Welltech WG5290 User Manual

Welltech Computer Co., Ltd.
WellGate 5290
Universal VoIP Gateway
User Guide
Version 6.1
WellGate 5290 User Guide – V6.1 - 1 -
Welltech Computer Co., Ltd.
CONTENTS
CHAPTER 1 WELLGATE 5290 INTRODUCTION ..........................................................................4
SYSTEM DESCRIPTION .......................................................................................................................... 4
TECHNICAL SPECIFICATION ..................................................................................................................5
AUDIO FEATURE ...................................................................................................................................5
WELLGATE 5290 V6.1 DETAIL SPEC....................................................................................................7
WELLGATE 5290 APPEARANCE DESCRIPTION....................................................................................13
CHAPTER 2 LOGON WELLGATE 5290 ......................................................................................... 15
LOGON WELLGATE 5290....................................................................................................................15
NETWORK CONFIGURATION................................................................................................................16
SYSTEM TIME CONFIGURATION ..........................................................................................................18
ACCOUNT MANAGER..........................................................................................................................19
RELOGIN.............................................................................................................................................21
CHAPTER 3 H.323 GATEKEEPER AND SIP PROXY MODE CONFIGURATION .................. 22
ENVIRONMENT USED IN THIS CHAPTER ...............................................................................................22
INTERFACE CONFIGURATION ..............................................................................................................22
T1/E1 TRUNK CONFIGURATION ..........................................................................................................24
SS7 CONFIGURATION..........................................................................................................................26
H.323 CONFIGURATION ......................................................................................................................27
SIP CONFIGURATION ..........................................................................................................................28
DIGIT MANIPULATION ........................................................................................................................30
CHAPTER 4 CALL FLOW EDITOR................................................................................................34
CONFIG MANAGER.............................................................................................................................46
APPLY CHANGE ..................................................................................................................................48
CHAPTER 5 PEER TO PEER MODE CONFIGURATION ........................................................... 49
ENVIRONMENT USED IN THIS CHAPTER ...............................................................................................49
NETWORK CONFIGURATION................................................................................................................49
ACCOUNT MANAGER..........................................................................................................................49
INTERFACE CONFIGURATION ..............................................................................................................49
H.323 CONFIGURATION ......................................................................................................................49
VOIP HUNTING..................................................................................................................................50
DIGIT MANIPULATION ........................................................................................................................52
CALL FLOW EDITOR ...........................................................................................................................52
CONFIGURATION MANAGER ...............................................................................................................53
APPLY CHANGE ..................................................................................................................................53
CHAPTER 6 SIP TO H.323 MODE CONFIGURATION ................................................................54
ENVIRONMENT USED IN THIS CHAPTER ...............................................................................................54
NETWORK CONFIGURATION................................................................................................................54
ACCOUNT MANAGER..........................................................................................................................54
INTERFACE CONFIGURATION ..............................................................................................................54
H.323 CONFIGURATION ......................................................................................................................54
SIP CONFIGURATION ..........................................................................................................................54
ADDRESS BOOK..................................................................................................................................54
DIGIT MANIPULATION ........................................................................................................................55
CALL FLOW EDITOR ...........................................................................................................................55
CONFIGURATION MANAGER ...............................................................................................................55
APPLY CHANGE ..................................................................................................................................55
CHAPTER 7 ADVANCE CONFIGURATION REFERENCE.........................................................56
CONFIGURATION.................................................................................................................................56
System Configuration....................................................................................................................56
Interface Configuration ................................................................................................................57
Reset Configuration......................................................................................................................67
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H.323 Configuration.....................................................................................................................68
SIP Configuration.........................................................................................................................71
SS7 Configuration.........................................................................................................................75
Access Control..............................................................................................................................76
Number Replace............................................................................................................................81
Data Base Number........................................................................................................................82
Routing Plan.................................................................................................................................83
Radius Setting...............................................................................................................................84
Apply Change................................................................................................................................85
CHAPTER 8 SYSTEM CONTROL...................................................................................................87
SYSTEM ..............................................................................................................................................87
SYSTEM TIME .....................................................................................................................................87
NETWORK...........................................................................................................................................89
SNMP................................................................................................................................................89
PROMPT MANAGER ............................................................................................................................90
CALL FLOW EDITOR ...........................................................................................................................96
ACCOUNT MANAGER..........................................................................................................................96
UPGRADE ...........................................................................................................................................96
RELOGIN.............................................................................................................................................96
CHAPTER 9 SYSTEM MONITOR ...................................................................................................97
LINE SUMMARY STATUS .....................................................................................................................97
LINE DETAIL .......................................................................................................................................97
SYSTEM STATUS .................................................................................................................................98
EVENT LOG ........................................................................................................................................99
DEBUG INFO .....................................................................................................................................102
PING .................................................................................................................................................102
CHAPTER 10 TELNET & RS-232 CONFIGURATION................................................................ 103
LOGON WELLGATE 5290 BY TELNET ...............................................................................................103
CHAPTER 11 LCD DISPLAY CONFIGURATION ....................................................................... 110
APPENDIX 1 CALL FLOW EXAMPLE......................................................................................... 113
ONE STAGE DIALING (GATEKEEPER MODE).....................................................................................113
ONE STAGE DIALING (SIP PROXY MODE) ........................................................................................115
ONE STAGE DIALING (PEER TO PEER MODE)....................................................................................117
TWO STAGE DIALING (VOIP, PSTN MIXED CALL) ............................................................................119
TWO STAGE DIALING WITH AAA (IP SIDE AAA).............................................................................122
APPENDIX 2 JAVA PLUG-IN INSTALLATION ...........................................................................126
APPENDIX 3 RETRIEVE CDR INFORMATION.........................................................................127
APPENDIX 4 INTERFACE LED DESCRIPTION.........................................................................129
APPENDIX 5 BUILD-IN VOICE PROMPT INDEX .....................................................................130
APPENDIX 6 TIME ZONE TO COUNTRY MAPPING LIST .....................................................131
APPENDIX 7 IP BANDWIDTH REQUIREMENT........................................................................ 133
APPENDIX 8 RELEASE COMPLETE CAUSE CODE ................................................................134
APPENDIX 9 RADIUS FORMAT ATTRIBUTES..........................................................................137
APPENDIX 10 QUICK START CHECK LIST...............................................................................141
APPENDIX 11 SS7 REFERENCE DOCUMENTS......................................................................... 145
WELLGATE 5290 FAQ..................................................................................................................... 145
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Welltech Computer Co., Ltd.
Chapter 1 WellGate 5290 Introduction

System Description

A Universal VOIP/SS7 Gateway – Enabling SS7 to VoIP Network
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, QSIG, SS7/ISUP Support with Voice or Signal only SS7 Link 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 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 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
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Technical Specification

Interface
- Two 10/100MB Ethernet Ports (Host & VoIP stream)
- E1 SS7 Interface
- Up to 4 T1/E1 (120 Ohm-RJ48C connectors)
75 Ohm needs external 3rd party BNC/RJ-48C adapter cables

Protocol and Standard

- ITUT H.323 v5 compliance
- SIP RFC 2543/3261 compliance

Audio Feature

- Codec -- G.711 A/μ-Law, G.723.1 (5.3K/6.3K), G.729A, GSM
- Support G.168 echo cancellation (128 ms)
- Configurable audio payload size & adaptive jitter buffer
- Support silent suppression for G.729A, G.723
- VAD (Voice Activity Detection)
- CNG (Comfort Noise Generation)
DTMF Transmission
- Transparent
- H.245 signal/alphanumeric
- H.323 Q.931
- RFC 2833 (SIP only)
- SIP INFO
FAX Support
- Automatic voice/fax detection
- T.38 fax relay based on H.323 Annex D
- SIP T.38 fax relay
- Up to G3 fax
- ECM support
- Redundant T.38 packet (0-2)
- CISCO compatible
Built-in IVR & call-flow controller
- Web-based GUI Drag and Drop interface
- Full control of call behavior (one-stage or two-stage dialing)
- IVR functions
- Support time duration play back (Chinese & English)
- Power call information branch
- Collected information validation
- Active disconnect & reconnect without hang up
- Selected disconnect cause code & behavior
Management Feature
- OS and program upgradeable
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- Console port: RS-232 port
- TELNET
- Full Web management interface & real time monitor
- Front panel LCD
- SNMP v2 (H.341) and SNTP v4 support
- User account management
- Time zone and day light saving support
- Support fixed IP and DHCP
- Support DNS and dynamic DNS
LED indicators for system status
- Power/Storage access indicators
- Front panel LCD (2
lines
X 16) status display
Power
- 90~240V auto switch
Environmental
- Operation temp: 0° C to 60° C
- Relative humidity: 5% to 95%
Dimension
- 483mm(L) x 450 mm(W) x 44mm(H)

Certification

- CE
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Wellgate 5290 V6.1 Detail Spec.

Feature Wellgate 5290 V6.1
Physical Dimension
1 Width 2 Height 3 Depth 4 Industrial rack mount 5 Color 6 Weight
Power / Environmental
1 Power 2 Operating temperature 3 Relative humidity
Processors & Storage
1 DSP vendor
2 Operation System 3 RAM 4 Program/Data Storage 5 OS Upgradeable 7 Program Upgradeable
Front Panel Display
1 LED status 2 LCD status
LAN Interface
1 10/100 Base Ethernet
2 IP Address Required
PSTN Interface
Customizable E1/T1 CAS
1
E1 CAS DTMF
2
E1 CAS R2 MF
3
E1 ISDN PRI Support
4 5 E1/T1 Interface Selectable
PCM law Support
6
483mm 88mm 450mm Yes (2U) Black 10Kg
90-240V auto switch
0~60 C 5%~95%
Intel Pentium, Audio Codes DSP Dialogic SS7 Interface
XP Embedded 1024 MB 512 MB DOM Yes Yes
Power/DOM/System Yes
10/100MB Ethernet ports *2 (host & RTP)
Yes Loop Start FXO Hot-Line
Argentina, Bolivia, Brazil, Chile, China, Czech-Republic, Egypt, India, Indonesia, Israel, ITU, Korea, Malaysia, Mexico, Philippines, Thailand, Uruguay, Venezuela, RomTelcom
Euro, Australia, Hong Kong, Korea, New Zealand, QSIC
Alaw/Mulaw selectable
2
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E&M Bell Core Feature Group D, Wink Start, E&M Delay Start, E&M Feature Group A Immediate Start, E&M Feature Group B Wink Start, E&M Feature Group D Wink
T1 CAS DTMF/R1MF
7
T1 ISDN PRI Support
8
9 E1 SS7 Support ISUP with voice/without voice
Trunk Spans
11
Default Trunk Channel Mask
12
PSTN Line Hunting
13
PSTN Line Hunting Channel Selection
14 15 Overlapping Sending for ISDN incoming call Yes
On the Fly Reset Channel/Trunk
16
Audio Codec Support
1 G.711 A-law 2 G.711 u-law 3 G.723.1 4 G.729A
5. GSM 5 Selectable Payload Size – G.711 6 Selectable Payload Size – G.723 7 Selectable Payload Size – G.729
Fax Transmission
1 T.38 Fax Relay 2 CISCO Compatible 3 ECM Support 4 FAX auto-detection 5 H.323 Annex D Support 6 SIP- T.38 Re-invite 7 T.38 During fast connect 8 T.38 Redundant Packet 9 Transparent mode
10 Up to G3 FAX
DTMF Transmission
1 RFC 2833 2 H.245 Alphanumeric mode 3 H.245 Signal mode
4 Q.931 UUI Yes 5 SIP INFO Yes
Start(ANI B4 ADDR), E&M Feature Group D Wink Start, E&M Immediate Start, E&M Wink Start, Ground Start FXO, Ground Start FXS, Loop Start FXO, Loop Start FXS, Loop Start FXO Hot-Line NI2 ISDN,5ESS 10 ISDN,DMS100 ISDN, NTT ISDN (INS-1500), Hong Kong, QSIC
1,2,4 (T1/E1s) per chassis Yes Yes Yes
Yes
Yes Yes Yes (5.3/6.3K) Yes Yes 20, 40, 60 ms 30, 60, 90 ms 20, 40, 60 ms
Yes Yes Yes Yes Yes Yes Yes 0-2 Yes Yes (up to 14400 bps)
Yes Yes Yes
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6 Transparent mode Yes
Voice Quality & Echo Cancellation
1 Adaptive Jitter Buffer 2 CNG 3 G.168 (Echo Cancellation) 4
Gain Control Yes 5 Improved Echo Tail Suppression 6 Silence Suppression
VAD
7
Maintenance
1 Administrative Log 2 Auto Daylight Saving 3 Customizable Time Zone 4 Front Panel LCD Setup 5 FTP Server 6 HTTP server 7 HTTP SSL support 8 Multiple configuration 9 NTP time synchronization
10 Password Security 11 RS232 12 System Event Log 13 Telnet 14 Time Zone Support 15 User Account Manager 16 Web-based GUI 17 Web-based Real Time Monitor 18 Web-based Voice File Management
Network Management
1 DHCP 2 Fixed IP 3 DNS 4 Dynamic DNS 5 Ping 6 TOS field setting 7 SNMP V2 MIB I & II 8 SNMP get command 9 SNMP set command
10 SNMP Trap 12 SysLog Support 13 System Status
H.323 Protocol Support
1 H.323 V5 2 H.323 ID 3 E.164 ID
4 Fast Connect
Yes Yes Yes (128ms)
Yes Yes Yes
Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes
Yes Yes Yes Yes Yes Yes (RTP only) Yes Yes Yes Yes Yes Yes
Yes Yes Yes
Yes (selectable for incoming/outgoing)
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6 H.245 Tunneling 7 Early H.245 8 Cause Code Mapping
SIP Protocol Support
1 Cause Code Mapping 2 HTTP Digest Authentication 3 SIP Call on Hold 4 SIP Early Media/Delay Media 5 SIP Overload Redirect 6 SIP Transfer (unattend) 7 SIP Transfer (attend) 8 SIP/TCP 9 SIP/UDP
10 SIP-180/SDP 11 SIP-183/SDP 12 SIP-PRACK 13 SIP-RFC 3261 14 SIP-RFC 3264 (Offer/Answer)
H.323 Gatekeeper Support
1 Gatekeeper Register 2 Direct call 3 Routed call 4 Light weight RRQ 6 Gatekeeper Call only
SIP Proxy Sever Support
1 SIP Outbound Proxy Support 2 SIP Redirect Server Support 3 SIP Register Server Support 4 Redundant SIP Proxy Server 5 Auto Fail Over
Dial Plan
1 P2P H.323/SIP Call 2 GK Call 3 SIP Call 4 PSTN Call 5 Mixed SIP, P2P, GK call 6 Build-in Phone Book 7 P2P Prefix Routing 8 Digits Manipulation 9 ISDN Dial Plan by Prefix
10 Overlap Sending Dial Plan
Call Type Support
1 Call Decision 2 H.323 to H.323 Call 3 H.323 to H.323 Fax Relay
Yes Yes Yes
Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes
Yes Yes Yes Yes Yes
Yes Yes Yes Yes Yes
Yes Yes Yes Yes Yes Yes Yes Yes Yes (Source & Destination) Yes
Dynamic Decided by Call Flow Yes Yes
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4 H.323 to PSTN Call 5 H.323 to SIP Call 6 H.323 to SIP FAX Relay 7 H.323 to SIP FAX Relay 7 PSTN to H.323 Call 8 PSTN to PSTN Call 9 PSTN to SIP Call
10 SIP to H.323 Call 11 SIP to PSTN Call 12 SIP to SIP Call 13 SIP to SIP Fax Relay 14 VoIP to VoIP RTP unRouted 15 VoIP to VoIP RTP Routed
Enhance Service
1 ANI Access List 2 DNIS Access List 3 DID/DOD 4 PSTN Two Stage Dialing 5 VoIP Two Stage Dialing
6 Intelligent PSTN Call Routing
7 In-trunk hunting method
8 Ring Back Tone Generation 9 Call Progress Tone Support
10 Web-based Call Flow GUI
11 Play Credit Time Duration 12 Play Credit Balance 13 Almost-time-expired notify tone 14 IVR for PSTN 15 IVR for SIP 16 IVR for H.323 17 IP Access List 18 ANI Replacement 19 DNSI Replacement
AAA
1 Call detail record (CDR) 2 RADIUS Authentication 2 RADIUS Authorization
Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes
Yes Yes Yes Yes Yes
Yes (Random, Round Robin, Priority)
cyclic, random, rotary, reverse cyclic, reverse rotary
Yes (per trunk enable/disable) Yes
Drag and Drop interface, Full control of call behavior (one-stage or two-stage dialing), IVR functions, Support time duration play back (Chinese & English), Power call information branch, Collected information validation, Active disconnect & reconnect without hang up, Selected disconnect cause code & behavior
Yes (Chinese & English) Yes (Chinese & English) Yes Yes Yes Yes Yes Yes Yes
Yes Yes Yes
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3 RADIUS Accounting 4 Redundant RADIUS Server Support 5 PSTN Prepaid Support 6 VoIP Prepaid Support
Embedded AAA
1 Embedded Prepaid Service 2 Embedded Postpaid Service 3 Point/second Calculation 4 Second/point calculation 5 Auto Disable/Clean User 6 PSTN Prepaid Support 7 VoIP Prepaid Support
System Limitation
1 Max DM 2 Max IP ACL 3 Max DNIS ACL 4 Max ANI ACL 5 Max Phone Book Entries 6 Max Call Flow Component 7 Max CDR Keep Days
8 Max Voice File Storage
Manual
1 English User Guide
Yes Yes, Active/Standby/Auto Failover Yes Yes
Yes Yes Yes Yes Yes Yes Yes
4096 2048 4096 4096 10000 256 5
10 hours
Yes
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WellGate 5290 Appearance Description

WellGate 5290 Front Panel:
1
2
Functions:
1: Power LED 2: H/D LCD 3: System Status LED
4: LAN or Management Interface
5: WG5290 Service Interface 6: Power Switch 7: LCD Panel 8: LCD Touch Panel 9: USB
l
WellGate 5290 Rear Panel:
3
4
7
5
Figure 1.1-1
6
8
9
1
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2
3
12
7
Figure 1.1-2
13
11
10
4
8
5
6
9
Welltech Computer Co., Ltd.
Functions:
1: Electric Fan 2: AC Power outlet 3: AC Power switch 4: Keyboard/Mouse 5: Management Ethernet port 6: Service Ethernet port 7: VOIP Ethernet port 8: External Monitor Connector
9: Console port
10: USB (not used) 11: E1 Trunk port 12: SS7 E1 (if you are using SS7 with voice E1, you need to configure the corresponding
SS7 settings according to Dialogic document and connect 2 order to separate voice from SS7 E1 to voice E1)
nd
E1 SS7 E1 to Voice E1 port in
13: Voice E1 derived from SS7 E1. It is used only for SS7 with voice E1 and
will be connect to E1 Trunk Port (10).
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Chapter 2 Logon WellGate 5290

After connected E1/T1 & Ethernet cables into the WellGate 5290,
turned on the power. The first step is to logon the system and set up the IP address.
Before you can use the Browser to config WellGate 5290, you need to
have Java Standard Runtime (1_4_1_02) to make it work. Please refer to
Appendix 2 Java plug-in Install for detail.

Logon WellGate 5290

Setp1: Start IE5.0 (or later version) to navigate WellGate 5290 Management
System by typing the default IP addresses (the default URL is http://192.168.111.111:10087). The screen will display User ID and Password as figure 2.1-1.
Figure 2.1-1
Note: The default network IP address is 192.168.111.111 and subnet mask is 255.255.255.0
Step 2: Enter log user name and password (the default user id is root and
user password is root). You can manage your user account via web (refer to Section “Account Manager”) later.
Figure 2.1-2
Step 3: The screen shows the Home Page of WellGate 5290 as figure 2.1-3.
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Figure 2.1-3
Figure 2.1-4(SS7)

Network Configuration

Step 1: After successfully logon to the system, we need to change the
network configuration. Click Control > Network -> SIP Service Interface to setup the network parameters as figure 2.2-1.
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Figure 2.2-1
Step 2: Enter the deserved IP address, Submask and default gateway. Apply
the change by clicking apply button as figure 2.2-2.
Figure 2.2-2
Step 3: When screen shows Setup network configuration successfully!”
It means the IP Network setting is successfully changed as figure 2.2-
3.
Figure 2.2-3
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Note: “Network Control” takes around 5-second to apply the new
network configuration. Please logon again with new IP address after 5 seconds.

System Time Configuration

Step 1: When re-logon to the new IP address; the next is to setup the system
time zone. Click Control > System Time Zone to setup the system time zone as figure 2.3-1.
Figure 2.3-1
Step 2: After apply the new time zone, click Back to adjust the date and time
as figure 2.3-2.
Figure 2.3-2
Step 3: Enter current date and time. Apply the change by clicking Apply
button as figure 2.3-3.
Figure 2.3-3
Step 4: The screen will shows “Setup system time successfully!” It means
the System Time setting is successfully changed as figure 2.3-4.
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Figure 2.3-4
Step 5: If you would like to use SNTP to sync time with a SNTP V4 Server,
click Time Sync button to setup it as figure 2.3-5.
Figure 2.3-5

Account Manager

Step 1: You can manage your user account by click Control>Account
Manager. Add a new user account, Click New button as figure 2.4-1.
Figure 2.4-1
Step 2: Enter the new user ID, password as you need. Apply the change as
figure 2.4-2.
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Figure 2.4-2
Field Description:
User ID: Login User ID
Password: Login Password
Confirm Password: Confirm new password again
Step 3: When screen shows “Create user account successfully!” It means
user account setting is successfully created as figure 2.4-3
Figure 2.4-3
Note: The system provides 2 USER ID by default:
User 1: “root” Password: “root”
User 2: “admin” Password: “admin”
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Relogin

Step 1: Click Control>Relogin to relogon by another user account as figure
2.5-1.
Figure 2.5-1
Step 2: Enter new User ID and Password to relogon the WellGate 5290 as
figure 2.5-2.
Figure 2.5-2
Step 3: The screen shows the Home Page of WellGate 5290 as figure 2-5-3.
Figure 2.5-3
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Chapter 3 H.323 Gatekeeper and SIP Proxy Mode
Configuration

Environment used in this chapter

Process:
PSTN → H.323 Call: DNIS (1001) → Make H.323 - Gatekeeper Call (1001) → SIP Call: DNIS (8888) → Make SIP – SIP Proxy Call (8888)
H.323 DNIS (5932111222) DM (H.323_in_drop) Make Call
SIP DNIS (11382265699) DM (SIP _in _drop) Make Call (82265699)
H.323 GK and SIP Proxy Mode
Gatekeeper
H.323 Gateway
Phone 1001
H323 VoIP Network
H323 Phone
(0932111222)
SIP Phone
SIP VoIP Network
Proxy Server
SIP USB Phone

Interface Configuration

This section is going to setup the VoIP interface.
Step 1: Now we are going to setup the VoIP interface, click
Configuration>Interface to setup VoIP T1/E1 interface as figure 3.1-
1.
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Figure 3.1-1
Step 2: Double-click the installed interface (i.e Interface ID: 0) to config it as
figure 3.1-2.
Figure 3.1-2
Step 3: Modify the VoIP Interface parameters (i.e. IP Address, Protocol Tag,
Subnet Mask and Default gateway) and apply the change by clicking Apply as figure 3.1-3.
Figure 3.1-3
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Frequency changed parameters: (Refer to section “Interface
Configuration for more detail)
IP Address: 192.168.19.174
Subnet Mask: 255.255.255.0
Default Gateway: 192.168.19.254
PCM Type: A-law or U-law
Caution: Subnet Mask does not support Super net.
Step 4: After successfully change the Interface configuration, the screen
come back the page of Interface Configuration as figure 3.1-4.
Figure 3.1-4

T1/E1 Trunk Configuration

This section is going to setup the PSTN trunk parameters.
Step 1: Select the installed interface to modify the trunk parameter by click
Detail button as figure 3.2-1.
Figure 3.2-1
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Step 2: Select the trunk to be modified, and click Modify button as figure 3.2-
2.
Figure 3.2-2
Step 3: Modify the trunk parameters (i.e. Trunk Type, Termin Side, Trunk
Mode, Protocol Tag, and Line Code) and apply the change by clicking Apply as figure 3.2-3.
Figure 3.2-3
Frequency Changed Parameters:
Line Code: HDB3
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Clock Master: external
Framing Method: Automatic framing
SS7 Group ID: Circuit Group ID in SS7 config.txt (refer to
ISUP_CFG_CCTGRP gid)
SS7 Circuit Identifier 1: First CID in SS7 config.txt (refer to ISUP_CFG_CCTGRP base_cid)
SS7 Circuit Identifier Count 1: set to 15 (internal user only)
SS7 Circuit Identifier 2: Start CID for time slot 17 (refer to
ISUP_CFG_CCTGRP based_cid + offset)
SS7 Circuit Identifier Count 2: set to 15 (internal use only)
Step 4: After succeeded to change the Trunk Configuration, the screen comes
back the page of Trunk Configuration as figure 3.2-4.
Figure 3.2-4

SS7 Configuration

WellGate 5290 is using Dialogic SS7 board with ISUP module. The user manual can be downloaded from
http://resource.intel.com/telecom/support/ss7/downloads/index.htm. WellGate 5290
provides the Dialogic SS7 configuration utility to edit the Dialogic SS7 board configuration file (config.txt). After you complete the editing of config.txt, you can upload to the system and soft-reset to take effective. However, some advance configuration (e.g. SS7 with voice) is not supported by the Dialogic utility. User need to use the normal editor to edit it.
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Frequency used parameters:
SS7: Enable

H.323 Configuration

This section is going setup the H.323 parameter. If you only need SIP
calls, you can skip it.
Step 1: Click Configuration>H.323 to setup the H.323 parameters for
Gatekeeper related information as figure 3.3-1.
Figure 3.3-1
Frequency used parameters:
Register to Gatekeeper: Yes
Gatekeeper IP: 222.173.67.138
E.164 Tel: 177
Register H.323 ID: Welltech
Step 3: You can see the screen display the new configuration of the H.323
Configuration as figure 3.3-3.
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Figure 3.3-3

SIP Configuration

This section is going setup the SIP parameter. If you only need H.323
calls, you can skip it.
Step 1: Click Configuration>SIP to setup the SIP parameters for SIP Proxy
Server related information as figure.3.4-1.
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Figure 3.4-1
Select SIP Proxy ID 0 and click modify as follows
Frequency used parameters:
Active Mode: Active
Register Server: SIP Proxy Server Register Server
Register Port: SIP Proxy Server Register Port
Register User: SIP User to register
Register Password: SIP User Password to register
Register TTL: 300 (time to live)
Step 2: You can see the screen display the new configuration of the SIP
Configuration as figure 3.4-2.
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Figure 3.4-2

Digit Manipulation

The purpose of “Digit Manipulation” is to add or drop dialed digits for
PSTN or IP side (Interface configuration for PSTN side & H.323 Configuration for IP side) at the selected interface in order to meet local PSTN dialing requirement. It can also be used in Call Flow Edit for flexible usage.
Step 1: We introduced the group and interface dependent digital manipulation
to meet the customer’s requires. Click Digit Manipulation to add a new Digit Manipulation Group, add as figure 3.5-1.
Figure 3.5-1
Step 2: Enter the related parameters and click Apply button as figure 3.5-2.
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Figure 3.5-2
Field Description:
Group ID: 0 (DM Group identify)
Description: H.323: H.323 In Drop
SIP: SIP In Drop
Step 3: Click the New created DM and Detail button to add digits setting as
figure 3.5-3.
Figure 3.5-3
Step 4: Click New button to add a new DM rule as figure 3.5-4.
Figure 3.5-4
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Step 5: Create a new H.323 DM Group “1” and DM detail is show as follows:
Figure 3.5-5
H.323 Incoming Call DM Setting:
Matched Pattern: 5(pattern to be matched)
Group ID: 1-add002 (belong to this DM group)
Insert: The Insert digits
H.323 incoming call
Dialed number: 582265699
Match the pattern 5
Add 002 (Insert)

New dialed number becomes 002582265699

Step 5: Also create a new SIP DM Group ‘2” and DM detail is show as follows:
Figure 3.5-6
SIP Incoming Call DM Setting:
Matched Pattern: 113 (pattern to be matched)
Group ID: 2-sophie (belong to this DM group)
Drop: 113 (drop digits)
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Welltech Computer Co., Ltd.
SIP incoming call
Dialed number: 11307688222
Match the pattern 113
Delete 113 (Drop)
New dialed number becomes 07688222
Step 6: Create a PSTN incoming call DM Group “3” and DM detail is show as
follows:
Figure 3.5-7
PSTN DM Setting:
Matched Pattern: 02822656 (pattern to be matched)
Group ID: 3-PSTN (belong to this group id)
Drop: 02822656 (drop digits)
PSTN incoming call (DNIS mode)
Dialed number: 028226561001
Match the pattern 02822656
Delete 02822656 (Drop)
New dialed number becomes 1001
Note: Digit Manipulation have to tapped for PSTN Side
(Trunk>Outbound/Inbound DM Group), VoIP Side (VoIP>Outbound/Inbound DM Group) or Call Flow (refer to section “Call Flow Editor”) to take effect.
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Chapter 4 Call Flow Editor

Call Flow Editor is used to control the call behavior including voice
prompt, AAA, DM…etc. It requires Java run time to run.
Step 1: Click Control>Call Flow Editor to create a Call Flow, click button
to activate IVR Tool as figure 4.1
Figure 4.1
Component Description:
New: Create a new call flow
Paste: Paste a component
Delete: Delete a component
Line: Connecting 2 components together
Select: Select the component at call flow workspace
Scroll: Scroll the call flow workspace
Zoom: Zoom in or zoom out the workspace
View Grid: View or not
Load Call Flow: Load call flow from WellGate 5290
Save: Save a call flow in WellGate 5290
Cut: Cut a component
Copy: Copy a component
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Show Component Table: Show all component table
Step 2: Drag and prop the required component icon into the workspace as
figure 4.2-1
Figure 4.2-1
Right click the component to bring up the component propriety to setup parameter:
AAA: Send Authorization or Authentication for validation
Figure 4.2-2
o Type: AAA type selection
- Authorization: Send RADIUS Authorization packet out
- Authentication: Send RADIUS Authentication packet out
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Success To: Success to component Failed other to: Failed to component
o Failed Reason: Return code from RADIUS server
o Line Propriety:
- Invalid Account
- Account In Use
- Zero Balance
- Account Expired
- Over Credit Limit
- Number of Retries Exceeded
- Insufficient Balance
Note: Detail response attributes, please refer RADIUS Format
Attributes
Answer: Answer incoming call (PSTN only)
Branch: Play an announcement and branch into different route
Figure 4.2-3
o Voice File: Voice prompt file (“. raw” format) to be playing
o DTMF Length: Number of DTMF to be receiving
o Others: Default flow if not match
o DTMF: DTMF match pattern
o Goto: The next component if matched
o Line Propriety:
-Branch Line: DTMF branch line setting
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Figure 4.2-4
CDV: Collected Digit Validation
Figure 4.2-5
o Check Parameter: Check parameter type (DNIS, ANI….)
o Digit From: Start digit from
o Digit To: End digit to
o Valid To: If the checked variable is success to validate
o Invalid To: If the checked variable is not success to validate
CIB: Call Information Branch
Figure 4.2-6
o Info Type: Information type selection
- ANI: Calling Number
- DNIS: Called Number
- IP: IP Address or network (e.g. 192.168.0.0)
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- PSTN: E1/T1 trunk and channel filter, format: interface id-
trunk id- trunk start- trunk stop
- Prefix: The prefix to be match
0-1-17-31:
0: Interface ID (Always 0) 1: Trunk ID: 1 17: Start from B Channel 17 31: Stop from B Channel 31
o Goto: The component to run next
o Call Info Branch Line: ANI, DNIS, IP or PSTN goto setting
CIV: Call Information Validation, the user need setup the ACL for DNIS and IP TO take effect
o Info Type: The info type to be validation
-DNIS: Called number
-ANI: Calling number
-IP: In coming IP address
-User: User ID
o Allow To: If it is met the ACL defined o Disallow To: If it is not met the ACL defined
CTB: Call Type Branch
o PSTN To: Route for PSTN call o H.323 To: Route for H.323 call o SIP To: Route for SIP call
Figure 4.2-7
Figure 4.2-8
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Cut Rule: Cut a system variable into different parts
Welltech Computer Co., Ltd.
Figure 4.2-9
o Cut From: Cut start digit from (start from 1) o Cut To: Cut end digit to o Assign To: Store the cut result into
Disconnect: Disconnect the call
DM: Digit Manipulation
Figure 4.2-10
o DM Parameter: Manipulation ANI or DNIS o DM Group ID: Apply to DM group
Make Call: Make Call to PSTN or H.323/SIP site
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Figure 4.2-11
o Route Mode: Gatekeeper Call or P2P Call or PSTN…etc. (for PSTN
incoming call, please select the Gatekeeper, P2P Call, or SIP Proxy call TA; for H.323/SIP incoming call, please select the PSTN call)
o Transport Address: When used for “H.323 TA” routing mode, the
format used is “Ipaddr: port” (e.g. 192.168.111.50:1720)
o Active Disconnect: Enable PSTN user can actively disconnect the
call or not
o Active Disconnect Digit: The DTMF digit to be tread as the
disconnect trigger. Only can be used “Active Disconnect” enable
o Active Disconnect To: The next component when active disconnect
is occurred
o Inter Digit Timeout: The max time to in seconds to wait between two
digits.
o RTP Route: Voice RTP routing over WG5290 or not, for VoIP to
VoIP call
o Finish To: Successfully connect to remote site o Failed Other to: The next component when default failed call o Failed Reason: Failed reason selection o Failed To: When the failed reason occurred go to o Line Propriety:
-PSTN: PSTN disconnect reason code:
-Normal Call Clear
-User Busy
-No User Response
-No Answer
-Call Reject
- VoIP: VoIP disconnect reason code:
-User Busy
-No Answer
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-Unreachable
-Other
PA: Play Announcement
o Dynamic Play: Dynamic play voice file by combine prefix and
variable as the file name
o Enable: Combine prefix to variable as the voice file to play
-Prefix: Voice file prefix (e.g. prefix: WT, variable: user1 (contact 201, played voice file is “WT201.raw”)
-Variable: Variable to be appending as the voice file name
o Disable: Use filter voice prompt file
-Voice File: Voice prompt file
o Interrupted: Voice can be interrupted or not
Welltech Computer Co., Ltd.
Figure 4.2-12
PB: Play Balance for prepaid purpose
Figure 4.2-13
o Voice File: Voice prompt file o Language: Play balance language section
-English
-Chinese.
o Interrupted: Voice can be interrupted or not
PCUI: Prompt and Connect User Information
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Figure 4.2-14
o Play Type: Dial tone or voice prompt selection o Voice File: Voice prompt file o Max DTMF: Maxtor of DTMF to be received. o Assign To: Result (received DTMF) will be assign to o End of DTMF: The digit to indicate dial end. o Interrupted: Voice can be interrupted or not
PD: Play Duration for prepaid purpose
Figure 4.2-15
o Voice File: Leading voice prompt file o Language: Play duration language section
-English
-Chinese
o Interrupted: Voice can be interrupted or not
Note: The RADIUS servers need to be setup to send H.323/SIP
credit time or internal RADIUS must be used.
PSTN L.H: PSTN Line Hunting
Figure 4.2-16
o Success To: If fine an available channel by system setup (call
hunting)
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o Failed To: If not fine an available channel by system setup (call
hunting)
Set Data: Assign value to a variable
Figure 4.2-17
o Assign To: Assigned variable o Use SysParam: Use system parameter to replace or not o Value: ANI, DNIS, User ID or other digits. You can also use
database number to generate the required random number.
o Support random ANI (use ? to indicate random digit)
Start: Call flow start
Figure 4.2-18
o Next Component
Quit: Disconnect calls
Example Call Flow as figure 4.3.
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Figure 4.3
Example Description:
Components Contents
Start
Next Component: 1001
Component ID: 1000
PSTN To: 1011
CTB
Component ID: 1001
H.323 To: 1009 SIP To: 1008 Info Type: ANI
CIB
Component ID: 1011
Prefix: 1 goto: 1010 (H.323 GK call) Prefix: 8 goto: 1004 (SIP Proxy call)
1011 Route for PSTN call
Route Mode: Gatekeeper
Make Call
Component ID: 1010
Finish To: 1005 Failed Other To: 1005 Route Mode: SIP Proxy Call
Make Call
Component ID: 1004
Finish To: 1005 Failed Other To: 1005 Next Component: 1006
Disc
Component ID: 1005
QUIT
Component: 1006
1001 Route for H.323Gatekeeper call
DM Parameter: DNIS
DM
DM Group ID: H.323 In Drop Next Component: 1007
Component ID: 1009
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Route Mode: PSTN
Make Call
Finish To: 1005 Failed Other To: 1005
Component ID: 1007
Next Component: 1006
Disc
Component ID: 1005
Quit
Component: 1006
1001 Route for SIP Proxy call
DM Parameter: DNIS
DM
DM Group ID: SIP In Drop Next Component: 1007
Component: 1008
Route Mode: PSTN
Make Call
Finish To: 1005 Failed Other To: 1005
Component ID: 1007
Next Component: 1006
Disc
Component ID: 1005
Quit
Component: 1006
Example Used Call Flow:
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to 1009 H.323 in
Welltech Computer Co., Ltd.
Start:1000
CTB:1001 PSTN to:1011 H323 to: 1009
SIP to: 1008
to 1008
SIP in
1011
PSTN in
DM: 1009
Call to PSTN
Make Call: 1007
Call to PSTN
Finish to
Failed other to
DM: 1008
Call to PSTN
Disc: 1005 Disconnect
1
Make Call: 1010
Mak e
Gat ekeeper
call to H. 323
Success /Failed to
CIB: 1011
Info Type: DNIS Prefix: 1 goto: 1010 Prefix: 8 goto: 1004
Success /Failed to
8
Make Call: 1004
Mak e
SIP Proxy
call to SIP
Quit: 1006 Disconnect

Config Manager

Configuration Management provides a way to save and reload the
system configuration for future use.
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Figure 4.4-1
Backup the working configurations: Step 1: To backup the running configuration, click on Backup button, to back
up local hard disk. The whole running configuration will be compress into a zip file (file name: export.zip) and transfer back to local as figure
4.4-2.
Figure 4.4-2
Restore configuration: Step 2: To restore the backup configuration file, click on Restore button,
when screen shows “Import Configuration file”, select backup file (i.e. c:\export.zip) click on Import button to restore the configuration to the working configuration as figure 4.4-3.
Figure 4.4-3
Compact the database file: Step 3: In order to optimize the system performance, you can optional
compact the database by click Compact button as figure 4.4-4.
Figure 4.4-4
Note: Please make sure that there is no others person to use database
concurrently.
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Apply Change

When you load a new working configuration, the system must be
restarted to take effect.
Step 1: Click Configuration>Apply Change, the screen show “The change
you mode need to restart the system for apply please confirm to restart or do it later.” Click on OK/Cancel to restart the system or
not as figure 4.3-1.
Figure 4.5
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Chapter 5 Peer to Peer Mode Configuration

Environment used in this chapter

Peer to Peer
H.323 Gateway
Phone 1001
H323 VoIP Network
H323 Phone
SIP Phone
SIP VoIP Network
SIP USB Phone 8888
Process: PSTN> H.323 Call: DNIS (822656991001) >DM (PSTN In Drop) >Make
H.323 - Peer to Peer Call (1001)
> SIP Call: DNIS (822656998888) >DM (PSTN In Drop) >Make SIP -
Peer to Peer Call (8888)
H.323>DNIS (50932123321)>DM (H.323_in_drop)>Make Call (0932123321)
SIP>DNIS (1130028610825123)>DM (SIP_in_drop)>Make Call (0028610825123)
Digit Manipulation: Please refer to section “Digit Manipulation

Network Configuration

Please refer to section “Network Configuration

Account Manager

Please refer to section “Account Manager

Interface Configuration

Please refer to section “Interface Configuration

H.323 Configuration

Step 1: Change Register to Gatekeeper to “No” to enable peer to peer mode
as figure 5.1-1.
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Frequency used parameters:
Register to Gatekeeper: No
Figure 5.1-1

VOIP Hunting

For making a Peer-to-Peer call, the IP device must has an address
record in the phone book for routing.
Step 1: To add a new VOIP routing group for the peer to peer calls, click
VOIP Hunting -> New to add as figure 5.2-1.
Figure 5.2-1
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Field Description:
Group ID: The Group ID
Description: The description of this VOIP hunting group ID
Step 2: To add a routing point into the VOIP hunting group, click the hunting
group ID and Detail button as figure 5.2-2.
Figure 5.2-2
Field Description:
Name: The used name
Tel/Prefix: The matched prefix
Trans Address: transport address
- H.323 Call: 192.168.5.102 or 192.168.5.102:1720
- SIP Call: sip:1001@192.168.5.61 or sip:8001@192.168.5.61:5060 or
sip:8001@ctivnet.net
Use SIP Proxy: use the selected SIP proxy instead of Transport address
Priority: It is used for priority hunting. Higher number is indicating the high priority to be used. Same priority routing item will be round-robin hunted.
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Remove Prefix: Remove the TEL/Prefix from the final destination number (DNIS) or not
Add Prefix: After removed the prefix (if remove prefix is checked), add the “Add Prefix” in the front of final destination number (DNIS)
DNIS DM Group: To execute the DNIS DM Group or not. It will be executed after “Add Prefix”.
Working Time: The effective time for this route. It needs to be used with the working day together.
Working Day: The effective week day for this route. It needs to be used together with working time.
Allow no ANI: Does this route is allowed for a call without ANI or not.
Step 3: The hunting can be stopped based on the following reason code. It
can be used as a load balancing purpose.
Figure 5.2-3

Digit Manipulation

Please refer to section “Digit Manipulation

Call Flow Editor

Please refer to section “Call Flow Editor
WellGate 5290 User Guide – V6.1 - 52 -
Call Flow (P2P Mode):
to 1009
H.323 in
Welltech Computer Co., Ltd.
Start:1000
CTB:1001 PST N to:1011 H323 to: 1009
SIP to: 1008
to 1008
SIP in
1011
PSTN in
DM: 1013
Call to PSTN
DM: 1009
Call to PSTN
Make Call: 1007
Call to PSTN
Finish to
Failed other to
DM: 1008
Call to PSTN
Disc: 1005 Disconnect
Make Call: 1004
Make
Peer to Peer
call to SIP
Success /Failed to
Quit: 1006 Disconnect

Configuration Manager

Please refer to section “Configuration Manger

Apply Change

Please refer to section “Apply Change
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S
SIP Phone

Chapter 6 SIP to H.323 Mode Configuration

Environment used in this chapter

Process: SIP>H.323 Call: DNIS (8861001)>DM (SIP In Drop)>Make H.323 (1001)
H.323>SIP (8868888)>DM (H.323_in_drop)>Make Call (8888)
Digit Manipulation: Please refer section “Digit Manipulation
IP to H.323 Mode
H323 VoIP Network
H.323 Gateway
Phone 1001
H323 Phone
SIP VoIP Network
SIP USB Phone 8888

Network Configuration

Please refer to section “Network Configuration

Account Manager

Please refer to section “Account Manager

Interface Configuration

Please refer to section “Interface Configuration

H.323 Configuration

Please refer to section “H323 Configuration

SIP Configuration

Please refer to section “SIP Configuration

Address Book

Please refer to section “Address Book
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Digit Manipulation

Please refer to section “Digit Manipulation

Call Flow Editor

Please refer to section “Call Flow Editor
Call Flow (P2P Mode):

Configuration Manager

Please refer to section “Configuration Manger

Apply Change

Please refer to section “Apply Change
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Chapter 7 Advance Configuration Reference

Configuration

System Configuration

Start Path: Configuration>System
Figure 7.1-1
Parameter Description:
CDR Mode: Call detail record generating mode (Refer to “Appendix 3
Retrieve CDR Information” for detail file description)
o File Only: Log CDR into the file only. It can be retrieved by ftp
(directory c:\cd cdr).
o Radius Start/Stop: Log CDR into the file and send RADIUS start/stop
billing message out.
- VoIP: enable VoIP site RADIUS billing message or not.
- PSTN: enable PSTN site RADIUS billing message or not.
o Radius Stop: Log CDR into the file and send RADIUS stop billing
message out.
- VoIP: enable VoIP site RADIUS billing message or not.
- PSTN: enable PSTN site RADIUS billing message or not.
CDR Keep days: CDR system keeping days
Hot Swappable: Hot swappable support (reserved)
First Digit Timeout: The max time (in second) waits for receiving the first
digit entered (5~20 sec).
Inter Digit Timeout: The max time (in second) waits for the between two digits (5~20 sec).
Debug Level: o Critical: Show critical error messages only
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o Warring: Show warring and critical error message only o Information: Show information, warring and critical message only o Debug: Show all debug messages o Full Trace: Show all status and debug messages
SS7 Debug Level: SS7 related debug info or not.
o Critical: Show critical error messages only o Warring: Show warring and critical error message only o Information: Show information, warring and critical message only o Debug: Show all debug messages o Full Trace: Show all status and debug messages
Note: Please set to “Critical” only, or the whole system performance
will be hit.
First Response Timeout: The default maximum time to wait for response. It’s depended on the network speed.
No Answer Timeout: The default maximum time (in second) to wait the remote party Answer (pick up phone).
Ext. AAA Failure Opt: Bypass or disconnect incoming calls when external RADIUS cannot be reached.
Version: 6.0
Data Base Number Select Mode: The Number Select Mode In Data Base
Enable System Log: Enable to send system information to syslogD
Server or not
SyslogD Server IP 1, 2: syslogd server IP address
Event Email Notice :The Email Notice For Event
Email Event Filter Level :The Filter Level for Email Event
SMTP Server : SMTP server host for email notice
Email From : Email sender account
Email To : Email receiver (semicolon is used for multiple receiver)
Subject: Email subject to be send to receiver. The following variable

Interface Configuration

Start Path: Configuration>Interface
Figure 7.2-1
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Basic Parameter Description:
Interface ID: The Interface ID
Card Slot: The slot that the card is plug in
Interface Type: The type of interface, E1 or T1
Description: Description
IP Address: IP address used for voice RTP stream
Subnet Mask: Sub-mask (doesn’t support super class)
Default Gateway: Default gateway for routing
PCM Type: PCM type encoding, E1 A-law; T1 u-law
Advance Interface Configuration: Start Path: Configuration>Interface >Advance
Figure 7.2-2
Advance Parameter Description:
Interface ID: System parameter
UDP Port Base: UDP port used for RTP stream, each channel needs 3
RTP ports and must be started by a multiple of 10
IP Precedence: Voice package priority setting
o Routine Precedence o Priority Precedence o Immediate Precedence o Flash Precedence o Flash Override Precedence o Critical Precedence o Internet work Precedence o Network Precedence
IP TOS: Top of Service with the following priority selection
o Normal Service o Minimize Monetary o Maximize Reliability o Maximize Thought o Minimize Delay
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PCM Idle Pattern: This pattern will be sending on each B channel PCM time slot when the channel is idling (not connected). The default value for A-Law is 0xff and for Mu-Law is 0x55. You only change it when SWITCH need.
CAS Idle Pattern: When channel is idle, ABCD (CAS) pattern to be applied CAS signaling bus
Jitter Min Delay: The minimum delay time of Jitter buffer. The range is 0 to 150ms. Default value is 150ms. Which has better voice quality but the delay time will be long.
Jitter Opt Factor: Jitter buffer optimization factor from 0 to 12. The default value is 7. Set to 0 will have lowest voice delay but have bad voice quality. Set to 12 will have long voice delay but with better voice quality
EC Tail Length: Echo Cancellation Length, default value is 32ms
Silence Compress: Enable silence compress or not
TDM Bus Clock: TDM Bus clock source
o Internal: derived from internal oscillator o External: derived from external PSTN E1/T1 clock
Detect Fax Mode: T.38 fax detecting method
o Answering Fax only: Detect answer tone to initiate T.38 o Fast Fax Detect: Detect calling and answer tone for T.38. Use
Answering fax only when fast fax detect miss detect a fax call.
Auto Clock: Automatic detect trunk clock source. It is recommended to specify clock source instead of use auto clock.
Local Reference: TDM bus reference clock. Please set to a connected trunk.
Overlap Digit: Enable to receive ISDN/PRI Q.931 overlapping sending or not.
DTMF Interval: Wait overlapping digits time out
CAS Send Restart: When channel is blocked, send restart or not.
DTMF Volume: DTMF relay volume (default 16: 0 db)
Active T38 when Open Channel: Re-open DSP channel when T.38 is
started.
PSTN Show Display Name: Send ISDN display name IE or not (network side only)
SIP Show Display Name: Relay ISN display name to SIP or not
Reset Trunk When Link Up: When trunk is started, restart channels or
not.
Dial Plan Configuration
Dial Plan can be used to assign the ISDN number plan based on prefix
setting.
Start Path: Configuration>Interface>Dial Plan
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Figure 7.3-1
Basic Parameter Description:
Prefix: Called party number prefix
Src Num Plan: ISDN Source number plan
Src Num Type: ISDN Source number type
Dest Num Plan: ISDN destination number plan
Dest Num Type: ISDN destination number type
Apply To: Trunks apply to
Overlap Configuration
It is used for incoming ISDN dialing plan.
Start Path: Configuration>Interface> Overlap
Figure 7.4-1
Click New to create a overlap dialing plan as follows:
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Figure 7.4-2
Prefix: Matching prefix (longest match first)
Max Length: Max length for this prefix dial plan. The dial plan will be fulfill
if max length is met.
Apply To: Apply Target to which trunk or all trunk
T1/E1 Trunk Configuration
Start Path: Configuration>Interface>Trunk
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Figure 7.5-1
Basic Parameter Description:
Interface ID: System parameter
Trunk ID: System parameter
Trunk Type: T1or E1 selection (SS7 support E1 only)
Description: Description for this trunk ID
Termin Side: Network site or User Site (normally, you set to “user side
when connect to switch)
o User Side o Network Side
Trunk Mode: Trunk operation mode
o Disable: Disable the trunk o Normal: Accept PSTN and VoIP calls o PSTN incoming only: Allow the PSTN incoming calls only o VOIP incoming only: Allow the VOIP incoming calls only
Hunting Method: PSTN trunk hunting method for available channel
o Random: Hunt randomly o Cyclic: Initial hunt (after power-up/reboot) begins with B channel 1;
subsequent hunts begin with position following last successfully allocated resource
o Rotary: Hunt always begins with B channel 1 o Reverser Rotary: Hunt always begins with B channel 31
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o Reverser Cyclic: Initial hunt (after power-up/reboot) begins with B
channel 31, follows next available channel in reverser order
CAS Variance: CAS counting variance
Framing Method:
o For T1
- Super frame
- 4-frame multi-frame
- 12 frame multi-frame (D4)
- extend super frame without CRC6
- extend super frame with CRC6
- 72-Frame Multi-Frame
o For E1:
- Automatic CRC4 or Double Frame selection
- Double Frame Format
- CRC4 multi-frame
- CRC4 extend multi-frame
Protocol Tag: supported protocol on T1/E1 interface with PSTN switch
o For T1:
- T1 CAS
- T1 RAW CAS
- T1 NI2 ISDN
- T1 4ESS ISDN
- T1 5ESS 9 ISDN
- T1 5ESS 10 ISDN
- T1 DMS100 ISDN
- T1 NTT ISDN: used to connect NTT INS-1500 ISDN standard (Japan
Only)
- T1 HKT ISDN
- T1 QSIG
- T1 EURO ISDN
- T1 DMS100 MERIDIAL ISDN
- T1 NI1 ISDN
o For E1:
- E1 EURO ISDN: used for most of European ISDN standard
- E1 MFCR2
- E1 CAS
- E1 RAW CAS
- E1 AUSTEL ISDN: Australia E1 ISDN Variance
- E1 HKT ISDN: Hong E1 ISDN Kong Variance
- E1 KOR ISDN: Korea E1 ISDN Variance
- QSIO
- E1 TNZ ISDN
Line Code: T1: you can choose AMI, B8ZS; E1: you can choose AMI, HDB3
PSTN Trace: PSTN layer debug trace. It will generate a debug trace file for tracing purpose. Only enables it under Welltech technical supports instruction and disable it when complete the debug
Inbound DM Group: Digit Manipulation group used for incoming calls associated to this trunk
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Outbound DM Group: Digit Manipulation group used for outgoing calls
Local Ring Back: Provide ring back tone for PSTN or not. It only works
when VoIP outgoing Fast Start is disabled.
Channel Mask: Channel mask for incoming or outgoing calls (default: 0xffffffff) Start from MSB each bit, indicate a time, slot a trunk (e.g. 0x0000ffff: 0~15 B channel mask, 17~31 B channel free)
Clock Master: PSTN trunk clock source
Framing Method: Automatic framing
SS7 Circuit Group Mode: Enable the this SS7 Circuit group (IN Service)
or not (Out Service)
SS7 Group ID: Circuit Group ID in SS7 config.txt (refer to ISUP_CFG_CCTGRP gid)
SS7 Circuit Identifier 1: First CID in SS7 config.txt (refer to ISUP_CFG_CCTGRP base_cid)
SS7 Circuit Identifier Count 1: set to 15 (internal user only)
SS7 Circuit Identifier 2: Start CID for time slot 17 (refer to
ISUP_CFG_CCTGRP based_cid + offset)
SS7 Circuit Identifier Count 2: set to 15 (internal use only)
Advance Trunk Configuration: Start Path: Configuration>Interface>Trunk>Advance
Figure 7.5-2
Advance Parameter Description:
Interface ID: System parameter
Trunk ID: System parameter
Src Num Plan: ISDN source number plan
Dest Num Plan: ISDN destination number plan
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Src Num Type: ISDN source number type
Dest Num Type: ISDN destination number type
Src Num Presen: ISDN source number presentation
Src Num Screen: ISDN source number display
Input Gain: Voice Gain from PSTN to IP side (default: 0 db)
Output Gain: Voice Gain from IP to PSTN side (default: 0 db)
Q.931 General Opt.: used for Q.931 general behavior.
o 0x00000001: No sending of STATUS message on receipt of an
unknown IE. If this bit is set, then ISDN Stack does not generate a STATUS message after having received a message containing 1 or more unknown/unrecognized IE(s). This bit applies only for network variants for which the sending of STATUS in this case is optional (where recommendation says “ a SATUS MAY be returned”)
0x00000002: No sending of STATUS message on receipt of an
o
optional IE with invalid content. If this bit is set, then ISDN Stack does not generate a STATUS message after having received a message containing 1 or more optional IE(s) with invalid content. This bit applies only for network variants for which the sending of STATUS in this case is optional (where recommendation says “a STATUS MAY be returned”).
o
0x00000004: Unknown/unrecognized Facility IE accepted. If this bit
is set, then NS does not reject a received message containing a Facility IE that it does not recognize and acts as if the IE was correct. It applies in network variants where a complete ASN1 decoding is made on Facility IE
o 0x00000080: Only applicable when configuration is Euro ISDN,
User-side, outgoing call . If this bit is set, then the board sends a CONNECT_ACK message when receives a CONNECT.
o 0x00000100: Only applicable when configuration is ETSI. If this bit
is set, BC and HLC IE’s in Alert, CALL_PROC, CONNECT and PROGRESS message are rejected.
0x00000200: Only applicable when configuration is 4ESS, DMS,
o
5ESS, N12 or HKT, NFAS-both incoming and outgoing calls. If this bit is set, then ISDN calls on NFAS have an interface ID different to the default one (refer to NFAS Appendix). The interface ID value is taken from the Board param of the NFAS Member Trunk config. (i.e.,board_params. Trunk Config[i]. Protocol Specific. ISDN Trunk. ISDN_NFAS_Interface ID).
o
0x00000800: Only applicable when configuration is 4ESS, DMS,
5ESS or N12. If this bit is set, the channel ID is always set to EXPLICIT interface ID, even if the B-channel is on the link bearing the D-channel.
o 0x00008000: Only applicable when configuration is ETSI. If this bit
is set, Mu-law (μ-Law) is also accepted. Otherwise, only A-law is accepted.
o 0x00010000: Only applicable when configuration is ETSI. If this bit
is set, calling party number octet 3a is always present, even if presentation and screening have default values. Otherwise, usual rule is used.
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o 0x00020000: Only applicable when configuration is N12, QSIG or
ETSI. If this bit is set, the call is cleared on reception of STATUS with incompatible state. Otherwise, no action is taken
Q.931 Incoming Opt.: used for Q.931 incoming call behavior o 0x00000010: Automatic answering on TELEPHONY incoming
calls When this bit is set, Board sends a CONNECT (answer) message on TELEPHONY incoming calls.
o 0x00000020: Automatic answering on NOT TELEPHONY incoming
calls: When this bit is set, Board sends a CONNECT (answer) message on NOT TELEPHONY incoming calls.
o 0x00000800: When this bit is set, the ISDN Stack forces the
sending of the Channel-id Information Element in the first response to an incoming call indication. When this bit is not set, The ISDN Stack forces it only if it is mandatory for the running operator, or if on NT-side. This have been identified as solving a problem with Coral PBX that requests a Channel in the first respond to an incoming call from it .
o 0x00001000: When this bit is set, the ISDN Stack does NOT
automatically generate a Setup Ack. When this bit is not set, the ISDN Stack sends a Setup Ack when it detects overlap receiving in progress.
o 0x00002000: When this bit is set, the ISDN Stack forces the
sending of Channel-id Information Element(IE) in the Call­Proceeding message. When this bit is not set, the ISDN Stack forces it only if it is mandatory for the running operator, or if on NT­side.
o 0x00010000: When this bit is set, the board automatically sends a
Progress-Indicator #8 (“in-band tones/announcements available”) in the Setup-Ack message that is sent back automatically by the board when it receives and accepts a Setup message in overlap-receiving mode (e.g no digit in the Setup), for Voice calls only (BC encoded as Speech or 3.1-Audio). This enables tone generation to the Terminals after handset off-hook.
Q.931 Outgoing Opt.: used for Q.931 outgoing behavior o 0x00000002: Automatic generation of ‘Sending-complete’
information element in SETUP (as result of Place Call)When this bit is set, Board does not generate automatically the 'Sending-
complete' information element in the SETUP. The User application must request it in the Place Call function. Otherwise (if this bit is not set), the Board generates it automatically in the SETUP message only.
o 0x00000010: use Mu-law if this bit is set, or A-law will be used.
Apply only for Korea variance, you can OR the required option together
o 0x00000080: When this bit is set, ISDN stack uses the KEYPAD IE
to store the called number digits instead of the CALLED_NB IE. Only applicable for KOR variant (Korean network). Useful for Korean switches that do not accept the Called_nb IE.
o 0x00000100: When this bit is set, ISDN Stack forces the sending of
a Channel-id IE in an outgoing SETUP message even if it is not
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required by the standard(i.e., it is optional only ), and no Channel-id was specified in the establishment request. This is useful to increase compatibility with switches that require it. On PRI lines it indicates a not yet used channel id, preferred only.
o 0x00000200: When this bit is set, ISDN Stack sends G..711 A-Law
in outgoing voice calls. When this bit is cleared, ISDN Stack sends default G.711-Mu-Law in outgoing voice calls. Applicable for 5ESS variant(T1).
Trans Cap: Transfer Capability
o Voice Service o Data Service o Modem Service
Call ID Transfer Type: Call ID transfer type
o Disable Caller ID: default parameter o Transparent Caller ID o Relay Caller ID o Bypass Caller ID

Reset Configuration

Reset a specified channel or full trunk.
Start Path: Configuration>Interface>Detail>Reset
Figure 7.6-1
Start Path: Configuration>Interface>Detail>Reset
Figure 7.6-2
Basic Parameter Description:
Trunk: Reset trunk ID
Channel: Rest channel selection
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- All Channel: Reset all channel
- 0~31: Reset 0~30 logical channel to reset

H.323 Configuration

Start Path: Configuration>H.323
Figure 7.7-1
Basic Parameter Description:
Register To Gatekeeper: Register to Gatekeeper or not
o Yes: Register to GK o No: Not register to GK
Gatekeeper IP: Gatekeeper IP Address
Gatekeeper RAS: UDP Port number listened on Gatekeeper (default:
1719)
E.164 Tel: Telephone number to be registered to Gatekeeper
Register H.323 ID: H.323 alias name to be registered to Gatekeeper
Register Time To Live (sec): The registration maximum time to live
setting when registered to the Gatekeeper
Response Timeout (Q.931)(sec): The maximum time to wait for response from sending call setup signal out
Connect Timeout (Q.931)(sec): The maximum time to wait for connection (answer) from dialing out the destination number
DTMF Relay: DTMF transfer type selection o RTP relay (RFC 2833): DTMF relay via RTP packet (RFC2833
standard)
o DTMF transparent: transmitter DTMF over voice channel o H.245 Signal input: DTMF relay via H.245 user signal input o H.245 Alphanumeric: DTMF relay via H.245 Alphanumeric signal o Q.931 User Information: DTMF relay via Q.931 User to user
information
Fax Transport: Fax transport type selection
o Transparent mode: Transparent mode (by voice packet) o T.38 Fax Relay (H.245 mode): T.38 Fax relay (H.323 Annex D) o T.38 Fax Relay (Propriety mode): T.38 Fax Relay (propriety mode)
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o FRF11 Fax Relay (Propriety mode): FRF11 Fax Relay (propriety
mode)
Fast Connect Mode: Connection of H.323 call fast mode
o Disable: Don’t use Fast Start. o Enable Fast Start Both Site: Use Fast Start for incoming call and
outgoing H.323 calls
o Fast Start-H.323 incoming only: Enable Fast Start for H.323
incoming calls only
o Fast Start-H.323 outgoing only: Enable Fast Start for H.323
outgoing calls only.
o Early H.245: Use Early H.245
H.245 Tunneling: Transfer the H.245 message over the Q.931 channel
FS Enable 1-6 (Codec Priority 1-6): Enable Fast Start codec selection
for each codec
Inbound DM Group: Digit Manipulation Group for H.323 incoming calls
Outbound DM Group: Digit Manipulation Group for H.323 outgoing
calls
Advance H.323 Configuration: Start Path: Configuration>H.323 >Advance
Figure 7.7-2
Advance Parameter Description:
RAS Multicast IP: RAS multicast IP for Gatekeeper searching
RAS Multicast Port: RAS multicast Port for Gatekeeper searching
Max Call: The maximum H.323 calls
Max Channel: The maximum channel of each H.323 call
RAS Port: Local RAS port (default: 1719)
Q.931 Port: Local TCP port number of Q.931
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T.38 ECM Mode: T.38 Error Correction Mode
o T.38 ECM Interoperable mode o T.38 ECM Backward Compatible Mode
FAX RDepth: T.38 relay redundancy packet depth for high-speed mode.
H.245 Option: Separate the H.245 channel in the call of the Fast Start
mode or not.
G.723 Psize: G.723 transmission packet size in ms (default: 30ms)
G.729 Psize: G.729 transmission packet size in ms (default: 20ms)
G.711 Psize: G.711 transmission packet size in ms (default: 20ms)
Wait H323 Idle Timeout (sec): Max time to wait H.323 idle when call is
in disconnected state. After time out, the call will be terminated.
H.323 Delay Drop Timeout (msec): Delay to drop a H.323 call (internal use only)
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SIP Configuration

Start Path: Configuration>SIP
Figure 7.8-1
Basic Parameter Description:
Codec Selection Policy: Selection order to match the remote SDP for codec selection.
o Local SDP Order: Use local SDP order to match codec o Remote SDP Order: Use Remote SDP order to match codec
Local Codec 1~4: Codec selection priority (1 to 4) (1: highest, 4: lowest)
G.723 Bit Rate Used: G.723.1 high bits rate (6.3k) or low bit rate (5.3k)
is used
180 SDP: Set SDP for 180 ring message
183 SDP: Set SDP for 183 call progress indication.
DTMF Relay Method: DTMF transport type selection
o Transparent: transmit DTMF over audio channel o SIP INFO: Use SIP INFO Message to relay DTMF o RFC2833: Use RFC2833 for DTMF over RTP packet
- RFC2800 Payload Type: RTP payload type used for RFC2833
DTMF relay
Fax Transmission: Fax transparent type selection
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o T.38 Fax Relay: T.38 fax relay o Transparent: Transparent mode (by voice packet)
Accept Proxy Call Only:
o Yes: Only call from outbound proxy server is allowed o NO: Accept any SIP calls
Inbound DM Group: Digit Manipulation Group for SIP incoming calls
Outbound DM Group: Digit Manipulation Group for SIP outgoing calls
Advance SIP Configuration: Start Path: Configuration>SIP >Advance
Figure 7.8-2
Advance Parameter Description:
TCP Enable: Receive SIP TCP call or not.
Max TCP Connection: Max Call: The maximum SIP TCP calls.
Outbound Use TCP: Use SIP TCP for outbound call or not. If it set to
no, UDP is used.
Register Use TCP: Use SIP/TCP to register to SIP register.
TCP Port: The local TCP port on which the SIP Stack listens.
UDP Port: The local UDP port on which the SIP Stack listens.
Reliable Provision: Support PRACK or not (100rel)
Max Call Leg: The maximum number of call-legs the SIP Stack
allocates. You should set this value to the maximum number of call you expect the SIP Stack to handle simultaneously.
Max Transaction: The maximum number of transactions the SIP Stack allocates. You should set this value to the maximum number of call you expect the SIP Stack to handle simultaneously.
Max Register Client: The maximum number of Register-Clients the SIP Stack allocates. You should set this value to the maximum number of call you expect the SIP Stack to handle simultaneously.
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Send Receive Buffer Size: The buffer size used by SIP Stack for receiving and sending SIP messages.
Reject Unsupported Extension: Yes or No
Message Pool Page Size: Used to hold and process all incoming and
outgoing message in the form of encoded messages or message objects. It is recommended that you configure the page size to the average message size your system is expected to message.
General Pool Page Size: Used by SIP Stack objects, such as call-legs and transaction, to store the internal fields. For example, the call-legs object will store the To, From and Call-ID headers and the local and the remote contact addresses on the general pool pages. The general pool is also used from other activities that demand memory allocation.
Application Pool Page Size: The size of page in the application pool
Retransmission T1: T1 determines several timers as defined in
RFC3261. For example, when an unreliable transport protocol is used, a Client Invite transaction retransmits requests at an interval that start at T1 seconds and doubles after every retransmission. A Client General transaction retransmits requests at an interval that starts at T1 and doubles until it reaches T2. (Default Value: 500)
Retransmission T2: Determines the maximum retransmission interval as defined in RFC3261. For example, when an unreliable transport protocol is used, general requests are retransmitted at an interval which starts at T1 and doubles until reaches T2. If a provisional response is received, retransmission continue but at an interval of T2. (Default Value: 4000)
Retransmission T4: T4 represents the amount of time the network takes to clear message between client and server transactions as defined in RFC3261. For example, when working with an unreliable transport protocol, T4 determines the time that UAS waits after receiving an ACK message and before terminating the transaction. (Default Value: 5000)
Invite Linger Timer: After sending an ACK for an INVITE final response, a client cannot be sure that the server has received the ACK message; the client should be able to retransmit the ACK upon receiving retransmissions of the final response for invite Linger Timer milliseconds.
General Linger Timer: After a server sends a final response, the server cannot be sure that the client has received the response message. The server should be able to retransmit the response upon receiving retransmissions of the request for general Linger Timer milliseconds. (Default Value: 32000)
Provisional Timer: When a client receives a provisional response, it continues to retransmit the request, but with an interval of provisional Timer milliseconds.
Cancel General No Response Timer: When sending a CANCEL request on a General transaction, the User Agent waits cancel General No Response Timer milliseconds before timeout termination if there is no response for the cancelled transaction.
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Cancel Invite No Response Timer: When sending a CANCEL request on an Invite transaction, the User Agent waits cancel Invite No Response Timer milliseconds before timeout termination if there is no response for the cancelled transaction.
General Request Timeout Timer: After sending a General request, the User Agent waits for a final response general Request Timeout Timer milliseconds before timeout termination (in this time the User Agent retransmits the request every T1, 2*T1,…T2,…milliseconds)
183 to Alerting: When receive a SIP 183 message from remote site, send Alerting in stead of Call Progress Indicator
Auto Send 183: WellGate5290 always send Call Progress Indicator (SIP 183) to VoIP party. It can be used for CAS protocol to enable early media.
Overload Redirect: SIP overload redirect when WellGate5290 is not able for service the call
Redirect Host: Redirect host URI (format: user@sip host, sip host)
Redirect Port: Redirect port number
Send 487 When Recv CANCEL: When receive CANCEL form remote
site, send “487 Request canceled” or not
Caller ID Mode:
o Local: use WellGate 5290 proxy user id o Caller: use SIP calling party ANI
Receive Hold music source:
o Auto: Auto determinate to play hold tone based on SIP signaling. o Local: Play holds tone locally.
On Hold music: Hold tone music file name
SIP UnRegist When Trunk Down: Enable or disable un-register when
all trunks are down. It can be used for load balancing.
SIP Over Load Reason: When all trunk are busy, the reason to send to caller. Default is 438. When work with WS6500, 438 is required for load balancing hunting.
Refer Call Use Refer By User: When a call is been transferred, new call will use REFER-BY ID instead of original caller ID.
SIP Proxy Server Setting:
WellGate 5290 can support up-to 16 SIP proxy server. You can select the SIP
proxy in call flow editor or VOIP Routing list.
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Figure 7.8-3
ID: SIP Proxy Identification
Active Mode: Active or inactive
Register Server: SIP register proxy server IP Address
Register Port: SIP register proxy server port number
Register User: SIP register proxy server User ID
Register Password: SIP register proxy server User Password
Register TTL: The registration maximum time to live setting when
registered to the SIP proxy server
Register Doman Name: SIP register proxy domain
Outbound Proxy Server: The IP address of the outbound Proxy
Outbound Proxy Port: The port of the outbound Proxy
Outbound Proxy User: The User ID of the outbound Proxy
Outbound Proxy Password: The password of the outbound Proxy

SS7 Configuration

WellGate 5290 is using Dialogic SS7 board with ISUP module. The user
manual can be downloaded from
http://resource.intel.com/telecom/support/ss7/downloads/index.htm. WellGate 5290
provides the Dialogic SS7 configuration utility to edit the Dialogic SS7 board configuration file (config.txt). After you complete the editing of config.txt, you can
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upload to the system and soft-reset to take effective. However, some advance configuration (e.g. SS7 with voice) is not supported by the Dialogic utility. User need to use the normal editor to edit it.
Start Path: ConfigurationSS7
Figure 7.9-1
SS7: Enable SS7 or not
Total Channel: read only parameters
ISUP Module ID (HEX): ISUP message module ID used for Dialogic
SS7 card and configuration (advance user only).
Tool: Get Dialogic SS7 configuration tool
Get Config: Download SS7 configuration files from 5290 for Dialogic
SS7 board
Upload: Upload the modified SS7 configuration files to 5290 for Dialogic SS7 board

Access Control

Access Control list can be used to filter the calls forms the IP Network, DNIS, and ANI. It must be used in call flow edit to take effect.
IP ACL (IP Access Control List)
Start Path: Configuration>Access Control>IP ACL
Create a new IP ACL group used for call flow editor by click the New
button.
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Figure 7.10-1
Group ID: IP ACL grouping ID Description: Description of IP ACL
IP ACL Detail
After create an IP ACL group, you can click the Detail button to add the corresponding IP list.
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Figure 7.10-2
IP Network: IP Address or prefix used to be filtered
Access Mode:
o Allow: the inputs IP Network are allowed for calls. o Disallow: The inputs IP Network are disallowed for calls.
Account Code: Account Code is part of called number prefix. It can be used for billing purpose. Account Code can be retrieved by call flow editor and make it for RADIUS billing user.
Customer Name: Name of this item
Default: If there is no Account prefix code can match, this default item
(account code) will be used
Remove Account Code: Remove Account Code from the DNIS (called number)
ANI ACL: If it is enabled, only ANI is matched the ANI ACL group can be allowed to use.
Add Prefix: Add the prefix to the DNIS (called number)
Max concurrent call: The maximum concurrent calls can be allowed for
this IP item.
Note: If in the system has both allowance and disallowance setup, the
system will check allowance first and disallowance later. If only disallowance inputted, all IP will allow to work except disallowed network. If only allowance inputted, only those IP from allowance list will work.
ANI ACL
Calling Number Screening
Start Path: Configuration>Access Control>ANI ACL
Click New to create an ANI ACL list group
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Figure 7.10-3
Group ID: ANI ACL Group
Description: Description of this ANI ACL
Click Detail to add the ANI item into the ANI ACL group.
Figure 7.10-4
Parameters:
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ANI: Calling party number used to filter
Access Mode:
o Allow: the calling numbers are allowed for calls o Disallow: The calling numbers are disallowed for calls
Note: If in the system has both allowance and disallowance setup, the
system will check allowance first and disallowance later. If only disallowance inputted all ANI will allow to work except disallowed ANI. If only allowance inputted, only those ANI from allowance list will work.
DNIS ACL
Start Path: Configuration>Access Control>DNIS ACL
Click New to add a DNIS Access Control List Group.
Figure 7.10-5
Group ID: DNIS ACL Group ID
Description: Description
Click Detail to add DNIS prefix into the IP ACL Group.
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Figure 7.10-6
Parameters:
DNIS: Called party number used for filter
Access Mode:
o Allow: The called numbers are allowed for calls o Disallow: The called numbers are disallowed for calls
Note: If in the system has both allowance and disallowance setup, the
system will check allowance first and disallowance later. If only disallowance inputted all DNIS will allow to work except disallowed DNIS. If only allowance inputted, only those DNIS from allowance list will work.

Number Replace

The purpose of “Number Replace” is to replace called number or calling number for PSTN or IP. It must be used in call flow to take effect.
Step 1: It is useful for real PSTN number to virtual VoIP number replacement.
Click Number Replace to add a new Number Replace Group, add as figure 7.8-1.
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Figure 7.11-1
Field Description:
Group ID: 1 (Number Replace Group identify)
Description: Description
Step 2: Click the New created NR and Detail button to add digits setting as
figure 7.8-2.
Figure 7.11-2
Field Description:
Original Number: Original number filter
Target Type: ANI or DNIS
Target Number: The ANI or DNIS are change to target number

Data Base Number

The purpose of Data Base Number is to change called number for PSTN or IP. It must be used in call flow to take effect.
Start Path: Configuration>Data Base Number
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Figure 7.12-1

Routing Plan

The purpose of Routing Plan is to select T1/E1 trunk and channels by your preference when there is a call from IP side to PSTN side. The PSTN
must be used in call flow edit or line hunting component to take effect.
Hunting Group
Start Path: Configuration>Routing Plan>Hunting Group
Figure 7.13-1
Parameters:
Group ID: Hunting Group ID
Description: Description of Hunting Group
Hunting Method: Route selection
o Random: Random select a trunk within this hunting group o Priority: Select a trunk by priority. (Priority 1 has lowest priority; 9
has highest priority)
o Round Robin: Call is hunting rotationally
Start Path: Configuration>Routing Plan>Hunting Group>Detail
Figure 7.13-2
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Parameters: Group: 1 Description: 0932
Interface ID: Interface ID
Trunk ID: trunk id for group 1
Priority: Trunk priority
Channel Mask: Channel mask for incoming or outgoing calls (refer
T1/E1 Trunk Configuration)
Note: When a Route Plan channels mask is cooperated to trunk
channel mask to decide the channel availity 17~31 channels are available:
Example 1: Trunk ID: 0 channel mask: 0xffffffff Route Plan channel mask: 0x0000ffff Available channel: 0x0000ffff (17~31) channels.
Example 2: Trunk ID: 0 channel mask: 0xffff0000 Route Plan channel mask: 0xffc00000 Available channel: 0xffc00000 (1~9) channels.
Call Routing The call routing can be used for hunting a PSTN trunk by prefix.
Start Path: Configuration>Routing Plan>Call Routing
Figure 7.13-3
Parameters:
Group ID: Select the T1/E1 according to the selection of Hunting Group ID when dialed number is matched
Number To Route: The dialed telephone number to be matched
Matched ANI Prefix: Calling party number used to filter
Allow Use Others: To select other T1/E1 trunk when all trunk are busy
at your desired Hunting Group. o Allowed: The call will use other T1/E1 trunks which is not belong to
the selected hunting group
o Forbad: The call will be disconnected immediately

Radius Setting

When you have an external RADIUS server to do the AAA
(Authorization, Authentication and Accounting), enter the correct parameter to the Radius setting. It must be used in call flow to take effect.
Start Path: Configuration>Radius Setting
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Figure 7.14-1
Parameters:
Auth IP: Radius Authentication Server IP address (default)
Auth Port: Radius Authentication Server Port
Acct IP: Radius Account Server IP address
Acct Port: Radius Account Server Port
Backup Auth IP: Backup Radius Authentication Server IP address
Backup Auth Port: Back Radius Authentication Server Port
Backup Acct IP: Back Radius Account Server IP address
Backup Acct Port: Back Radius Account Server Port
Secret Key: The shared secret key with RADIUS Server
Max Retry: The maximum retry times
Response Timer (sec): The maximum wait for response time from
RADIUS Server
Auth Retry Interval (sec): The internal to resend the Authentication packet to RADIUS Server.
Acct Retry Interval (sec): The internal to resend the Account packet to RADIUS Server.
Switch Threshold: Switch to alternate RADIUS Server when failures are occurred more than switch threshold.

Apply Change

1. Some of modification needs to restart system before it is effective to system operation. “Apply the change” shows “The change you mode need to
restart the system for apply please confirm to restart or do it later?”
Click on OK button to reboot the system.
Figure 7.15-1
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2. For the modification can be changed to fly, “Apply the Change” shows “Are
you sure to apply the running system?” Click on OK button to take
effecting.
Figure 7.15-2
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Chapter 8 System Control

System

Start path: Click Control>System
Figure 8.1-1
Parameter:
Soft Reset: Soft Reset at WellGate 5290
Restart: Restart the WellGate 5290
Shutdown: Shutdown the WellGate 5290
System Time Time zone Setting
Step 1: If you would like to use time zone, click Time zone button to setup the
system time zone as figure 8.2-1.
Figure 8.2-1
Standard: Step 2: Select the Standard option to setup the system predefined time zone
as figure 8.2-2
Figure 8.2-2
Parameter:
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Time Zone:
o Standard: Use a predefined standard time zone (Refer Timezone
to Country Mapping List)
o Customize: Use a user defined time zone
Auto Daylight Saving: Auto adjust daylight saving time or not
User defined time zone: Step 3: Select the Customized option and enter the time zone bias to set a
user defined time zone as figure 8.2-3
Figure 8.2-3
Parameter:
Daylight Bias: The offset added to the Bias when the time zone is in
daylight saving time
Daylight Start: The date that a time zone enters daylight time
o Month: 01 to 12 o Week Day: Sunday to Saturday o Apply Week (Day:01 to 05, Specifies the occurrence of day in the
month; 01 = First occurrence of day, 02 = Second occurrence of day, ...and 05 = Last occurrence of day)
o Hour: 00 to 23
Standard Start: The date that a time zone enters daylight time
o Month: 01 to 12 o Week Day: Sunday to Saturday o Apply Week (Day:01 to 05, Specifies the occurrence of day in the
month; 01 = First occurrence of day, 02 = Second occurrence of day, ...and 05 = Last occurrence of day)
o Hour: 00 to 23
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Network

DNS Server Setting: Step 1: After successfully logon to the system, we need to change the
network configuration. Click Control>Network to setup the network parameters as figure 8.3-1.
Figure 8.3-1
Parameter:
Primary DNS Server: Primary DNS Server IP network
Secondary DNS Server: Secondary DNS Server IP network
Host Name: Host name used to register to DNS Server
Domain Name: Domain name used to
Dynamic DNS Registration: Enable Dynamic DNS registration or not

SNMP

Start path: Click Control>SNMP>Community
Figure 8.4-1
Parameter:
Community Name: Community name for network manager system
accessing
Access Rights: Giving access right to the community
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Start path: Click Control>SNMP>Trap
Figure 8.4-2
Parameter:
Trap Community: Trap community name for NMS
Trap Host: Trap host IP address
Note: It takes around 1-minute to update SNMP configuration and
display successful message.

Prompt Manager

Start path: Click Control>Prompt Manager
Figure 8.5-1
Note:
1. You mast has a sound card in your PC to record the voice. You
need to set Network security in order to execute this recording.
Click Tool>Internet Option>Security>Custom Level.
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2. Enable the following security to active setting:
Voice prompt editor:
- Download unsigned ActiveX control: Enable
- Initialize and script ActiveX control not marked as safe: Enable
New, Record:
Step 1: Make sure you have installed microphone or other device when you
want to record, Click New and Record buttons to record as figure
8.5-2.
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Figure 8.5-2
Stop, Pause, Play:
Step 2: Click Stop or Pause button to stop record, and click Play button to
listen the voice prompt as figure 8.5-3.
Figure 8.5-3
Save:
Step 3: Click Save button to saving the voice file at local path, and the screen
shows please input the file path and file name!! (i.e. c:\irene_test.raw) as figure 8.5-4.
Figure 8.5-4
Save Remote File:
Step 4: Click Save Remote File to saving the voice file at WellGate 5290,
and the screen shows “please input the file path and file name!!” (i.e. 9999.raw) as figure 8.5-5
Figure 8.5-5
Note: The file name must be “.raw” file format.
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Open Remote File:
Step 5: Click Open Remote File button to open voice file at WellGate 5290
and screen shows Voice File List as figure 8.5-6.
Figure 8.5-6
Open:
Step 6: Click Open button to open local host voice file and screen shows
Choose File as figure 8.5-7.
Figure 8.5-7
Close:
Step 7: Click Close button to close the voice file as figure 8.5-8.
Figure 8.5-8
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Copy:
Step 8: Select the desired voice range and click Copy button as figure 8.5-9
Figure 8.5-9
Paste:
Step 9: Click Paste button to paste the voice range as figure 8.5-10.
Figure 8.5-10
Cut:
Step 10: Select the desired voice range and click Cut button as figure 8.5-11.
Figure 8.5-11
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Save As: Refer the Section “Save
Save Remote As: Refer the Section “Save Remote File
Undo:
Step 13: Click Undo button to return modification, you can see the
configuration that hasn’t be changed as figure 8.5-12.
Figure 8.5-12
Redo: Refer Section “Undo
Zoom Zoom In Zoom Out:
Step 14: Select the desired voice range click Zoom button as figure 8.5-13.
Figure 8.5-13
Step 15: The screen shows the zoom out voice file range as figure 8.5-14.
Figure 8.5-14
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Delete Remote file:
Step 16: Click Delete Remote file button to delete remote voice file as figure
8.5-15.
Figure 8.5-15

Call Flow Editor

Please refer section “Call Flow Editor

Account Manager

Please refer section “Account Manager

Upgrade

Step 1: Click “Control>Upgrade” to upgrade the software as figure 8.6-1.
Figure 8.6-1
Field Description:
File Name: Upload the software file name
Upload: Remote Upload the software at WellGate 5290
Apply: Remote apply the upload at WellGate 5290

Relogin

Please refer section “Relogin
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Chapter 9 System Monitor

It provides a way to monitor the system status.

Line Summary Status

Show channel summary status.
Start Path: Monitor>Line Summary Status
Figure 9.1-1
Field Description:
Refresh Interval (second): Refresh interval time (1, 5, 10 seconds)
Line ID: Line ID (format: Interface: trunk: channel)
Talk Time: Total conversation time
Successfully calls: Total successfully calls (connected calls)
Unsuccessfully calls: Total unsuccessfully calls (unconnected calls)
See the line detail:
Selection the line and click Detail button as figure 9.1-2.
Figure 9.1-2

Refer to line detail for field description

Line Detail

Show detail channel status.
Start Path: Monitor>Line Detail
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Figure 9.2-1
Field Description:
Refresh Interval (second): Refresh interval time (1, 5, 10 seconds)
Line ID: Line ID
Line Status: Current time status
Call Originate: Call originate site
ANI String: Calling party number
DNIS String: Called party number
PSTN Status: PSTN site status
VoIP Status: IP site status
Escape Time: Talk time

System Status

Show current VoIP and Trunk status
Start Path: Monitor>System Status
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Figure 9.3-1
Field Description:
Green light indicates OK.
Yellow light indicates unused or D channel down
Red light indicates failure

Event Log

Show system log status.
Start Path: Monitor>Event Log
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Figure 9.4-1
Field Description:
Date: Event created date
Time: Event created time
Event ID: Event Log
Type: Event Log type
o Information o Warring o Error
Source: Executable program
Description: Description
Note: You can click Clear button to clear all event log.
See the detail event log:
Double click the log or select the log and click detail to see the log
detail.
Figure 9.4-2
Event Description:
Event ID Event Description Description
8003
[GK]: [xxx.xxx.xxx.xxx:xxxx] not found or registered failure [SIP Register]: [xxx.xxx.xxx.xxx:xxxx] not found or registered failure
Failed to register to H323 Gatekeeper
Failed to registered to SIP Registrater Server
TRUNK ALARM: TrunkId=x,
8700
Comment='', LOS=1, LOF=0, RAI=0,
Disconnect to PSTN
AIS=0, RAI_CRC=0
8702 BOARD x TRUNK(S) FOUND Board x Trunk(s) found
[0]: evt: D CHANNEL STATUS:
8703
runkId=3, Status=1, Comment='', LOS=14, LOF=0, RAI=108,
D Channel and Trunk ID (ID: 0) not available
AIS=145, RAI_CRC=-1
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