 64 participants in a conference room, WellXfer 6300 provides you an
              unlimited conference service for scheduled, meet me, web initiating, phone
              initiating and virtual conference. Its intuited web controlling interface enables
              you to have full control of your conference: mute, volume, speak, coach, private
              talk…etc. You have all of controlling to your conference, no pain anymore.


              Features
                                                                   Application Example
              ●   SIP RFC 3261 Compliance
                                                                   ●   Enterprise Conference Server
              ●   draft-ietf-sip-cc-transfer-07
                                                                   ●   Conference Service Rental
              ●   RFC 4579 (without XML)
                                                                   ●   Click to Conference Service
              ●   Support G.711, G.729A, G.723.1
                                                                   ●   Web Calling Service
              ●   Support out of band DTMF
              ●   Full Web Management Interface
              ●   Up-to 64 Participants in a Conference Room
              ●   up-to 120 Universal DSP Resource per Chassis
              ●   Virtual Conference
                                                                                 64 parties
              ●   Web Initiated Conference                                      Conferencing
              ●   Phone Initiated Conference
              ●   Scheduled Conference
              ●   Meet Me Conference
              ●   Automatic Join Interactive Voice Response
              ●   Conference Information/Summary Notice by Email
              ●   Selectable Inbound or Outbound Subscriber
              ●   Support Discuss or Speaker (Master) mode
              ●   Conference on-demand Recording
              ●   Support Conference Join Prompt
              ●   Local User Profile or RADIUS Authentication
              ●   Web Conference Controlling Interface
                    Web Base GUI
                    Speaking Request Notice
                    On-Demand Recording
                    Coach
                    Mute
                    Stop Conference
                    Add Participant
                    Kick Out
                    Private Talk
                    Volume Control
                    Text Chatting
              ●   RADIUS Authentication and Billing Message
              ●   Flat CDR File in Local Storage
              ●   Recording Storage 240GB (RAID 5)
WellSC 5900




                                                                                                            VOIP SERVER
Universal SIP/H.323 Signal Converter
WellSC 5900 is a universal signal converter for
H.323 and SIP. It not only provides simple SIP to H.323
conversion but also DTMF conversion. For H.323 users, it normally
use H.245 for DTMF relay while SIP is mainly using RFC 2833. WellSC
5900 takes care of the conversion. Beside the SIP to H.323 conversion,
WS6500 can also support SIP to SIP, H.323 to H.323 call without any limitations.

WellSC 5900 provides an easy to use web-based drag and drop call-flow editor
to help the system administrator to design their VOIP service very quick and
flexible. You build your business intelligent on WellSC 5900.

Key Features
●   Signal Convert for SIP RFC 3261 and H.323 V5            ●   Provides Call Detail Record

●   H.323 to SIP call                                       ●   Full Web Management Interface

●   SIP to H.323 call

●   SIP to SIP call                                         Application Example
●   H.323 to H.323 call                                     ●   Internet Exchange Center Traffic Hiding

●   Support up-to 16 Multiple SIP Proxy Servers             ●   H.323 Migration to SIP

●   Support SIP Proxy, Gatekeeper and P2P Calls             ●   Connect to H.323 Service Provider/Trunk

    Simultaneously                                          ●   Play as a SIP Trunk for Service Center

●   Support Route and un-Route RTP mode

●   H.323 DTMF Relay and SIP RFC 2833

    Conversion (RTP Routed)

●   H.323 DTMF Relay and SIP INFO Conversion

    (RTP un-Routed)

●   Support H.323 and SIP Video

●   Built-in Universal VOIP Address Book

●   Support H.323 to SIP non Fast Start (Delay Media)

●   Support H.323 to SIP Fast Start mode (Early Media)                       SIP Domain

●   Support RADIUS Authentication, Authorization

    and Accounting                                                                                 H.323
                                                         SIP Domain                               Network
●   Flexible Digit Manipulation Plan                                                wellsc 5900


●   Support Calling/Called Number Replacement
                                                                              H.323
●   Support IP, ANI, DNIS based Access Control
                                                                             Network
●   Support Flexible IP Routing and Account Code

●   Dynamic Call Treatment based on Drag and

    Drop Call Flow Editor
              WellGate 5260
VOIP SERVER

              A Universal VOIP Gateway

              Wellgate 5260 is a UNIVERSAL VOIP GATEWAY which navigates the calls in
              between H.323, SIP and PSTN freely, not simple PSTN to VOIP calls or vice verse.
              It can easily implement SIP and PSTN, H.323 and PSTN, SIP and H.323, SIP and
              SIP, H.323 and H.323 calls simultaneously. With built-in PSTN/VOIP IVR helps
              service provider to create their own voice service very quick.

              Wellgate 5260 provides an easy to use web-based managing interface. Administrator
              can use the web based drag and drop call-flow editor to design their VOIP service
              very quick and without losing any flexibilities. A web-based voice prompt management
              GUI is also provided to simplify the deployment of IVR related service.

              Key Features
              ●   Navigate Call Freely in SIP, H.323 and PSTN
              ●   Support SIP RFC 3261 and ITU-T H.323 V5

                  Simultaneously
              ●   Up to 4 Programmable E1/T1 Trunks
              ●   PSTN Signaling: ISDN/PRI, CAS, MFC R2, QSIG , SS7
                                                                                                  PSTN
              ●   Support Audio Codec G.711, G.723.1, G.729A, GSM
              ●   SIP to PSTN Call and vice versa
              ●   H.323 to PSTN Call and vice versa
                                                                                                          WG 5260   H.323 GK
                                                                                    WS 6500
              ●   H.323 to SIP Call and vice versa
              ●   SIP to SIP Call
              ●   H.323 to H.323 Call
                                                                                SIP Network                    H.323 Network
              ●   Support up-to 16 Multiple SIP Proxy Servers
              ●   Support SIP Proxy, Gatekeeper and P2P Calls

                  Simultaneously

              ●   Built-in Universal VOIP Address Book
                                                                       IP Phone         IAD              IAD             IP Phone
              ●   Support Early Media and SIP Delay Media
              ●   Support RADIUS Authentication, Authorization

                  and Accounting
              ●   Intelligent PSTN Call Routing and in-Trunk Hunting
                                                                            ●     Built-in IVR & call-flow editor
              ●   Support Flexible VOIP Routing and Account Code
              ●   Flexible Digit Manipulation Plan
              ●   Support Calling/Called Number Replacement
              ●   In-band and out of Band DTMF Transmission
              ●   T.38 Fax Relay up to 14400 bps
              ●   Dynamic Call Treatment Based on Drag and Drop

                  Call Flow Editor
              ●   Built-in PSTN and VOIP IVR
              ●   Provides Call Detail Record
              ●   Full Web Management Interface
WellGate 5500




                                                                                                                      VOIP SERVER
A cPCI based Universal VOIP Gateway
Wellgate 5500 is a cPCI(Compact PCI) based UNIVERSAL VOIP
GATEWAY which navigates the calls in between H.323, SIP and
PSTN freely, not simple PSTN to VOIP calls or vice verse. It can easily
implement SIP and PSTN, H.323 and PSTN, SIP and H.323, SIP and SIP, H.323
and H.323 calls simultaneously. With built-in PSTN/VOIP IVR helps service
provider to create their own voice service very quick.

Wellgate 5500 provides an easy to use web-based managing interface.
Administrator can use the web based drag and drop call-flow editor to design
their VOIP service very quick and without losing any flexibilities. A web-based
voice prompt management GUI is also provided to simplify the deployment of IVR
related service.

Key Features
●   Compact PCI 2U Chassis, 19 inches
●   Up to 16 Programmable E1/T1 Trunks
●   Navigate Call Freely in SIP, H.323 and PSTN
●   Support SIP RFC 3261 and ITU-T H.323 V5
                                                                                   PSTN
    Simultaneously
●   PSTN Signaling: ISDN/PRI, CAS, MFC R2, QSIG , SS7
●   Support Audio Codec G.711, G.723.1, G.729A
●   SIP to PSTN Call and vice versa                                  WS 6500              WG 5500
                                                                                                      H.323 GK

●   H.323 to PSTN Call and vice versa
●   H.323 to SIP Call and vice versa
●   SIP to SIP Call                                             SIP Network                     H.323 Network
●   H.323 to H.323 Call
●   Support up-to 16 Multiple SIP Proxy Servers
●   Support SIP Proxy, Gatekeeper and P2P Calls
    Simultaneously
                                                         IP Phone        IAD              IAD              IP Phone
●   Built-in Universal VOIP Address Book
●   Support Early Media and SIP Delay Media
●   Support RADIUS Authentication, Authorization and
    Accounting
●   Intelligent PSTN Call Routing and in-Trunk Hunting
●   Support Flexible VOIP Routing and Account Code            ●     Built-in IVR & call-flow editor
●   Flexible Digit Manipulation Plan
●   Support Calling/Called Number Replacement
●   In-band and out of Band DTMF Transmission
●   T.38 Fax Relay up to 14400 bps
●   Dynamic Call Treatment Based on Drag and Drop
    Call Flow Editor
●   Built-in PSTN and VOIP IVR
●   Provides Call Detail Record
●   Full Web Management Interface
              PRO 9510
VOIP SERVER

              Making the CPE Management Easy
                                                                                                Coming soon


              Pro 9510 is a dedicated Element Management System for Welltech CPE products.
              It supports fulfilled features to upgrade firmware or change configuration of
              managed devices automatically either by scheduled or real time based. With the
              updated status reports, administrator can easily know the exactly device status. It
              makes the CPE management easily.


                                                                                                          Remote Maintenance
              Key Features
                                                                                                             Reboot
              ●   Web Management

              ●   Firmware Version Management
                                                                                            PRO 9510
              ●   Support Behind NAT Device                                 Internet

              ●   Scheduled Update by Server or Client

              ●   Parameter Management
                                                                                                      Refresh Profile
              ●   Device Type Level Parameters
                                                                            NAT
              ●   Device Level Parameters

              ●   Device Profile Update                        Private IP Device   Public IP Device

              ●   Device Firmware Upgrade

              ●   Device Alarm Trap

              ●   Device Detail Status Enquiry
                                                                                                          Real Time Debugging
              ●   Device Remote Control for Maintenance

              ●   Device Debugging

              ●   Device Debug Information Logging by Syslog                                   PRO 9510

              ●   Extendable Event/Alarm Log                                Internet

              ●   Provisioning and Access Reports




                                                                            NAT




                                                               Private IP Device   Public IP Device
SIPPBX 6200/6200S
The Best Choice of Your IP-PBX




                                                                                                                                            IP-PBX
The Welltech SIPPBX 6200 is an all in one IP-PBX
which including PABX telephony service, Auto Attendant,
Voice Mail, Enterprise Coloring Ring Back Tone, Conference,
Announcement and VOIP router features together. It supports up-to 1000
subscribers and is a cost effective solution for small to large enterprise. Also the
traveler soft-phone, operator console and billing software provide you a completed
transition from traditional PABX to the new generation IP-PBX.




                                   SIPPBX 6200                                                             SIPPBX 6200S


 ●    SIP RFC 3261 Compliance                                          Selected Telephony Features
 ●    Support G.711, G.729A, G.723.1 *                                 ●   Call Transfer
 ●    Support SIP/RTP Encryption/Decryption                            ●   Call Forward
 ●    Support RADIUS Server or Enterprise Billing via TCP              ●   Call Forwarded Notice
 Auto Attendant                                                        ●   Call Screening
 ●    Web-Base Auto Attendant Flow Editor                              ●   Caller ID Privacy
 ●    Scheduled Special Announcement                                   ●   Call Waiting
 ●    Holidays Working Time Support                                    ●   Call Hold
 ●    Multiple Language Support                                        ●   Call Pickup (Global, Group)
 ●    Support Branch Office                                            ●   Specified Call Pickup
 ●    Support Transit Call                                             ●   Find Me
 Voice Mail                                                            ●   Short Code
 ●    Web-Base Voice Mail Flow Editor                                  ●   Do Not Disturb
 ●    Personal Greeting                                                ●   Miss Call Notify by Email
 ●    Multiple Language Support                                        ●   ANI Replacement
 ●    Native TTS (Chinese & English & Japanese) Support                ●   Call Return
 ●    Support Additional Customized TTS Language                       ●   Hide ANI/Show ANI Selection
 ●    Message Waiting Indication                                       ●   Call Park/Retrieve
 ●    Email Notify                                                     ●   Call Camp on
 ●    Web Retrieve                                                     ●   Display Name Replacement
 ●    Phone Retrieve                                                   ●   PSTN Number
 Conference Bridge                                                     ●   Ring PSTN & IP Device Simultaneously
 ●    Support RFC 4579 (without XML)                                   ●   Broadcasting Service
 ●    Ad-Hoc Conference                                                ●   Recording on demand **
 ●    Virtual Conference (Meeting Me)
 ●    Virtual Conference (Ad-hoc)
 ●    Event Tone Notice
 ●    Up-to 16 Participants per Room(6200)                                                                           Traveling User
 ●    Up-to 8 parties for softwave DSP version(6200S)                                               Traveling User
                                                                    Remote Office
 ●    Quick Conference by Soft-phone (SP362)
 Enhanced Service
 ●    System Announcement Service                                                           SIPPBX-6200S
                                                            Ext. 2004 Ext. 2003 Ext. 2002            WAN/Internet
 ●    Company-wide Coloring Ring Back Tone Service
 Voice Router                                                                                                          Customer & Partner

 ●    Public and Private IP Legs
 ●    SIP-Aware RTP Routing                                                                                           SIPPBX-6200
                                                                                 Enterprise Network
 ●    Natural VOIP Firewall/NAT


  Optional Features                                                Ext. 1004 Ext. 1003 Ext. 1002 Ext. 1001
                                                                                                                                  WG5250

  ●   Traveler Soft-phone (SP 362)
  ●   Operator Console Software
  ●   Enterprise Billing Software                                                                                     PSTN
  ●   Web Caller Module
  ●   Microsoft Live Communicator/
      Exchange 2007 Module
         ePBX-100/100A                                                                     NEW
         An economical IP-PBX Selection
IP-PBX


         The Welltech ePBX-100/100A is the next generation IP PBX system for small
         enterprise. It is also designed to operate on a variety of VoIP applications,
         such as auto-attendant, voice conference, call transfer, call pick up and IP-based
         communications. With the tiny box, small to medium enterprise or homes can use
         it to access the Internet and to make VoIP phone calls.

         Customers can select different suite and optional products to meet their request.
         To Integrate with WellGate 3804A/3802A can provide PSTN access function, IP
         Phone LP-388, LP-399,LP-305 and WellGate 3504A/3502A can provide extensions.
         With flexible and full functionality, Welltech ePBX-100/100A give a complete
         transition from traditional PABX to the new generation IP-PBX.


         Specifications
                                                                Technical Features
         Protocol
                                                                ●   Subscriber NAT traversal
         ●   SIP (Session Initiation Protocol)
                                                                ●   Phone set record Greeting
                                                                ●   Management: Web Browser Management
         Call Features                                          ●   HTTP upgrade firmware and ring back tone file
         ●   Authentication                                     ●   Export/Import configuration
         ●   Automated Attendant                                ●   Network Interface: 1WAN 1LAN
         ●   Call Transfer (IP Phone, WG3504A) *                ●   Support Voice Codec: G.729, G.711μ, G.711A
         ●   Blind Transfer (IP Phone, WG3504A) *               ●   DTMF: In-band, RFC 2833, SIP-Info
         ●   Call Forward on Busy (IP Phone, WG3504A) *         ●   Network: Support Fixed IP,DHCP mode,
         ●   Call Forward on No Answer (IP Phone, WG3504A) *                 and PPPoE Mode
                                                                    Built-in 1G storage (ePBX-100A)
             Call Forward Unconditional (IP Phone, WG3504A) *
                                                                ●
         ●

         ●   Call Hold/Retrieval (IP Phone, WG3504A) *
                                                                Capacity
         ●   Call Routing
                                                                ●   100 register
         ●   Call Waiting (IP Phone, WG3504A)*                  ●   13 concurrent calls
         ●   Caller ID (IP Phone, WG3504A) *
         ●   Do Not Disturb (IP Phone) *                         Dimension
         ●   Flexible Extension Logic                           ●   17.5 x 12.5 x 3.2 cm
         ●   Music On Hold
         ●   Music On Transfer                                                     Order Information
         ●   Call Pickup                                            Model                                    Version
         ●   Three-way Conference (IP Phone) *                  ePBX-100       Standard Version
         ●   Time and Date
                                                                ePBX-100A Voice Mail+1G Storage
         ●   Trunking (WG3804A)
         ●   VoIP Gateways (WG3804A)
                                                                      ITSP
         ●   Voice Mail to e-mail
         ●   Voice Mail (ePBX-100A)                                          Internet

         ●   Call Detail Records                                                                     SoftPhone




         It is strongly suggested to purchase fully
                                                                                    ePBX-100
         compatible products as below
                                                                                                                     PSTN
         ●   Welltech IP Phone:LP-388                                                                     WG 3804A
                                                                                               WG 3504A

         ●   Welltech FXS Gateway: WellGate 3502A/3504A
         ●   Welltech FXO Gateway: WellGate 3802A/3804A


         *   IP Phone: LP-388, LP-399, LP-305/305A
                                                                                                                  VOIP SERVER
Wellgate 5290
A Universal VOIP/SS7 Gateway
Wellgate 5290 is a universal VOIP/SS7 GATEWAY
which navigates the calls in between H.323, SIP and PSTN
freely, not simple PSTN to VOIP calls or vice verse. It can be easily
used to implement SIP and PSTN, H.323 and PSTN, SIP and H.323, SIP
and SIP, H.323 and H.323 calls simultaneously. With built-in SS7/ISUP interface,
Wellgate 5290 bridges traditional PSTN/SS7 to VOIP network. Also it provides
PSTN/VOIP IVR helps service provider to create their own voice service very
quick.

Wellgate 5290 provides an easy to use web-based managing interface.
Administrator can use the web based drag and drop call-flow editor to design
their VOIP service very quick and without losing any flexibilities. A web-based
voice prompt management GUI is also provided to simplify the deployment of IVR
related service.

Key Features
●   Navigate Call Freely in SIP, H.323 and PSTN
●   Support SIP RFC 3261 and ITU-T H.323 V5
    Simultaneously
●   Up to 4 Programmable E1/T1 Trunks
●   PSTN Signaling: ISDN/PRI, CAS, MFC R2,
                                                                       SS7/ISUP/Voice E1
    QSIG, SS7/ISUP
●   Support with Voice or Signal only SS7 Link
●   Support Audio Codec G.711, G.723.1,                      PSTN                      Voice E1
                                                                                       .......



    G.729A, GSM                                                                                   Wellgate 5290
●   SIP to PSTN Call and vice versa
●   H.323 to PSTN Call and vice versa
●   H.323 to SIP Call and vice versa
●   SIP to SIP Call
●   H.323 to H.323 Call
●   Built-in Universal VOIP Address Book
●   Support up-to 16 Multiple SIP Proxy Servers
●   Support SIP Proxy, Gatekeeper and P2P Calls
                                                         ●   Built-in IVR & call-flow editor
    Simultaneously
●   Support Early Media and SIP Delay Media
●   Support RADIUS Authentication, Authorization
    and Accounting
●   Intelligent PSTN Call Routing and in-Trunk Hunting
●   Support Flexible VOIP Routing and Account Code
●   Flexible Digit Manipulation Plan
●   Support Calling/Called Number Replacement
●   In-band and out of Band DTMF Transmission
●   T.38 Fax Relay up to 14400 bps
●   Dynamic Call Treatment Based on Drag and Drop
    Call Flow Editor
●   Built-in PSTN and VOIP IVR
●   Provides Call Detail Record
●   Full Web Management Interface
              Welltech Web Call Solution
VOIP SERVER

              Freely Contacting Anywhere

              By using Welltech Web Call Solution, It provides an easy and free way for your
              customers or supplier to contact you anywhere. It is based on the standard SIP
              protocol and can be running under the popular Windows platform.

              With the provided OCX component, the customer can be easily integrate into
              their owned Web server to provide the web call service. The centralized Web
              Call Solution can be shared for all of their customers of VOIP service provider
              as well as enterprise users.



              Features                                                    Support Multiple Companies
              ●   Provide Web Call OCX for Web
                                                           ● Company MIS Designed Web
                  Integration

              ●   Support SIP RFC 3261                                           Company A
                                                                                                                                              Service Center


              ●   Codec Support G.711, G.729A
                                                                                                                               WellSIP 6500              WellBilling 6600
              ●   Support Acoustic Echo Cancellation

              ●   Support Non-Registered Call                                                                                                    4 Route to Company B
                                                                                                                               5. Talking

                  Support Automatically NAT Traversal
                                                                                                   3. Make Call
              ●
                                                                                                                                                  Company B

              ●   Support MD5 Authentication
                                                                                             2. Download Web Call OCX
              ●   Support Call Hold & Call Transfer               1. Click Talk to
                                                                    Company B
              ●   Support Max Current Calls per Customer

              ●   Support DNIS Screening

              ●   Support Billing Charge                   ● Using URL Redirect (Hosted Web)

              ●   Fully Integrated with WellSIP 6500 or
                                                                                     Company A                                                  Service Center
                  SIPPBX 6200                                                                                5 Route to Company B


                                                                                                                                                                 WellBilling 6600



                                                                                                                               Hosted Web Server
              Application Examples                         2. Redirect to Hosted      6. Talking
                                                                                                                                                                 WellSIP 6500

                                                           Company A Talk To me

              ●   Call Me Now Center                                                    3.Show Company A Click to Talk
                                                                                          Download Web Call OCX


              ●   Prepaid/Postpaid Web Dialer                                                                                                        Company B

                                                                                           4. Make Call
              ●   Personal Web Talk Link

              ●   Hosted Web Call Service                       1. Click Talk to
                                                              Company B Web Page
Soft Phone SP-365




                                                                                             SOFT PHONE
Making your VOIP Service Easy and Unique


Welltech SP365 is a dedicate soft-phone designed for VOIP service provider.
With the built-in Account Balance Display, Voice Mail Indicator and Text
Message Chatting, SP365 is a best choice for running the VOIP service.
It is not only providing required features for service provider, but also
granting them to customize their own features such as customized skin,
service URL and local language. It makes your VOIP service easy and unique.


Features
●   SIP RFC 3261 Compliance
●   Codec Support: G.711, G.729A
●   Support Acoustic Echo Cancellation
●   Support Multi-language User Interface
●   Account Balance Display
●   Text Message Send/Receive
●   Phone Book
●   Call Hold
●   Call Transfer (Transferred)
●   Call Mute
●   Missed Call
●   Call History
●   Message History
●   SIP Voice Mail Message Indicator
●   Quick Voice Mail Access Button
●   Coloring Ring Tone, Ring Back Tone and Busy Tone
●   Customizable Service or Home URL
●   Customizable Skin and Button
●   Customizable Local Language
●   Support Welltech Proprietary SIP/RTP
    Encryption/Decryption
                                                                     e
●   Fully Integrated with WellSIP 6500 SIP Proxy

                                                               1         2     3
                                                               4         5     6

Recommended System Requirements                                7         8     9

                                                               *         0     #
●   Intel Pentium 4 or above
●   256 MB RAM
●   20 MB HD space                                     * Contact Welltech for Availability
●   Welltech USB phone or Headset/Microphone
●   Windows 2000/XP
          WellGate                                3502A/3504A
          WellGate 3502A/3504A FXS is a two-port/four-port telephone extension to IP
          network gateway and support H.323 v2/v3/v4 or SIP RFC 3261 protocol. It provides
GATEWAY



          telephone services and T.38 fax over IP network with easy operation and
          configuration for ITSP / ISP (Internet Telephony Services Provider) and Office /
          SOHO IP-PBX application.

          Benefits
          ●    Easy access to IP from phone set or PBX
          ●    Cost Saving - Telephone call from VPN or public Internet
          ●    Follows the existing telephone call dial plan
          ●    Easy interface to ADSL/Cable Modem or Leased line
               equipment
          ●    Easy to integrate with all kinds of IP-PBXs
          ●    Provision Management via Welltech PRO-9500 Server


          Specification

                                              H.323 Version                                 SIP version
                Protocol       ●   ITU-T H.323 v2/v3/v4 compliance          ●   SIP RFC3261
          Compatible Server    ●   GnuGK(Behind NAT)                        ●   OpenSER Asterisk
                               ●   Call Hold, Call Transfer, Call           ●   Call Hold
               Telephony           Forward (H.450)                          ●   Call Transfer
               Features
                                                                            ●   Call Forward
                               ●   Automatically FAX detection              ●   Automatically FAX detection
              FAX Support      ●   T.38 G3 FAX                              ●   T.38 G3 FAX
                               ●   Group 3 Fax relay at 2.4 - 14.4 kbps     ●   Group 3 Fax relay at 2.4 - 14.4 kbps

              Audio Codec      ●   G.711A/μ-law, G.723.1, G.729A, G.729     ●   G.711A/μ-law, G.723.1, G.729A, G.729
              Support              G.729B, G.729AB                              G.729B, G.729AB

                Network
                               ●   10/100Base-T Ethernet RJ-45 port x 2     ●   10/100Base-T Ethernet RJ-45 port x 2
                Interface          for WellGate3502A/3504A                      for WellGate3502A/3504A
                               ●   RJ-11 Telephone port (FXS) x 2 for       ●   RJ-11 Telephone port (FXS) x 2 for
                                   WellGate 3502A                               WellGate 3502A
                                                                            ●   RJ-11 Telephone port (FXS) x 4 for
                               ●   RJ-11 Telephone port (FXS) x 4 for
                                                                                WellGate 3504A only
                                   WellGate 3504A only                      ●   2-wire loop start
                               ●   2-wire loop start                        ●   Programmable AC impedance, Feeding
                               ●   Programmable AC impedance, Feeding           voltage, Ring Voltage, Ring Cadence,
              FXS Interface                                                     Loop current and call progress tone
                                   voltage, Ring Voltage, Ring Cadence,
                                                                            ●   ON-Hook Voltage: 48Vdc
                                   Loop current and call progress tone      ●   Ring Voltage: 50 V RMS
                               ●   ON-Hook Voltage: 48Vdc                   ●   Loop Current: Constant 23mA
                               ●   Ring Voltage: 50 V RMS                   ●   Support Payphone Charge signal
                                                                                12K/16K/Polarity Reversal
                               ●   Loop Current: Constant 23mA
                               ●   VAD (Voice Activity Detection)           ●   VAD (Voice Activity Detection)

              Voice Quality    ●   CNG (Comfort Noise Generation)           ●   CNG (Comfort Noise Generation)
                               ●   AEC (Acoustic Echo Cancellation)-G.168   ●   AEC (Acoustic Echo Cancellation)-G.168
                                                                                                                            GATEWAY
                    ●   Dynamic Jitter Buffer                                ●    Dynamic Jitter Buffer
     QoS            ●   DiffServ                                             ●    DiffServ
   Caller ID        ●   FSK (Bellcore)/DTMF generation                       ●    FSK (Bellcore)/DTMF generation
 Console Port       ●   1 D-SUB 9 pin RS-232 port                            ●    1 D-SUB 9 pin RS-232 port
                    ●   DTMF / CPT (Call Progress Tone)                      ●    DTMF / CPT (Call Progress Tone)
     Tone
                        generation/detection                                      generation
                    ●   Statically and DHCP for IP address                   ●    Statically and DHCP for IP address
   Network              assignment                                                assignment
   Support          ●   Behind NAT Router or IP sharing device               ●    Behind NAT Router or IP sharing device
                    ●   PPPoE                                                ●    PPPoE
                    ●   Command line interface and Web                       ●    Command line interface and Web
   Security
                        management interface Password protected                   management interface Password protected

Configuration &     ●   Console port, TELNET and Web Browser                 ●    Console port, TELNET,Web Browser,and
Management                                                                        Welltech Provision Server
                    ●   Firmware upgrade through network by                  ●    Firmware upgrade through network
System Upgrade
                        TFTP/FTP                                                  by TFTP/FTP
    Power           ●   Input AC 100V~240V Output DC12V.                     ●    Input AC 100V~240V Output DC12V
                    ●   FCC Part 15, CE Class B, VCCI Class                  ●    FCC Part 15 Class B, CE Class B, VCCI
                    ●   FCC Part 15 Class B, CE Class B,                          Class
 Certification
                        VCCI Class
Operation Temp      ●   0° C to 40° C                                        ●    0° C to 40° C
   Humidity         ●   10% to 90% (Non-condensing)                          ●    10% to 90% (Non-condensing)
                    ●   Dimension: 223mm(W) x 35mm(H)                        ●    Dimension: 223mm(W) x 35mm(H)
  Dimension             x 152mm(D)                                                x 152mm(D)
  and Weight
                    ●   Weight (unit): 1.4 kg                                ●    Weight (unit): 1.4 kg


       ●   WellGate 3504A Application Diagram




                            Branch                                                                Headquarter


                                              Router          Internet or            Router
                             WellGate 3504A                   Intranet
           EXT       PBX                                                                      WellGate 3504A    PBX   EXT

                                     FAX
                                                                    Router


                                                       SOHO                                          Phone



                                                                 WellGate 3504A                       FAX
          WellGate                               3701B / 3702B (FXS+FXO)

          WellGate 3701B/3702B is a one-port/two-port FXS + FXO gateway and support
          H.323 v2/v3/v4 or SIP RFC 3261 protocol. It supports an innovative intelligent call
GATEWAY



          routing function that transparently routes calls to destination either through PSTN
          or internet. WellGate 3701B/3702B provides Voice over IP and FAX over IP services
          for ITSP / ISP (Internet Telephony Services Provider) and Office / SOHO IP-PBX
          application.
          Benefits
          ●    Easy access to IP from phone set or PBX
          ●    Cost Saving - Telephone call from VPN or public Internet
          ●    Follows the existing telephone call dial plan
          ●    Easy interface to ADSL/Cable Modem or Leased line equipment
          ●    Easy to integrate with all kinds of IP-PBXs
          Specification

                                              H.323 Version                                    SIP version
                Protocol      ●   ITU-T H.323 v2/v3/v4 compliance          ●   SIP RFC3261

          Compatible Server   ●   GnuGK(Behind NAT)                        ●   OpenSER, Asterisk
                              ●   Call Hold, Call Transfer, Call Forward   ●   Call Hold
               Telephony          (H.450)                                  ●   Call Transfer
               Features
                                                                           ●   Call Forward
                              ●   Automatically FAX detection              ●   Automatically FAX detection
                              ●   Group 3 Fax relay at 2.4 - 14.4 kbps     ●   Group 3 Fax relay at 2.4 - 14.4 kbps
                              ●   Support T.38 protocol                    ●   Support T.38 protocol
                              ●   Support T.38 ECM: Error correction       ●   Support T.38 ECM: Error correction
              FAX Support         during the high speed mode                   during the high speed mode
                              ●   Support T.38 FAX Redundancy Depth        ●   Support T.38 FAX Redundancy Depth
                              ●   Support Abstract Syntax Notation 1       ●   Support Abstract Syntax Notation 1
                                  (ASN.1)                                      (ASN.1)
                              ●   G.711 FAX Mode                           ●   G.711 FAX Mode

              Audio Codec
                              ●   G.711A/μ-law, G.723.1, G.729A, G.729     ●   G.711A/μ-law, G.723.1, G.729A, G.729
              Support             G.729B, G.729AB                              G.729B, G.729AB
          Network Interface   ●   Two 10/100Base-T Ethernet RJ-45 port     ●   Two 10/100Base-T Ethernet RJ-45 port
                              ●   One Telephone (FXS) RJ-11 port for       ●   One Telephone (FXS) RJ-11 port for
                                  WellGate 3701B                               WellGate 3701B
           FXS Interface
                              ●   Two Telephone (FXS) RJ-11 ports for      ●   Two Telephone (FXS) RJ-11 ports for
                                  WellGate 3702B                               WellGate 3702B
                              ●   One Analog PSTN (FXO) RJ-11 port for     ●   One Analog PSTN (FXO) RJ-11 port for
                                  WellGate 3701B                               WellGate 3701B
           FXO Interface
                              ●   Two Analog PSTN (FXO) RJ-11 ports for    ●   Two Analog PSTN (FXO) RJ-11 ports for
                                  WellGate 3702B                               WellGate 3702B
                              ●   Support auto-attendant (Tone or voice    ●   Support auto-attendant (Tone or voice
                                  greeting)                                    greeting)
           FXS Features
                              ●   PSTN polarity reversal detection         ●   PSTN polarity reversal detection
                                                                                                               GATEWAY
                  ●   Provide 2nd dial tone to PSTN             ●     Provide 2nd dial tone to PSTN
                  ●   Disconnect tone detection                 ●     Disconnect tone detection
                  ●   Support auto-attendant (Tone or voice     ●     Support auto-attendant (Tone or voice
                      greeting)                                       greeting)
 FXO Features     ●   PSTN polarity reversal detection          ●     PSTN polarity reversal detection
                  ●   Provide 2nd dial tone to PSTN             ●     Provide 2nd dial tone to PSTN
                  ●   Disconnect tone detection                 ●     Disconnect tone detection
                  ●   VAD (Voice Activity Detection)            ●     VAD (Voice Activity Detection)
                  ●   CNG (Comfort Noise Generation)            ●     CNG (Comfort Noise Generation)
                  ●   AEC (Acoustic Echo Cancellation)          ●     AEC (Acoustic Echo Cancellation)
                      -- G.168/165                                    -- G.168/165
     Voice
                  ●   Dynamic Jitter Buffer                     ●     Dynamic Jitter Buffer
                  ●   Completed voice band signaling support    ●     Completed voice band signaling support
                  ●   Provide In-band or Out-band DTMF          ●     Provide In-band or Out-band DTMF
                      generation/detection                            generation/detection
                  ●   QOS by setting DSCP (Differentiated       ●     QOS by setting DSCP (Differentiated

     QoS              Service Code Point) parameters of VoIP          Service Code Point) parameters of VoIP
                      packet                                          packet
                  ●   FSK(Bellcore)/DTMF detection and          ●     FSK(Bellcore)/DTMF detection and
   Caller ID
                      generation                                      generation

 Console Port     ●   1 D-SUB 9 pin RS-232 port                 ●     1 D-SUB 9 pin RS-232 port
     Tone         ●   Provide call progress tone                ●     Provide call progress tone
                  ●   Support Fixed IP and DHCP                 ●     Support Fixed IP and DHCP
   Network        ●   PPPoE                                     ●     PPPoE
   Support
                  ●   Behind NAT Router or IP sharing device    ●     Behind NAT Router or IP sharing device

Configuration &   ●   Console port, TELNET and Web Browser      ●     Console port, TELNET and Web Browser
Management            configuration                                   configuration
System Upgrade    ●   TFTP/FTP software upgrade                 ●     TFTP/FTP software upgrade
    Power         ●   Input AC 100V~240V Output DC12V.          ●     Input AC 100V~240V Output DC12V.
Operation Temp    ●   5° C to 40° C                             ●     5° C to 40° C

   Humidity       ●   10% to 90% (Non-condensing)               ●     10% to 90% (Non-condensing)
                  ●   Dimension: 223mm(W) x 35mm(H)             ●     Dimension: 223mm(W) x 35mm(H)
  Dimension           x 152mm(D)                                      x 152mm(D)
  and Weight
                  ●   Weight (unit): 1.4 kg                     ●     Weight (unit): 1.4 kg



                             Internet                                                   Internet




                FXO                        FXS                            FXO                           FXS
     PSTN                                                      PSTN


                        WellGate 3701B                                             WellGate 3702B
                        SOHO Application                                           Enterprise Application
          WellGate                                 3802A/3804A (FXO)
          WellGate 3802A/3804A is a two/four-FXO-port Gateway, which represents an interface
          to access PSTN and provides voice and fax over IP services. By connecting with ADSL
GATEWAY



          or cable Modem, WellGate 3802A/3804A provides saving in network infrastructure and
          greatly reduced telephone charges. With H.323 v2/v3/v4 or SIP protocol, Welltech
          create ideal solution for providing low cost communications between headquarters and
          branch offices in the world, as well as for SOHO and office telephony applications.


          Benefits
          ●    Easy access to PSTN from IP
          ●    Cost saving - save long and international call expense by using IP network.
          ●    No extra investment - just take advantage of existing Ethernet LAN and IP environment.
          ●    Easy interface with Cable Modem, ADSL Modem and Leased Line equipment
          ●    Easy to integrate with all kinds of IP-PBXs




          Specification

                                             H.323 Version                                      SIP version

                Protocol      ●   ITU-T H.323 v2/v3/v4 compliance            ●   SIP RFC3261
          Compatible Server   ●   GnuGK(Behind NAT)                          ●   OpenSER Asterisk
                              ●   Call Hold, Call Transfer, Call Forward     ●   Call Hold
               Telephony
               Features           (H.450)                                    ●   Call Transfer
                                                                             ●   Call Forward

              FAX Support     ●   Provide T.38/FAX and voice                 ●   Provide T.38/FAX and voice
                                  Auto-Switching                                 Auto-Switching

              Audio Codec     ●   G.711A/μ-law, G.723.1, G.729A, G.729       ●   G.711A/μ-law, G.723.1, G.729A, G.729
              Support
                                  G.729B, G.729AB                                G.729B, G.729AB
          Network Interface   ●   Two 10/100Base-T Ethernet RJ-45 port       ●   Two 10/100Base-T Ethernet RJ-45 port
                              ●   Two Analog PSTN (FXO) RJ-11 port for       ●   Two Analog PSTN (FXO) RJ-11 port for
                                  WellGate 3802A                                 WellGate 3802A
           FXO Interface      ●   Four Analog PSTN (FXO) RJ-11 port for      ●   Four Analog PSTN (FXO) RJ-11 port for
                                  WellGate 3804A                                 WellGate 3804A
                              ●   VAD (Voice Activity Detection)             ●   VAD (Voice Activity Detection)
                              ●   CNG (Comfort Noise Generation)             ●   CNG (Comfort Noise Generation)
                              ●   G.168/165-compliant adaptive echo          ●   G.168/165-compliant adaptive echo
                                  cancellation                                   cancellation
                              ●   Dynamic Jitter Buffer                      ●   Dynamic Jitter Buffer
                              ●   Bad Frame Interpolation                    ●   Bad Frame Interpolation

              Voice Quality
                              ●   Completed voice band signaling support     ●   Completed voice band signaling support
                              ●   Receive Caller ID (DTMF or FSK) from       ●   Receive Caller ID (DTMF or FSK) from
                                  PSTN                                           PSTN
                                                                             ●   PSTN Current Loop Break(CPC)detection
                                                                                 to release FXO port
                                                                                                                                   GATEWAY
                     ●   Provide Inband or Outband DTMF                    ●   Provide Inband or Outband DTMF
                         generation/detection between LAN and                  generation/detection between LAN and
                         PSTN interface                                        PSTN interface
                     ●   Gain/Attenuation Settings                         ●   Gain/Attenuation Settings
                     ●   QOS by setting DSCP (Differentiated               ●   QOS by setting DSCP (Differentiated

      QoS                Service Code Point) parameters of VoIP                Service Code Point) parameters of VoIP
                         packet                                                packet

   Caller ID         ●   Caller ID (FSK) detection                         ●   Caller ID (FSK) detection
                     ●   Asynchronous Console RS-232C DB-9                 ●   Asynchronous Console RS-232C DB-9
    Console
                         connector                                             connector
                     ●   DTMF detection/generation                         ●   DTMF detection/generation
                     ●   DTMF Dialing                                      ●   DTMF Dialing
     Tone            ●   Support auto-attendant (tone or voice             ●   Support auto-attendant (tone or voice
                         greeting)                                             greeting)
                     ●   Provide 2nd dial tone to PSTN                     ●   Provide 2nd dial tone to PSTN
                     ●   Fixed IP and DHCP                                 ●   Fixed IP and DHCP
   Network
   Support           ●   PPPoE                                             ●   PPPoE
                     ●   Behind NAT Router or IP sharing device            ●   Behind NAT Router or IP sharing device

Configuration &      ●   Console port, TELNET                              ●   Console port, TELNET
Management               and Web Browser                                       and Web Browser
                     ●   Firmware upgrade through network by               ●   Firmware upgrade through network by
   Upgrade
                         TFTP/FTP                                              TFTP/FTP
     Power           ●   Input AC 100V~240V Output DC12V                   ●   Input AC 100V~240V Output DC12V
                     For WellGate 3802/3804A︰                              For WellGate 3802/3804A︰
 Certification
                     ●   CE Class B, LVD                                   ●   CE Class B, LVD
Operation Temp       ●   5° C to 40° C                                     ●   5° C to 40° C
   Humidity          ●   10% to 90% (Non-condensing)                       ●   10% to 90% (Non-condensing)
                     ●   Dimension: 223mm(W) x 35mm(H)                     ●   Dimension: 223mm(W) x 35mm(H)
  Dimension
                         x 152mm(D)                                            x 152mm(D)
  and Weight
                     ●   Weight (unit): 1.4 kg                             ●   Weight (unit): 1.4 kg




                                                                                                             IP Nework
                                                     Netmeeting or H.323   Gatekeeper or
                  Netmeeting or H.323   LAN Phone                          Call Manager
   Analog Phone                                      Endpoint equipment
                  Endpoint equipment                                                        Router




        PSTN                                                  ETHERNET                                     ADSL Modem
                                                                                                                         IP Phone or
                                                                                                                         Gateway
                          WellGate 3802A/3804A FXO
          WellGate 2500 Series
          4FXS+PSTN, 4FXO, 2FXS+2FXO

          The WellGate 2500 series are 4-port FXS/FXO voice, fax, and modem over IP
GATEWAY



          gateway, which include three models. WG-2504 built-in 4 FXS ports and one
          PSTN backup line; WG-2540 built-in 4 FXO ports; WG-2522 built-in 2 FXS and
          2 FXO ports. When network fails, WG-2504 can pass call to PSTN port
          automatically. WG-2500 series provide ideal solution for providing low cost
          communications between headquarters and branch offices in the word.


          Benefits
          (1) Easy access to PSTN from IP

          (2) Cost saving - save long and international

             call expense by using IP network.

          (3) No extra investment - just take advantage of

             existing Ethernet LAN and IP environment.

          (4) Easy interface with Cable Modem, ADSL Modem

             and Leased Line equipment

          (5) Easy to integrate with all kinds of IP-PBXs

          Physical interface
          (1) RJ-45 (10/100 Base-T)

             A. WAN x 1 for connecting to HUB or ATU-R directly

             B. LAN x 1 for PC or other devices connection

          (2) RJ-11

             A. FXS x 4 / FXO x 4 / FXS x 2, FXO x 2

             B. PSTN x 1

          (3) Panel Status Indications:

             FXS / FXO, Ethernet, SIP System, PSTN

          Network and Protocol
          (1) SIP v2(RFC 3261)

             A. Outbound proxy

             B. Support backup proxy registration

             C. Support IP or domain name for primary and secondary

                proxy address and auto switching is enabled.

             D. Provide SIP URI Format,support both number and text.

          (2) TFTP Client/DHCP Client/PPPoE Client

          (3) Telnet/HTTP Server

          (4) Support T.38 FAX

          (5) DNS Client
                                                                                       GATEWAY
(8) Network support Static IP/ DHCP/ PPPoE    Environment
(9) NAT , DHCP server                         (1) AC Power
(10) Support DSCP for voice packet               100~240V, 12Vdc output
     quality control                          (2) Environment
                                                                    o     o
(11) Security                                    A. Temperature 0 C~40 C
    A. HTTP 1.1 basic/digest authentication      B. Humidity 10%~90%RH
        for Web setup                         Voice feature
     B. MD5 for SIP authentication            (1) Voice codec
       (RFC2069/RFC2617)                         A. G.711(A/μ-law): 64k bit/s (PCM)
(12) Support Session Timer(RFC 4028)             B. G.723.1: 6.3k/5.3k bit/s
Call function                                    C. G.729A: 8k bit/s (CS-ACELP)
(1) Call Hold                                    D. G.729B: adds VAD & CNG to G.729
(2) Call Waiting                              (2) VAD : Voice activity detection
(3) Call Forward                              (3) CNG : Comfortable noise generation
(4) Caller ID(FSK, DTMF)                      (4) G.165/168 Echo canceller
(5) Transfer                                  (5) Configurable Jitter Buffer
(6) Volume Adjustment                         (6) Packet Loss Compensation
(7) Speed dial key                            (7) DTMF
(8) Support Message Waiting Indication           A. In-Band DTMF
(9) Support phone book                           B. Out-Band DTMF RFC 2833
(10) Line Hunting                                C. SIP Info
(11) Hotline (IP,PSTN)                        (8) Tone generation and detection
(12) Provide Dial tone and downloadable          A. Ring Tone
     file for voice prompt                       B. Ring Back Tone
(13) Provide IVR for 2-Stage Dialing             C. Dial Tone
(14) One-Stage Dialing                           D. Busy Tone
Firmware And Configuration Update                E. Programming Tone
(1) Web Browser                                  F. PSTN, FAX and Modem Tones
(2) Telnet                                       G. ROH Tone
(3) TFTP/FTP
                                                                Order Information
(4) RS-232                                       Model                     Interface
                                                WG-2504      4 FXS + 1 PSTN
                                                WG-2540      4 FXO
                                                WG-2522      2 FXS + 2 FXO
          WellGate 2600 Series
          8FXS+PSTN, 8FXO, 4FXS+4FXO

          The WellGate 2600 series are 8-port FXS/FXO voice, fax, and modem over IP
GATEWAY



          gateway, which include three models. WG-2608 built-in 8 FXS ports and one
          PSTN backup line; WG-2680 built-in 8 FXO ports; WG-2644 built-in 4 FXS and
          4 FXO ports. When network fails, WG-2608 can pass call to PSTN port
          automatically. WG-2600 series provide ideal solution for providing low cost
          communications between headquarters and branch offices in the word.


          Benefits
          (1) Easy access to PSTN from IP

          (2) Cost saving - save long and international
                                                                        6
             call expense by using IP network.

          (3) No extra investment - just take advantage of

             existing Ethernet LAN and IP environment.

          (4) Easy interface with Cable Modem, ADSL Modem

             and Leased Line equipment

          (5) Easy to integrate with all kinds of IP-PBXs

          Physical interface
          (1) RJ-45 (10/100 Base-T)

             A. WAN x 1 for connecting to HUB or ATU-R directly

             B. LAN x 1 for PC or other devices connection

          (2) RJ-11

             A. FXS x 8 / FXO x 8 / FXS x 4, FXO x 4

             B. PSTN x 1

          (3) Panel Status Indications:

             FXS / FXO, Ethernet, SIP System, PSTN

          Network and Protocol
          (1) SIP v2(RFC 3261)

             A. Outbound proxy

             B. Support backup proxy registration

             C. Support IP or domain name for primary and secondary

                proxy address and auto switching is enabled.

             D. Provide SIP URI Format,support both number and text.

          (2) TFTP Client/DHCP Client/PPPoE Client

          (3) Telnet/HTTP Server

          (4) Support T.38 FAX

          (5) DNS Client
                                                                                       GATEWAY
(8) Network support Static IP/ DHCP/ PPPoE    Environment
(9) NAT , DHCP server                         (1) AC Power
(10) Support DSCP for voice packet               100~240V, 12Vdc output
     quality control                          (2) Environment
                                                                    o     o
(11) Security                                    A. Temperature 0 C~40 C
    A. HTTP 1.1 basic/digest authentication      B. Humidity 10%~90%RH
        for Web setup                         Voice feature
     B. MD5 for SIP authentication            (1) Voice codec
       (RFC2069/RFC2617)                         A. G.711(A/μ-law): 64k bit/s (PCM)
(12) Support Session Timer(RFC 4028)             B. G.723.1: 6.3k/5.3k bit/s
Call function                                    C. G.729A: 8k bit/s (CS-ACELP)
(1) Call Hold                                    D. G.729B: adds VAD & CNG to G.729
(2) Call Waiting                              (2) VAD : Voice activity detection
(3) Call Forward                              (3) CNG : Comfortable noise generation
(4) Caller ID(FSK, DTMF)                      (4) G.165/168 Echo canceller
(5) Transfer                                  (5) Configurable Jitter Buffer
(6) Volume Adjustment                         (6) Packet Loss Compensation
(7) Speed dial key                            (7) DTMF
(8) Support Message Waiting Indication           A. In-Band DTMF
(9) Support phone book                           B. Out-Band DTMF RFC 2833
(10) Line Hunting                                C. SIP Info
(11) Hotline (IP,PSTN)                        (8) Tone generation and detection
(12) Provide Dial tone and downloadable          A. Ring Tone
     file for voice prompt                       B. Ring Back Tone
(13) Provide IVR for 2-Stage Dialing             C. Dial Tone
(14) One-Stage Dialing                           D. Busy Tone
Firmware And Configuration Update                E. Programming Tone
(1) Web Browser                                  F. PSTN, FAX and Modem Tones
(2) Telnet                                       G. ROH Tone
(3) TFTP/FTP
                                                                Order Information
(4) RS-232                                       Model                     Interface
                                                WG-2608      8 FXS + 1 PSTN
                                                WG-2680      8 FXO
                                                WG-2644      4 FXS + 4 FXO
           IP Phone LP-201
           Dual Mode IP Phone Solution
IP PHONE


           Benefits
           ●   Provide IP call or PSTN call selection
           ●   To be a Plain old Telephone set during external
               power failure
           ●   Easy to integrate with all kinds of IP-PBXs
           ●   Easy access to Internet phone call
           ●   Cost Saving - Telephone call from VPN or public
               Internet
           ●   Follows the existing telephone call dial plan
           ●   Easy interface to ADSL/Cable Modem or Leased
               line equipment
           ●   Built-in Switch/Hub of two 10/100 Base-T ports
           ●   Built-in analog line telephone function

           Specification

                                                         H.323 Version                         SIP version

                   WAN Interface          ●   RJ-45 10/100 Base-T               ●   RJ-45 10/100 Base-T

                   LAN Interface          ●   RJ-45 10/100 Base-T               ●   RJ-45 10/100 Base-T

                   PSTN Interface         ●   RJ-11                             ●   RJ-11

                   Power Adapter          ●   9V DC                             ●   9V DC

                    LCD Display           ●   24 character x 2 lines            ●   24 character x 2 lines

                    Memory Key            ●   10                                ●   10

                    Function Key          ●   8                                 ●   8

                 Menu Control Key         ●   4                                 ●   4

                                          ●   5 LED (Speaker, Message, PSTN,    ●   5 LED (Speaker, Message, PSTN,
                  Button Indicators
                                              Mute, Hold)                           Mute, Hold)

                   Speaker Phone          ●   With Echo cancellation            ●   With Echo cancellation

                  Network Support         ●   Fix IP, DHCP, PPPoE               ●   Fix IP, DHCP, PPPoE

                  Operation Mode          ●   Gatekeeper or Peer-to-Peer mode   ●   Proxy or Peer-to-Peer mode

                          DDNS            ●   DDNS Support

                 Configuration and        ●   TELNET, Web Brouser, and LCD      ●   TELNET, Web Brouser, and LCD
                   Management                 Panel                                 Panel
                      Protocol            ●   H323 (V4)                         ●   SIP (RFC3261)

                                          ●   G.711U, G.711A, G.723.1, G.729,   ●   G.711U, G.711A, G.723.1, G.729,
                Audio Codec Support
                                              G.729A, G.729B, G.729AB               G.729A

                                          ●   DiffServ                          ●   DiffServ
                    Qos Support
                                                                                                                       IP PHONE
Application
●   ISP/ITSP (Internet Telephony

    Service Provider)                                                                           PC

●   IP-PBX with office telephony
                                                       ADSL or
                                                  PC                    Internet or
    services                                           Cable Modem
                                                                        Intranet
                                                       Home                                     SOHO
●   Multi-nation enterprise

    communication                                                           Router
                                                                   IP-PBX
●   SOHO Telephony
                                                                            IP Phone



                                                                     Enterprise




                              ●   Transfer, Mute, Forward, Hold,            ●   Transfer, Mute, Forward, Hold,
    Telephony Features
                                  Redial                                        Redial

     Security Support         ●   H.235 Token Password                      ●   MD5 Authentication

                              ●   Support Fast Start and H.245
       H.245 Support
                                  Tunneling

                              ●   LCD Menu, Web Browser, Telnet,            ●   LCD Menu, Web Browser, Telnet,

       Management                 TFTP/FTP, Welltech Provision                  TFTP/FTP, Welltech Provision

                                  Server PRO-9510                               Server PRO-9510

                              ●   In-band, H245 Alphanumeric, H.245         ●   In-band, RFC 2833, SIP INFO
           DTMF
                                  Signaling, Q.931, RFC 2833
 power Failure Support        ●   PSTN Line Backup                          ●   PSTN Line Backup
       Function
                              ●   CE                                        ●   CE

        Certification         ●   FCC                                       ●   FCC

                              ●   3C                                        ●   3C

                              ●   (10 to 90%) non-condensing,               ●   (10 to 90%) non-condensing,
         Humidity
                                  operating, and non-operating/storage          operating, and non-operating/storage

Operational Temperature       ●   (5 to 40°C)                               ●   (5 to 40°C)

    Storage Temperature       ●   (-20 to 65°C)                             ●   (-20 to 65°C)

        Dimension             ●   215mm(W) x 71mm(H) x 198mm(D)             ●   215mm(W) x 71mm(H) x 198mm(D)

          Weight              ●   834 g                                     ●   834 g
           IP Phone LP-305/305A
           Key Features
               Multi-server registration: Can register
IP PHONE



           ●

               up to 5 different servers
           ●   3-way Conference Bridge built-in
           ●   Headset Jack support
           ●   Full Duplex Speaker Phone
           Application
           ●   ISP/ITSP (Internet Telephony Service Provider)
           ●   IP-PBX with office telephony services
           ●   Multi-nation enterprise communication
           ●   SOHO Telephony
           Calling Features
           ●   Call Hold
           ●   Call Transfer
           ●   Call Forward
           ●   Call Waiting                                               Dimension: 215mm(W) x 71mm(H) x 198mm(D)
           ●   Call Conference                                            We i g h t ( u n i t ) : 8 3 4 g r a m s

           ●   DND (Do Not Disturb)
           ●   Call List (Missed,Received,Dialed)
           ●   5 configurable speed dials                                 Certification
           ●   Support up to 5 multi-lines                                ●   CE, FCC
           Network Support                                                Power Supply
           ●   Fixed IP
                                                                          ●   DC 9V with Adaption 110V/230V AC Input
           ●   Dynamic Host Configuration Protocol (DHCP)
           ●   PPPoE connection (When PPPoE disconnect,
                                                                          Operation Environment
               SIP-Phone can automatically re-connect)                    ●   Humidity: 10 to 90 % (Non-condensing)
                                                                                                                       o
           ●   Behind NAT IP Sharing Device                               ●   Operational Temperature: 0 to +40 C
           ●   Support QOS by setting DSCP                                Storage Environment
               (Differentiated Service Code Point)
                                                                          ●   Humidity: 10 to 90 % (Non-condensing)
               parameters of VoIP packet                                                                           o
           ●   Support STUN
                                                                          ●   Storage Temperature: -10 to +50 C
           Technical Specification
           ●   Two 10/100 Base-T Ethernets port with Switch inside                      Order Information
           ●   LCD Configuration Password Protection
           ●   2*24 dot matrix LCD display                                    IP Phone LP-305              Standard Version
           ●   LCD Display Time, Date, Caller ID, Call Duration               IP Phone LP-305A             Support PoE
           ●   9 one-touch function keys: Speaker, Redial, Mute, Hold,
               Transfer, Message, SPEED, Conference, Headset.
           ●   5 LED Display: Headset, Message, Hold, Mute, Speaker
           ●   Support Power over Ethernet (IEEE 802.3af)
           Audio Features
           ●   G.711 A/μ-Law, G.729, G.723                               PC
                                                                                                                       PC
           ●   Dynamic Jitter Buffer
           Management Features                                                  ADSL or
                                                                                                Internet or
           ●   Software Upgrade: TFTP/FTP download                              Cable Modem
                                                                                                Intranet
                                                                                                                   SOHO
                                                                                Home
           ●   Easy ways for system configuration
               A. LCD Front Panel                                                                   Router
                                                                                          IP-PBX
               B. Web Browser
                                                                                                   IP Phone
               C. TELNET
               D. Provision Server
                                                                                              Enterprise
IP Phone LP-399
Multi-Lines IP Phone Solution




                                                                                                            IP PHONE
Application                                                 Management Features
●   ISP/ITSP (Internet Telephony Service Provider)          ●   Software Upgrade: TFTP download
●   IP-PBX with office telephony services                   ●   Easy ways for system configuration
●   Multi-nation enterprise communication                       A. LCD Front Panel
●   SOHO Telephony                                              B. Web Browser
Calling Features                                                C. TELNET
●   Call Hold                                                   D. Provision Server
●   Call Transfer
                                                            Certification
●   Call Forward
●   Call Waiting                                            ●   CE, FCC
●   Call Conference (3-Way)                                 Power Supply
●   DND (Do Not Disturb)
                                                            ●   DC12V output adaptor, AC 100-240Vac input
●   Call List (Missed,Received,Dialed)
●   4 configurable speed dials                              Operation Environment
●   Support up to 3 multi-lines                             ●   Humidity: 10 to 90 % (Non-condensing)
                                                                                                    o
Network Support                                             ●   Operational Temperature: 0 to +40 C
●   Fixed IP                                                Storage Environment
●   Dynamic Host Configuration Protocol (DHCP)              ●   Humidity: 10 to 90 % (Non-condensing)
                                                                                                o
●   PPPoE connection                                        ●   Storage Temperature: -10 to +50 C
●   Support ToS
●   Support NAT function
●   Support STUN Function

Technical Specification
●   Two 10/100 Base-T Ethernet ports with Switch

    or NAT inside
●   LCD Configuration Password Protection
●   2x16 character LCD display
●   LCD Display Time, Date, Caller ID, Call Duration

Audio Features
●   G.711 A/μ-Law, G.729, G.726
●   Provide In-band and Out-band DTMF
    generation/detection                                                                       PC

Provisioning and Configuration
                                                                ADSL or
●   SIP (RFC3261) compliance                           PC       Cable Modem
                                                                                Internet or
                                                                                Intranet
                                                                                               SOHO
●   LCD configuration password protection                       Home

●   Provide adjustable Ringer , Speaker and                                         Router
                                                                          IP-PBX
    Handset volume                                                                 LAN Phone

●   Support DNS server inquiry
                                                                              Enterprise
           IP Phone LP-388
           Smart Multi-Calls IP Phone Solution
           Welltech IP Phone 388 is a full-featured IP
IP PHONE



           desktop telephone set. For ease-of-use
           functionality and ergonomic design, home user
           or enterprise can integrate this state-of-the-art
           phone into daily life. With two line buttons, user
           may switch and receive different calls more
           handily. Welltech IP Phone 388 Built-in 2
           Ethernet ports allows user to connect PC or
           network device directly through IP phone
           without extra Hub. Based on innovative
           technology, Welltech IP Phone 388 has
           many user-friendly feature buttons, including
           Conference, Call Pick Up, Transfer, Redial,
           Hold, …etc. With high-quality speakerphone,
           user can enjoy VoIP more conveniently. It is
           no doubt IP Phone 388 will be the best
           choice for everyone with VoIP.

           Application                                                 Audio Features
           ●   ISP/ITSP (Internet Telephony Service Provider)          ●   Acoustic Echo Cancellation
           ●   IP-PBX with office telephony services                   ●   Codec support G.711A/μ , G.729 ,
           ●   Multi-nation enterprise communication                       G.729B, G.723
           ●   SOHO Telephony
                                                                       ●   Jitter buffer control
           Telephony Features
           ●   Support up to 2 multi-lines
                                                                       ●   DMTF :In-band,RFC 2833,SIP INFO
           ●   Support Full Duplex Speaker Phone                       Certification
           ●   DND (Do Not Disturb)                                    ●   CE, FCC
           ●   Call waiting                                            Power Supply
           ●   Call Conference (3-way)                                 ●   Input: AC 110 – 240v
           ●   Call hold / retrieve                                    ●   Output: DC 9V/1A
           ●   Call forward (Busy, No-Answer and Unconditional)
           ●   Call transfer (attended / unattended )
                                                                       Operation Environment
           ●   Call pickup
                                                                       ●   Humidity: 10 to 90 % (Non-condensing)
           ●   MWI                                                     ●   Operational Temperature: 0 to +40°C
           ●   Call List (Missed, received, dialed numbers)            Storage Environment
           Network Support                                             ●   Humidity: 10 to 90 % (Non-condensing)
           ●   Fixed IP                                                ●   Storage Temperature: -10 to +50° C
           ●   Dynamic Host Configuration Protocol (DHCP)
           ●   PPPoE connection (When PPPoE disconnect,
               SIP-Phone can automatically re-connect)
           ●   Support QOS by setting DSCP (Differentiated
               Service Code Point) parameters of VoIP packet
           Technical Specification                                                                         PC
           ●   Two 10/100 Base-T RJ-45 Ethernet port with
               Switch inside                                               ADSL or
                                                                  PC                       Internet or
           ●   LCD Configuration Password Protection                       Cable Modem
                                                                                           Intranet
           ●   2*16 character LCD display                                  Home                            SOHO
           ●   LCD Display Time, Date, Caller ID, Call
               Duration                                                                        Router
           Management Features                                                        IP-PBX
                                                                                               LAN Phone
           ●   Software Upgrade: TFTP/FTP download
           ●   Easy ways for system configuration
                 LCD Front Panel
                                                                                         Enterprise
                 Web Browser
                 TELNET
ATA-171/172/171P/171M                                                                             NEW
Dazzling Gateway series meet everyone’s demand




                                                                                                                             GATEWAY
The ATA-17x series contain four models of gateway products: ATA-171,
ATA-172, ATA-171P, ATA-171M. With outstanding design and dazzling
appearance, Welltech ATA-17x series can satisfy all users and meet their
different requirements. ATA-171/172 is one/two-port analog telephone adapter,
and user can connect with one/two analog phone set to enjoy VoIP application.
ATA-171P is one-port analog telephone adapter plus one PSTN backup lifeline,
which allows user to dial and receive PSTN/VoIP call in one identical phone set.




Physical interface
● RJ-45
  A. WAN X 1 for connecting to HUB or
  ATU-R directly
  B. LAN X 1 for PC connection
● RJ-11

  A. Phone X 1 for ATA-171
  B. Phone X 2 for ATA-172
  C. Phone X 1, Line X1 for ATA-171P/
                                                                                      ATA-171/172/171P/171M
     ATA-171M
● Dimension: 9.9 X 9.9 X 3.2 cm

                                                                          Order Information
Network and Protocol                                 Model              Interface Specification                    Color
●   SIP v1(RFC2543), v2(RFC 3261)                   ATA-171    One Port FXS                                 Champagne Gold
    A. Outbound proxy                               ATA-172    Two Port FXS                                 Pearl White
    B. Support backup proxy registration            ATA-171P   One Port FXS + One PSTN backup line Port     Metal Blue
    C. Support IP or domain name for                ATA-171M   One FXS + One FXO                            Metal Purple
       primary and secondary proxy address
       and auto switching is enabled.
●   IP/ICMP/ARP/RARP/SNTP
                                               Voice feature                              Firmware and
●   TFTP Client/DHCP Client/PPPoE Client
                                               ● Voice codec                              configuration update
●   Telnet/HTTP Server
●   DHCP Server For LAN Port                     A. G.711: 64k bit/s (PCM)             ● Web Browser

●   NAT transversal                              B. G.726: 16k/24k/32l/40k bit/s       ● Telnet

    A. STUN                                         (ADPCM)                            ● Voice configuration

●   Support ToS                                  C. G.729A: 8k bit/s (CS-ACELP)        ● TFTP

    Security                                                                           ● HTTP
●                                                D. G.729B: adds VAD & CNG to
     A. HTTP 1.1 basic/digest authentication        G.729                              Environment
        for Web setup                          ● VAD : Voice activity detection        ● AC Power
     B. MD5 for SIP authentication             ● CNG : Comfortable noise generation
                                                                                          100~240V Input 12V dc/output
        (RFC2069/RFC2617)                      ● LEC : Line echo canceller             ● Environment
                                               ● Packet Loss Compensation                                       o     o
                                                                                           A. Temprature :0 C~40 C
Call function                                  ● DTMF                                      B. Hmidity:10%~90%RH
●   Call Hold                                    A. In-Band DTMF
●   Call Waiting                                 B. Out-Band DTMF
●   Call Forward
                                                 C. SIP Info
●   Caller ID                                                                                          ADSL Modem
                                               ● Tone generation
●   Flash                                                                                      WAN                  Internet
                                                 A. Ring Tone
●   Volume Adjustment
●   Speed dial key                               B.Ring Back Tone
●   Phone book                                   C.Dial Tone                             LAN

●   Call Transfer between FXS, FXO and IP        D.Busy Tone                     Phone 2(ATA-172 only)      Phone 1

    port (ATA-171M only)                         E.Programming Tone
                                                                                     PSTN Backup (ATA-171P only)
●   Call Forwarding between FXS, FXO and                                     PSTN
    IP port (ATA-171M only)                                                                         ATA-171M
●   Support T.38 Fax
          ATA-175/175P
          FXS+IP Sharing+PSTN Backup

          The ATA-175/175P is a two-port FXS voice, fax and modem over IP gateway,
          which provides user convenient telephony and data service. When network
GATEWAY



          fails, with one port of PSTN backup, ATA-175P can pass calls to PSTN port
          automatically. ATA-175/175P built-in one WAN port and 4 LAN ports, user can
          integrate it with original network environment without extra switch or hub.
          ATA-175/175P’s IP Sharing function allows other computers or devices connect
          with network more easily.

          Benefits
          (1) Easy access to PSTN from IP

          (2) Cost saving - save long and international

             call expense by using IP network.

          (3) No extra investment - just take advantage of

             existing Ethernet LAN and IP environment.

          (4) Easy interface with Cable Modem, ADSL Modem

             and Leased Line equipment

          (5) Easy to integrate with all kinds of IP-PBXs

          Physical interface
          (1) RJ-45 (10/100 Base-T)

             A. WAN X 1 for connecting to HUB or ATU-R directly

             B. LAN X 4 for PC or other devices connection

          (2) RJ-11

             A. Phone X 2

             B. PSTN X 1 (ATA-175P )

          (3) Panel Status Indications:

             FXS, Ethernet, SIP System, PSTN (ATA-175P)

          Network and Protocol
          ( 1) SIP v2(RFC 3261)
             A. Outbound proxy

             B. Support backup proxy registration

             C. Support IP or domain name for primary and secondary

                proxy address and auto switching is enabled.

             D. Provide SIP URI Format,support both number and text.

          (2) IP/ICMP/ARP/RARP/SNTP

          (3) RTP - IETF RFC1889, RTCP - IETF RFC1890

          (4) TFTP Client/DHCP Client/PPPoE Client for WAN Port

          (5) Telnet/HTTP Server

          (6) DNS Client

          (7) Network support Static IP/ DHCP/ PPPoE
(8) NAT , DHCP Server for LAN Port             Voice feature




                                                                                                                                GATEWAY
(9) STUN                                       (1) Voice codec
(10) Support T.38 FAX                               A. G.711(A/μ-law): 64k bit/s (PCM)
(11) Support ToS for voice packet                   B. G.726: 16k/24k/32k/40k bit/s (ADPCM)
    quality control                                 C. G.729A: 8k bit/s (CS-ACELP)
(12) Security                                       D. G.729B: adds VAD & CNG to G.729
    A. HTTP 1.1 basic/digest                   (2) VAD : Voice activity detection
       authentication for Web setup            (3) CNG : Comfortable noise generation
    B. MD5 for SIP authentication              (4) G.168 Echo canceller
       (RFC2069/RFC2617)                       (5) Packet Loss Compensation
(13) Support Session Timer(RFC 4028)           (6) DTMF
(14) Up to 95MB throughput at Bridge Mode           A. In-Band DTMF
Call function                                       B. Out-Band DTMF RFC 2833
(1) Call Hold                                       C. SIP Info
(2) Call Waiting                               (8) Tone generation and detection
(3) Call Forward                                    A. Ring Tone
(4) Caller ID(FSK, DTMF)                            B. Ring Back Tone
(5) Transfer                                        C. Dial Tone
(6) Volume Adjustment                               D. Busy Tone
(7) Speed dial key                                  E. Programming Tone
(8) Support Message Waiting Indication              F. PSTN, FAX and Modem Tones
Firmware and Configuration Update
(1) Web Browser

(2) Telnet                                                                 Order Information

(3) Voice configuration
                                                  Model                                    Interface
                                             ATA-175                 2- Port FXS
(4) TFTP
                                             ATA-175P                2- Port FXS+One PSTN
(5) HTTP

Environment                                 ● A p p l i cat i o n D i ag r am

(1) AC Power

   A. 100~240V AC Input, 12V DC Output

(2) Environment
                      o    o
   A. Temperature 0 C~40 C

   B. Humidity 10%~90%RH
                                                              ADSL Modem




                                                        Internet

                                                                                Your Computer     Analog Phone   Analog Phone
WIFI PRODUCTS
                WiFi Phone WP-508                                                          NEW
                The best Wireless Solution

                Welltech announce new WiFi Phone solution WP-508, which                                    WP-508


                complies with IEEE 802.11b/g and allows users to access VoIP
                service roaming from home and through IEEE 802.11b/g hotspots.

                With classic and elegant design, WP-508 represents noble
                quality. No doubt, WP-508 is the best choice of WiFi Phone.

                Features
                ●   Direct Sequence with data rates at                ●   DTMF Generation
                    1, 2, 5.5, 11, 6, 9, 12, 18, 24, 36, 48, 54Mbps   ●   Acoustic Cancellation
                ●   Support Site Survey --- scan available APs in     ●   Echo Cancellation: ITU-T G.168 or G.165
                    handset’s environment                             ●   Adaptive, low delay jitter buffer for voice data
                ●   Support both DHCP and static IP
                ●   Internet Protocol Support: IP, TCP, UDP, ARP,     Security
                    HTTP 1.0, DHCP, DNS                               ●   WEP 64-bit (hardware) and 128-bit (software)
                ●   Phone keypad and graphic LCD display              ●   WMM
                ●   Rechargeable Batteries – Li-ion                   ●   WPA-PSK
                                                                      ●   WPA (EAP-TLS,EAP-TTLS)
                Wireless Specification                                ●   WPA2
                ●   Wireless Protocol: IEEE 802.11b / g               ●   AP Profile
                ●   Data rate: 1, 2, 5.5 and 11 Mbps~54Mbps
                ●   Frequency band: 2.400 ~ 2.497 GHZ                 Phone Function Support
                ●   Operating range: Out-door up to 300m,             ●   Caller ID
                    In-door up to 75m                                 ●   Keypad lock/ Keypad tone
                ●   Channel: FCC Ch1 ~ 11, ETSI Ch1~13,               ●   Redial
                    Japan Ch1~14                                      ●   Mute
                ●   Output Power: 16 dBM(802.11b),18dBM(802.11g)      ●   Call forwarding
                ●   Sensitivity: -80dBM@11Mbps                        ●   Call rejection
                ●   Site Survey: Scan available APs in                ●   Handset, ringer volume adjustment
                    handset’s environment                             ●   Call Progress tone:Dial tone, DTMF generator,
                                                                          ring back tone, busy tone, and ring tone
                VoIP Protocol Support                                 ●   SMS
                ●   IETF RFC 3261 SIP Protocols                       ●   DTMF Relay RFC-2833
                ●   SDP (RFC2327)                                     ●   Time protocol support (SNTP).
                ●   RTP (RFC1889) , RTP Profile – IETF RFC1890        ●   Battery and signal level metering.
                ●   RTCP - IETF RFC1889                               ●   Phone books, call history for friendly usage.
                ●   Support outbound proxy for NAT Traversal          ●   Phone Books support for editing/deleting
                ●   STUN for NAT Pass-Through                             personal data
                ●   QoS:ToS/Diff Serv                                 ●   Standby time up to 80 hours, talking time
                ●   RFC 2833 Out-of-Band / In-band DTMF Relay             3-4 hours
                ●   Session Timer IETF RFC 4028                       ●   Speed Dial
                ●   Privacy mechanism IETF RFC 3323 / 3325            ●   Speaker Volume Control
                                                                      ●   USB Charge
                Voice                                                 ●   Missed Call Indicator
                ●   Codec: ITU-T iLBC, G.729A,G.729AB, G.711
                ●   VAD (Voice Activity Detection)                    Management
                ●   CNG (Comfort Noise Generation)                    ●   Phone Configuration through handset LCD
                ●   Arbitrary Tone Generation                             and web
                                                                      ●   TFTP Firmware Update
                                                                      ●   HTTP Firmware Update
Wi-Fi VoIP Gateway WellGate 3512




                                                                                                         WIFI PRODUCTS
WiFi + 2 FXS + 1 PSTN
WellGate 3512 is a two-port FXS + one PSTN wireless gateway, which supports
RFC3261 SIP protocol. Telephone will switch to PSTN port automatically under
power failure. User can also select to dial out through PSTN line manually.
WG-3512 complies with wireless protocol 802.11 b/g, and can operate as Access
Point or Client. Built-in four LAN ports and NAT function allows other devices
to access network more easily.

Benefits
●   Easy access to IP from phone set or PBX
●   Cost Saving - Telephone call from VPN or
    public Internet
●   Follows the existing telephone call dial plan
●   Easy interface to ADSL/Cable Modem or Leased        ●   Bandwidth Control
    line equipment
                                                        ●   PPPoE/PPTP client
                                                        ●   UPnP IGD
●   Easy to integrate with all kinds of IP-PBXs
                                                        ●   STUN
                                                        ●   DDNS and NTP client
Physical interface
                                                        ●   QOS - DSCP class0-7 and EF
●   RJ-45                                               ●   URL Filtering and DoS (Deny of Service)
      WAN X 1 for connecting to HUB or ATU-R directly   ●   DNS relay
      LAN X 4 for PC or other devices
●   RJ-11                                               Telephony Features
      Phone X 2 for regular phone connection            ● Caller ID: TypeI/II DTMF, FSK

      PSTN X 1 for Dialing and Receiving PSTN call.     ● Flash Hook Timer Configuration

●   Power: Input AC 100V~240V Output DC12V              ● Gain Adjustments

●   LED Indicator: Power, WLAN, WAN, LAN, FXS, PSTN     ● PSTN Bypass: Power Failure, Manually

                                                        ● Call Waiting

                                                        ● Call Hold
Voice Feature
                                                        ● Blind Transfer
●   Codec: G.711u/A-Law, G.729A , G.726, GSM-FR
                                                        ● Call Forward
●   VAD/CNG
                                                        ● 3-way Conference
●   Adaptive Jitter Buffer
                                                        ● 10 Speed Dials
●   Line Echo Cancellor
                                                        ● T.38 FAX
●   FAX/Modem tone detection and pass through
●   DTMF: Inband, RFC-2833, SIP Info
                                                        Dimension
                                                        ● 17.5 x 12.5 x 3.2 cm
Network
●   802.11 b/g Access Point, WiFi compliant             Protocol
      802.1x, WEP, WPA TKIP and WPA2 AES/Mixed          ● SIP RFC3261
      mode for PSK and TLS (Radius)                     ● TCP/UDP/IP/ICMP/ARP

      802.11f (IAPP)
      Wireless Auto-channel selection                   Management
      Wireless access control by MAC address            ● System log

                                                        ● Display real-time information for system
      (deny or accept)
●   802.11b/g client mode                                 setting, statistics, and associated wireless
●   802.1d with spanning tree protocol                    client status.
                                                        ● User name/password authentication for web
●   NAT/NAPT
                                                          server login and logout
●   Firewall
                                                        ● Web-based configuration and management
●   Virtual DMZ
                                                          interface
●   ALG for:FTP, SIP, VPN pass-through with             ● Firmware update through web
    multiple sessions (IPSEC, L2TP, PPTP)               ● Configuration backup/restore to/from a file.
●   DHCP client and server                                Reset configuration to factory default
●   Up to 98MB throughput at Bridge Mode                ● IVR-Configuration by phone keys with
●   Qos: 802.1Q (VLAN)                                    Interactive Voice Prompts
USB PERIPHERAL
                 Welltech K-1030
                 Amazing Skype™ Compatible
                 USB Phone Solution




                 World famous Skype TM makes millions of users enjoy free long-distance call,
                 however, user has to use headset,microphone and make all control via PC.

                 Through newest Welltech K-1030 USB Phone, user can use Skype TM without
                 changing general habit and it become so convenient even for children and elder
                 people.With built-in LCD, on K-1030 can check Skype TM contact list and
                 on/off–line status, user can dial out or answer Skype TM call directly without
                 handling computer anymore.

                 With tiny and lightweight design, travelers can bring K-1030 anywhere and
                 integrate it with daily life easily. No doubt,Welltech K-1030 will be the best USB
                 Phone solution for everybody.

                 Hardware features
                               Item                                            Content
                 ●   Interface               ●   USB1.1: 12Mbps
                 ●   Device Class            ●   USB complex Device(Audio Class & HID Class)
                 ●   Transmission            ●   Supports control.Interrupt.Bulk.Isochroous data
                 ●   Operation System        ●   Windows 98 Second Edition.Windows 2000.Windows Me
                                                 Windows XP
                 ●   Cabling                 ●   USB A-type connector
                 ●   Cable length            ●   1.5M(Tolerance ±10%)
                 ●   Operating Voltage       ●   DC 4.5 ~5.5V
                                                     o
                 ●   Operating Temperature   ●   0~50 C
                                                           o
                 ●   Storage Temperature     ●   -10 to +55 C
                 ●   Storage Humidity        ●   10 to 90 % (Non-condensing)
                 ●   Operating Humidity      ●   20% ~ 80% RH
                 ●   Certification           ●   CE, FCC
                 ●   Dimension               ●   122 x 46 x 17 (mm)
                 ●   LCD                     ●   29.00 x 19.40 x 2.20
                                                                                                        USB PERIPHERAL
Features
●   Speaker
●   Microphone
●   Numeric Buttons 0,1,2,3,4,5,6,7,9, *, #
●   Green button (Answer/Dial)
●   Red Button (Cancel/Hang up)
●   Menu Up, Menu Down
●   Button “M”(Menu), Button “C”(Cancel)
                                                      K-1030A   K-1030B   K-1030C   K-1030D   K-1030E
●   Volume Decrease/Soft Button 1, Volume
    Increase/Soft Button 2


Soft Phone Support
●   Skype
●   Welltech Soft Phone


Special Features
●   Display contact and Call list
    on LCD with data
●   Lightweight and size
●   Adjustable Ring Melody


Certifications
●   CC, FCC (Proceeding)


Dimension
●   122mm(W) x 46mm(H) x 17mm(D)




                                    ● USB phone
                                                            IP Network
                                    Application Diagram




                                              USB
                                                                            PC/Windows
                                              cable
                                                                            Based softphone
USB PERIPHERAL
                 USB Phone K-1000/1010
                 You can use Skype™ with Welltech USB Phone now


                 The K-1000/K-1010 is a phone with USB interface, just plug and play. No drive
                 needed. It’s streamline and fashion design, like a mobile phone. The K-1000/1010
                 provided H.323/SIP Internet telephony function for different softphone software
                 providers. It is easy to operate and configure for Home and Office / SOHO
                 users.




                                                      K-1000                                 K-1010


                 Benefits                                                       Specification
                 ●   High Mobility                                              ●    Physical Connectors
                 ●   Provided real-time volume adjustable button                     One USB port
                 ●   Can be mounted on monitor or wall(K-1000)                  ●    Cabling
                 ●   Provided WAV voice file for different ring type                 USB A-type connector
                 ●   Keypad mapping program for                                 ●    Standard support

                     different soft phone software:                                  Conform to USB 12 Mbps Spec.
                                        TM
                     NetMeeting,Skype        ,X-ten,MSN,SJphone                      Version 1.1
                 ●   Support Windows HID driver                                      Conform to USB Audio Device
                 ●   Redial key for K-1010 only                                      Release 1.0
                 ●   Volume control key                                         ●    Operating Voltage
                 ●   Free PC-to-PC calls over the Internet                           4.5V ~ 5.25V
                 ●   Provided API for softphone software developer              ●    Operating Temperature

                 System Requirement                                                  0 ~ 50 °C
                 ●   Processor                                                  ●    Operating Humidity

                     Intel 486 DX 66 or above                                        20% ~ 80% RH
                 ●   Storage                                                    ●    Certification

                     32MB RAM                                                        CE, FCC

                     10MB Hard Disk
                                                                       ● USB phone
                 ●   Internet service                                  Application Diagram           IP Network

                     Dial up, ISDN, lease line, xDSL or LAN
                 ●   Operating System

                     Microsoft Windows 98se/ME/2000/XP                              USB
                                                                                                                  PC/Windows
                                                                                    cable
                                                                                                                  Based softphone

				
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