Planet Technology Network Card VIP 2100 User Manual

1-port H.323/SIP E1/T1

Trunk Gateway

VIP-2100

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DIGIT MANIPULATION .................................................................................... 50

CALL FLOW EDITOR...................................................................................... 50

CONFIGURATION MANAGER ........................................................................... 50

APPLY CHANGE............................................................................................ 51

Chapter 7 Advance Configuration Reference .......................................... 52

CONFIGURATION........................................................................................... 52

System Configuration............................................................................. 52

Interface Configuration........................................................................... 53

Dial Plan Configuration........................................................................... 55

T1/E1 Trunk Configuration...................................................................... 55

Rest Configuration.................................................................................. 59

H.323 Configuration ............................................................................... 59

SIP Configuration ................................................................................... 62

Access Control ....................................................................................... 68

Number Replace .................................................................................... 71

Routing Plan........................................................................................... 71

Radius Setting........................................................................................ 73

Apply Change......................................................................................... 75

Chapter 8 System Control ......................................................................... 76

SYSTEM....................................................................................................... 76

SYSTEM TIME............................................................................................... 76

NETWORK.................................................................................................... 78

SNMP ........................................................................................................ 78

PROMPT MANAGER ...................................................................................... 79

CALL FLOW EDITOR...................................................................................... 85

ACCOUNT MANAGER..................................................................................... 85

UPGRADE .................................................................................................... 85

RELOGIN ..................................................................................................... 86

Chapter 9 System Monitor........................................................................ 87

LINE SUMMARY STATUS ................................................................................ 87

LINE DETAIL ................................................................................................. 88

EVENT LOG.................................................................................................. 88

DEBUG INFO ................................................................................................ 90

PING ........................................................................................................... 90

Chapter 10 Telnet & RS-232 Configuration............................................... 91

LOGON VIP-2100 BY TELNET........................................................................ 91

Chapter 11 LCD Display Configuration..................................................... 97

Appendix 1 Call Flow Example................................................................ 100

ONE STAGE DIALING (GATEKEEPER MODE) .................................................. 100

ONE STAGE DIALING (SIP PROXY MODE)..................................................... 102

ONE STAGE DIALING (PEER TO PEER MODE)................................................ 104

TWO STAGE DIALING (VOIP, PSTN MIXED CALL) .......................................... 106

TWO STAGE DIALING WITH AAA (IP SIDE AAA) ............................................ 109

Appendix 2 Java plug-in Installation ...................................................... 113

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Appendix 4 Interface LED Description .................................................. 116

Appendix 5 Build-in Voice Prompt Index................................................ 117

Appendix 6 Timezone to Country Mapping List...................................... 118

Appendix 7 IP Bandwidth Requirement................................................... 120

Appendix 8 Release Complete Cause Code............................................ 121

Appendix 9 RADIUS Format Attributes.................................................... 124

Appendix 10 Quick Start Check List....................................................... 128

VIP-2100 FAQ ............................................................................................. 132

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Chapter 1 VIP-2100 Introduction

System Description

VIP-2100 is a cost effective solution for VoIP trunk gateway supporting one-,

port T1/E1 VoIP trunks that provides voice and fax over IP network. It

supports ITU-T H.323 V3, SIP RFC 2543/3261, SNMP V2, Call Detail Record,

WEB management and other useful functions to meet customer requirements.

The built-in enhanced IVR (Interactive Voice Response) and Billing Service

of VIP-2100 is suitable for prepaid and postpaid service. It can rapidly provide

value added service for customers.

VIP-2100 Features:

Dual SIP/H.323 co-existing

ITU-T H.323 v3 and H.450 compliance

SIP RFC 2543/3261 standard compliance

PSTN signaling: ISDN/PRI, CAS (MFC R2, MFC R1, E&M), QSIQ

Mixed SIP, Gatekeeper and P2P calls

Support H.323 Gatekeeper register, direct and route calls

Support SIP outbound proxy, redirect and register server

Redundant SIP Proxy/Outbounbd Proxy Server Support (Outbond

Active/Active fail over, Register A/A no fail over)

Support SIP Overload Redirect

SIP supplemental service - on Hold, Call Transfer (Transferred)

Built-in phone book and prefix routing for SIP and H.323 P2P calls

Support H.323 fast connect, early H.245 and H.245 tunneling

Support H.323 and SIP early media

VoIP to VoIP calls support – SIP to H.323, SIP to SIP, H.323 to H.323

Global Trunk-Channel Block out: 0xffffffff (busy block out)

Intelligent PSTN call routing and in-trunk hunting: reverse rotary, channel

mask (default:0xffffffff), ANI prefix match

Reset a channel/trunk on the fly

Flexible digit manipulation plan

Support RADIUS Authentication, Authorization and Accounting

Support access control by ANI, DNIS, IP, Gatekeeper only, proxy only or

RADIUS

SIP UDP/TCP support

Behind NAT friendly for SIP calls

Inbound and out of band DTMF transmission

SIP/H.323 T.38 fax relay up to 14400 bps

Dynamic call treatment based on DNIS, ANI or collected DTMF

Grouping DNIS/ANI Number Replacement

Built-in IVR & call-flow controller for PSTN / VoIP side

CISCO compatible

Web-based graphic announcement edit and management

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Multiple configuration saving

Provides CDR (Call Detail Record)

Built-in internal user authentication for prepaid & postpaid users

Technical Specification

Interface

- Two 10/100MB Ethernet Ports (Host & VoIP stream)

- 1 xT1/E1 (120 Ohm-RJ48C connectors)

75 Ohm needs external 3rd party BNC/RJ-48C adapter cables

Protocol and Standard

- ITUT H.323 v3 and H.450 compliance

- SIP RFC 2543/3261 compliance

Audio Feature

- Codec -- G.711 A/μ-Law, G.723.1 (5.3K/6.3K), G.729A, G.729

- Support G.168 echo cancellation

- Configurable audio payload size & adaptive jitter buffer

- Support silent suppression for G.729A, G.723, G.729

- VAD (Voice Activity Detection)

- CNG (Comfort Noise Generation)

DTMF Transmission

- Transparent

- H.245 signal/alphanumeric

- H.323 Q.931

- RFC 2833

- 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

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- Selected disconnect cause code & behavior

Management Feature

- OS and program upgradeable

- 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 ^linesx 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, FCC, EMI

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VIP-2100 Detail Specifications

Feature

VIP-2100

Physical Dimension

1 Width

2 Height

3 Depth

4 Industrial rack mount

5 Color

483mm

44mm

450mm

Yes

Black

8Kg

6 Weight

Power / Environmental

1 Power

2 Operating temperature

3 Relative humidity

90-240V auto switch

0~60 C

5%~95%

Processors & Storage

1 DSP vendor

Intel Pentium, AudioCodes DSP

2 Operation System

3 RAM

XP Embedded

512 MB

4 Program/Data Storage

5 OS Upgradeable

7 Program Upgradeable

256 MB DOM

Yes

Front Panel Display

1 LED status

2 LCD status

Power/DOM/System

Yes

LAN Interface

10/100MB Ethernet ports *2 (host &

RTP)

1 10/100 Base Ethernet

2 IP Address Required

PSTN Interface

Customizable E1/T1 CAS

E1 CAS DTMF

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

E1 CAS R2 MF

Euro, Australia, Hong Kong, Korea,

New Zealand, QSIC

E1 ISDN PRI Support

5 E1/T1 Interface

PCM law Support

Selectable

Alaw/Mulaw selectable

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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

T1 ISDN PRI Support

Start(ANI B4 ADDR), E&M Feature

Group D Wink Start, E&M

Immediate Start, E&M Wink Start,

GroundStart FXO, GroundStart

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

Trunk Spans

1 (T1/E1s) per chassis

Default Trunk Channel Mask

PSTN Line Hunting

PSTN Line Hunting Channel Selection

On the Fly Reset Channel/Trunk

Yes

Audio Codec Support

1 G.711 A-law

Yes

2 G.711 u-law

Yes

3 G.723.1

4 G.729A

Yes (5.3/6.3K)

Yes

5 Selectable Payload Size - G.711

6 Selectable Payload Size - G.723

7 Selectable Payload Size - G.729

20, 40, 60 ms

30, 60, 90 ms

20, 40, 60 ms

Fax Transmission

1 Bypass mode

Yes

2 CISCO Compatible

3 ECM Support

Yes

4 FAX auto-detection

5 H.323 Annex D Support

6 SIP- T.38 Reinvite

7 T.38 During fast connect

8 T.38 Redundant Packet

9 Transparent mode

10 Up to G3 FAX

Yes

0-2

Yes

Yes (up to 14400 bps)

DTMF Transmission

1 RFC 2833

2 H.245 Alphanumeric mode

3 H.245 Signal mode

Yes

4 Q.931 UUI

5 SIP INFO

6 Transparent mode

Voice Quality & Echo Cancellation

1 Adaptive Jitter Buffer

2 CNG

Yes

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3 G.168 (Echo Cancellation)

Yes (32ms)

Yes

Gain Control

5 Improved Echo Tail Suppression

6 Silence Suppression

Yes

VAD

Yes

Maintenance

1 Administrative Log

2 Auto Daylight Saving

3 Customizable Time Zone

4 Front Panel LCD Setup

5 FTP Server

Yes

6 HTTP server

Yes

7 HTTP SSL support

8 Multiple configuration

9 NTP time synchronization

10 Password Security

11 RS232

Yes

Yes (SNTP V4)

Yes

12 System Event Log

13 Telnet

Yes

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

Yes

2 Fixed IP

Yes

3 DNS

Yes

4 Dynamic DNS

Yes

5 Ping

Yes

6 TOS field setting

7 SNMP V2 MIB I & II

8 SNMP get command

9 SNMP set command

10 SNMP Trap

Yes (RTP only)

Yes

11 H.341 MIB Support

12 SysLog Support

Yes

H.323 Protocol Support

1 H.323 V3

2 H.323 ID

Yes

3 E.164 ID

Yes

Yes (selectable for

incoming/outgoing)

4 Fast Connect

5 H.450

6 H.245 Tunneling

7 Early H.245

Yes

8 Cause Code Mapping

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SIP Protocol Support

1 Cause Code Mapping

2 HTTP Digest Authentication

3 SIP Call on Hold

4 SIP Early Media

Yes

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

Yes

3 Routed call

4 Light weight RRQ

5 IRQ: IRR sequence

6 Gatekeeper Call only

SIP Proxy Sever Support

1 SIP Outbound Proxy Support

2 SIP Redirect Server Support

3 SIP Registar Server Support

4 Redundant SIP Proxy Server

5 Auto Fail Over

Yes

Dial Plan

1 P2P H.323/SIP Call

2 GK Call

Yes

3 SIP Call

Yes

4 PSTN Call

Yes

5 Mixed SIP, P2P, GK call

6 Build-in Phone Book

7 P2P Prefix Routing

8 Digits Manipulation

9 ISDN Dial Plan by Prefix

Yes

Yes (Source & Destination)

Call Type Support

1 Call Decision

Dynamic Decided by Call Flow

2 H.323 to H.323 Call

3 H.323 to H.323 Fax Realy

4 H.323 to PSTN Call

5 H.323 to SIP Call

Yes

6 H.323 to SIP FAX Relay

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7 H.323 to SIP FAX Relay

7 PSTN to H.323 Call

8 PSTN to PSTN Call

9 PSTN to SIP Call

Yes

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

Yes

Yes (Random, Round Robin,

Priority)

6 Intelligent PSTN Call Routing

7 In-trunk hunting method

Cyclic, random, rotary, reverse

cyclic, reverse rotary

8 Ring Back Tone Generation

9 Call Progress Tone Support

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

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

Yes (Chinese & English)

Yes

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

3 RADIUS Accounting

4 Redundant RADIUS Server Support

5 PSTN Prepaid Support

Yes

Yes, Active/Standby/Auto Failover

Yes

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6 VoIP Prepaid Support

Yes

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

Yes

System Limitation

1 Max DM

4096

2048

4096

20000

10000

256

2 Max IP ACL

3 Max DNIS ACL

4 Max ANI ACL

5 Max User ACL

6 Max Phone Book Entries

7 Max Call Flow Component

8 Max CDR Keep Days

9 Max Voice File Storage

10 hours

Manual

1 English User Guide

Yes

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VIP-2100 Appearance Description

VIP-2100 Front Panel:

Functions:

1: Power LED

2: Network1 Interface LED

3: Network2 Interface LED (not used)

4: H/D LCD

5: Power Switch

6: System Status LED

7: LCD Panel

8: LCD Touch Panel

VIP-2100 Rear Panel:

Functions:

1: Electric Fan

2: AC Power outlet

3: AC Power switch (Keep on)

4: Trunk E1/T1 port

5: VoIP Ethernet port

6: Keyboard/Mouse

7: Com1 port

8: Ethernet port

9: VGA

10: print port (not available)

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Chapter 2 Logon VIP-2100

After connected E1/T1 & Ethernet cables into the VIP-2100, 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 setup VIP-2100, 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 VIP-2100

Setp1: Start IE5.0 (or later version) to navigate VIP-2100 Management

System by typing the default IP address (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

255.255.0.0

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

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Step 3: The screen shows the Home Page of VIP-2100 as figure 2.1-3.

Figure 2.1-3

Network Configuration

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 2.2-1.

Figure 2.2-1

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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-

Figure 2.2-3

☻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

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.

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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.

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, user role and description, 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 VIP-2100 as figure

2.5-2.

Figure 2.5-2

Step 3: The screen shows the Home Page of VIP-2100 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

H.323 GK and SIP Proxy Mode

SIP Phone

H323 VoIP

Network

Gatekeeper

SIP VoIP

Network

Proxy Server

H.323

Gateway

H323 Phone

SIP USB Phone 8888

Phone 1001

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

(0932111222)

SIP → DNIS (11382265699) → DM (SIP_in_drop) → Make Call

(82265699)

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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.

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 Mulaw

☻Caution: Subnet Mask does not support Supernet.

Step 4: After successfully to 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-

Figure 3.2-2

Step 3: Modify the trunk parameters (i.e. Trunk Type, Termin Side, Trunk

Mode, Protocol Tag, Line Code) and apply the change by clicking

Apply as figure 3.2-3.

Figure 3.2-3

Frequency Changed Parameters:

Trunk Type: E1 or T1

Termin Type: User Side or Network Side

Trunk Mode: Normal

Protocol Tag: ISDN protocol used

Line Code: T1 or E1 line code used

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Step 4: After modifications are made to the Trunk Configuration, the screen

comes back the page of Trunk Configuration as figure 3.2-4.

Figure 3.2-4

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:

Gatekeeper IP: 192.168.5.1

E.164 Tel: 113

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Step 3: You can see the screen display the new configuration of the H.323

Configuration as figure 3.3-3.

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.

Figure 3.4-1

Frequency used parameters:

SIP Register: Yes

Primary Registar Server: 192.168.19.150

Primary Registar Port: 5060

Primary Registar User: 173

Primary Registar Password: 173

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Primary Outbound Proxy Server: 192.168.19.150

Primary Outbound Proxy Port: 5060

Primary Outbound Proxy User: 173

Primary Outbound Password: 173

Step 3: You can see the screen display the new configuration of the SIP

Configuration as figure 3.4-2.

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

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Step 2: Enter the related parameters and click Apply button as figure 3.5-2.

Figure 3.5-2

Field Description:

Group ID: 0 (DM Group identify)

Description: H.323: H323 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-H323 In Drop (belong to this DM group)

Drop: 5 (drop digits)

H.323 incoming call

↓

Dialed number: 582265699

↓

Match the pattern 5

↓

Delete 5 (Drop)

↓

New dialed number becomes 82265699

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: 1-SIP In Drop (belong to this DM group)

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Drop: 113 (drop digits)

SIP incoming call

↓

Dialed number: 11307688222

↓

Match the pattern 11307

↓

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: 0282265699 (pattern to be matched)

Group ID: PSTN In Drop (belong to this group id)

Drop: 0282265699 (drop digits)

PSTN incoming call (DNIS mode)

↓

Dialed number: 02822656991001

↓

Match the pattern 0282265699

↓

Delete 0282265699 (Drop)

↓

New dialed number becomes 1001

☻Note: Digit Manipulation have to tapped for PSTN Side (Trunk→

Outbound/Inbound DM Group), VoIP Side (VoIP→

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Outbound/Inbound DM Group) or Call Flow (refer to section “Call

Flow Editor”) to take effect.

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

Load Call Flow: Load call flow from VIP-2100

Save: Save a call flow in VIP-2100

Cut: Cut a component

Copy: Copy a component

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

Show Component Table: Show all component table

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Step 2: Drag and prop the required component icon into the workspace as

figure 4-2.

Figure 4-2

Right click the component to bring up the component propriety to

setup parameter:

AAA: Send Authorization or Authentication for validation

o Type: AAA type selection

- Authorization: Send RADIUS Authorization packet out

- Authentication: Send RADIUS Authentication packet out

Success To: Success to component

Failed other to: Failed to component

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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

J Note: Detail response attributes, please refer RADIUS Format

Attributes

Answer: Answer incoming call (PSTN only)

Branch: Play an announcement and branch into different route

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

CDV: Collected Digit Validation

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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 Invaried To: If the checked variable is not success to validate

CIB: Call Information Branch

o Info Type: Information type selection

ANI: Calling Number

DNIS: Called Number

IP: IP Address or network (e.g. 192.168.0.0)

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 infor type to be validation

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-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

Cut Rule: Cut a system variable into different parts

o Cut From: Cut start digit from (start from 1)

o Cut To: Cut end digit to

o Assign To: Store the cutted result into

Disconnect: Disconnect the call

DM: Digit Manipulation

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o DM Parameter: Manipulation ANI or DNIS

o DM Group ID: Apply to DM group

MakeCall: Make Call to PSTN or H.323/SIP site

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 VIP-2100 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

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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

-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

PB: Play Balance for prepaid purpose

o Voice File: Voice prompt file

o Language: Play balance language section

-English

-Chinese.

o Interrupted: Voice can be interrupted or not

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PCUI: Prompt and Connect User Information

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

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

o Success To: If fine an available channel by system setup (call

hunting)

o Failed To: If not fine an available channel by system setup (call

hunting)

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Set Data: Assign value to a variable

o Assign To: Assigned variable

o Use SysParam: Use system parameter to replace or not

o Value: ANI, DNIS, User ID or other digits

Start: Call flow start

o Next Component

Quit: Disconnect calls

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Example Call Flow as figure 4-3.

Example Description:

Components

Contents

Start

Component ID:

1000

Next Component: 1001

PSTN To: 1011

H.323 To: 1009

SIP To: 1008

CTB

Component ID:

1001

Info Type: ANI

Prefix: 1 goto: 1010 (H.323 GK call)

Prefix: 8 goto: 1004 (SIP Proxy call)

CIB

Component ID:

1011

1011 Route for PSTN call

Route Mode: Gatekeeper

Finish To: 1005

Failed Other To: 1005

MakeCall

Component ID:

1010

Route Mode: SIP Proxy Call

Finish To: 1005

Failed Other To: 1005

MakeCall

Component ID:

1004

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Next Component: 1006

Disc

Component ID:

1005

Quit

Component: 1006

1001 Route for H.323Gatekeeper call

DM Parameter: DNIS

DM Group ID: H.323 In Drop

Next Component: 1007

Component ID:

1009

Route Mode: PSTN

Finish To: 1005

Failed Other To: 1005

MakeCall

Component ID:

1007

Next Component: 1006

Disc

Component ID:

1005

Quit

Component: 1006

1001 Route for SIP Proxy call

DM Parameter: DNIS

DM Group ID: SIP In Drop

Next Component: 1007

Component: 1008

Route Mode: PSTN

Finish To: 1005

Failed Other To: 1005

MakeCall

Component ID:

1007

Next Component: 1006

Disc

Component ID:

1005

Quit

Component: 1006

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Example Used Call Flow:

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Start:1000

CTB:1001

PSTN to:1011

H323 to: 1009

SIP to: 1008

1011

PSTN in

to 1009

H.323 in

to 1008

SIP in

CIB: 1011

DM: 1009

DM: 1008

Info Type: DNIS

Prefix: 1 goto: 1010

Prefix: 8 goto: 1004

Call to PSTN

Make Call: 1010

Make Gatekeeper

call to H.323

Make Call: 1004

Make SIP Proxy

call to SIP

Make Call: 1007

Call to PSTN

Success /Failed to

Finish to

Failed other to

Disc: 1005

Disconnect

Success /Failed to

Quit: 1006

Disconnect

Configuration Manager

Configuration Management provides a way to save and reload the

system configuration for future use.

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Load a Configuration:

Step 1: When you need to load a saved configuration, click a saved

configuration (i.e. 04/26/2004 Loading Test) item to load it back as figure

4.1-1.

Figure 3.7-1

Step 2: When screen shows “Current configuration will lost! Are you sure

to load this configuration?” click on OK button to load he saved

configuration to the working configuration as figure 4.1-1.

Figure 3.8-2

JNote: It is need to restart the system to take effect of the new-loaded

working configuration.

Save the working Configuration:

Step 3: To save the current configuration, select a new created configuration

and click Save button, when screen shows “Description”, please

enter the configuration description (i.e. Billing Test) for the saved

configuration as figure 4.2-2.

Figure 3.8-3

Step 4: You can see the screen display the changes as figure 4.2-4.

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Figure 3.8-4

Backup the working configurations:

Step 5: 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

4.2-2.

Figure 3.8-5

Restore configuration:

Step 6: 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 4.2-2.

Figure 3.8-6

Compact the database file:

Step 7: In order to optimize the system performance, you can optional

compact the database by click Compact button as figure 4.1-2.

Figure 3.8-7

J 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 4.3-1

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Chapter 5 Peer to Peer Mode Configuration

Environment used in this chapter

Peer to Peer

SIP Phone

H323 VoIP

Network

SIP VoIP

Network

H.323

Gateway

H323 Phone

SIP USB Phone 8888

Phone 1001

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 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”

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H.323 Configuration

Step 1: Change Register To Gatekeeper to “No” to enable peer to peer

mode as

figure 5.1-1.

Figure 5.1-1

Frequency used parameters:

SIP Configuration

Step 1: Change SIP Register to “No” to enable peer to peer mode as figure

5.2-1.

Figure 5.2-1

Frequency used parameters:

Primary SIP Register: No

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Address Book

For making a Peer-to-Peer call, the IP device must has an address

record in the phone book for routing.

Step 1: Click Address Book adds a new address book for the peer to peer

calls, New to add as figure 5.3-1.

Figure 5.3-1

Step 2: Enter the related parameters and click Apply button as figure 5.3-2.

Figure 5.3-2

Field Description:

Name: H.323 IP Phone or SIP-Cisco

Tel/Prefix: 1002

Trans Address:

- H.323 Call: 192.168.5.102 or 192.168.5.102:1720

- SIP Call: sip:[email protected] or sip:[email protected]:5060 or

sip:[email protected]

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Step 3: You can see the screen displays the new Address Book as figure 5.3-

Figure 5.3-3

☻Note: You must apply the change to take effect for the change.

Digit Manipulation

Please refer to section “Digit Manipulation”

Call Flow Editor

Please refer to section “Call Flow Editor”

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Call Flow (P2P Mode):

Start:1000

CTB:1001

PSTN to:1011

H323 to: 1009

SIP to: 1008

1011

PSTN in

to 1009

H.323 in

to 1008

SIP in

DM: 1013

Call to PSTN

DM: 1009

DM: 1008

Call to PSTN

Make Call: 1004

Make Peer to Peer

call to SIP

Make Call: 1007

Call to PSTN

Success /Failed to

Finish to

Disc: 1005

Failed other to

Disconnect

Quit: 1006

Disconnect

Configuration Manager

Please refer to section “Configuration Manger”

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Apply Change

Please refer to section “Apply Change”

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Chapter 6 SIP to H.323 Mode Configuration

Environment used in this chapter

SIP to H.323

SIP Phone

H323 VoIP

Network

SIP VoIP

Network

H.323

Gateway

H323 Phone

SIP USB Phone 8888

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”

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”

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Address Book

Please refer to section “Address Book”

Digit Manipulation

Please refer to section “Digit Manipulation”

Call Flow Editor

Please refer to section “Call Flow Editor”

Call Flow (P2P Mode):

Start:1000

CTB:1001

1009

H.323 in

PSTN:1010

H323: 1009

SIP: 1008

1010

PSTN in

1008

SIP in

Make Call: 1002

Make Peer to Peer

Call to SIP / H323

Make Call: 1010

Make Gatekeeper

call to H.323

Finish to

Failed other to

Disc: 1003

Disconnect

Success to

Quit: 1004

Quit

Configuration Manager

Please refer to section “Configuration Manger”

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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 Keepdays: CDR system keeping days

Hot Swappable: Hot swappable support (reserved)

First Digit Timeout: The max to time (in second) waits for receiving the

first digit entered (5~20 sec).

Inter Digit Timeout: The max to time (in second) waits for the between

two digits (5~20 sec).

Debug Level:

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 hitted.

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Time Expired Notify: Seconds to be notifying caller before

communication expired. This function is used for Pre-Paid calling card

service and must cooperate with RADIUS Server.

Almost Expired Tone: Communication expired notice tone selection

Fast Response Timeout: The maximum times to wait for response. It’s

depended on the network speed.

No Answer Timeout: The maximum the (in second) to wait the remote

party answer (pick up phone)

o Notify Tone#1:

o Notify Tone#2:

Authentication Mode: Authentication by VIP-2100 or RADIUS

o Internal: Authentication building User ACL

o External: Authentication by RADIUS

o Ext. AAA Failure Opt: Bypass or disconnect incoming calls when

external

Version: 5.1

Interface Configuration

Start Path: Configuration→Interface

Figure 7.2-1

Basic Parameter Description:

Interface ID: System parameter

Card slot: System parameter

Interface Type: System parameter

Description: System parameter

Serial No: System parameter

License Key: System parameter

IP Address: IP address used for voice RTP stream

Subnet Mask: Submask (doesn’t support super class)

Default Gateway: Default gateway for routing

PCM Type: PCM type encoding, E1 A-law; T1 u-law

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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 Internetwork 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

PCM Idle Pattern: This pattern will be sending on each B channel PCM

time slot when the channel is idle (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

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voice quality. Set to 12 will have long voice delay but with better voice

quality

EC Tail Length: Echo Cancellation Length, default value is 25ms

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

Dial Plan Configuration

Dial Plan can be used to assign the ISDN number plan based on prefix

setting.

Start Path: Configuration→Interface→Dial Plan

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

ApplyTo: Trunks apply to

T1/E1 Trunk Configuration

Start Path: Configuration→Interface→Trunk

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Figure 7.3-1

Basic Parameter Description:

Interface ID: System parameter

Trunk ID: System parameter

Trunk Type: T1or E1 selection

Description: Description for this trunk ID

Termin Side: Network site or User Site (normally, you set to “user site”

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 H.323 incoming only: Allow the H.323 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

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

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- 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

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

Advance Trunk Configuration:

Start Path: Configuration→Interface →Trunk →Advance

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Figure 7.3-2

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

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 IP to PSTN side (default: 0 db)

Output Gain: Voice Gain from PSTN to IP side (default: 0 db)

Q.931 General Opt.: used for Q.931 general behavior.

o 0x0001: No Status message send for unknown facility IE if it is set

o 0x0002: No Status message send for invalid content of a valid

facility IE if it is set

o 0x0080: Send Connect Ack message when receive Connect

message if it is set, you can OR the required option together

Q.931 Incoming Opt.: used for Q.931 incoming call behavior

o 0x0800: include Channel-ID IE in the first reply message (e.g. Call

Proceeding or Alerting)

o 0x2000: enable the system to include Channel-ID IE in the Call

Proceeding message, you can OR the required option together

Q.931 Outgoing Opt.: used for Q.931 outgoing behavior

o 0x0010: 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

Trans Cap: Transfer Capability

o Voice Service

o Data Service

o Modem Service

CallID Transfer Type: Call ID transfer type

o Disable Caller ID: default parameter

o Transparent Caller ID

o Relay Caller ID

o Bypass Caller ID

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Rest Configuration

Reset a channel or a trunk idle state.

Start Path: Configuration→Interface→Detail→Reset

Figure 7.4-1

Start Path: Configuration→Interface→Detail→Reset

Figure 7.4-2

Basic Parameter Description:

Trunk: Reset trunk ID

Channel: Rest channel selection

- All Channel: Reset all channel

- 0~31: Reset 0~30 logical channel to reset

H.323 Configuration

Start Path: Configuration→H.323

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Figure 7.5-1

Basic Parameter Description:

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

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)

o FRF11 Fax Relay (Propriety mode): FRF11 Fax Relay (propriety

mode)

Fast Connect Mode: Connection of H.323 call fast mode

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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

H.450 Service: Enable the H.450 calls transfer service

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 7.5-2

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

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)

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G.711 Psize: G.711 transmission packet size in ms (default: 20ms)

SIP Configuration

Start Path: Configuration→SIP

Figure 7.6-1

Basic Parameter Description:

Primary SIP Register: Register to SIP proxy server or not

o Yes: Register to proxy server

o No: Not register to proxy server

Primary Register Server: SIP register proxy server IP Address

Primary Register Port: SIP register proxy server port number (default:

1719)

Primary Register User: SIP register proxy server User ID

Primary Register Password: SIP register proxy server User Password

Primary Register TTL: The registration maximum time to live setting

when registered to the SIP proxy server

Secondary SIP Register: Register to SIP proxy server or not

o Yes: Register to proxy server

o No: Not register to proxy server

Secondary Register Server: SIP register proxy server IP Address

Secondary Register Port: SIP register proxy server port number

(default: 1719)

Secondary Register User: SIP register proxy server User ID

Secondary Register Password: SIP register proxy server User

Password

Secondary Register TTL: The registration maximum time to live setting

when registered to the SIP proxy server

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Primary Outbound Proxy Server: The IP address of an outbound Proxy

the SIP Stack uses.

Primary Outbound Proxy Port: The port of an outbound Proxy the SIP

Stack uses

Primary Outbound Proxy User: The User ID of an outbound Proxy the

SIP Stack uses.

Primary Outbound Proxy Password: The password of an outbound

Proxy the SIP Stack uses.

Secondary Outbound Proxy Server: The IP address of an outbound

Proxy the SIP Stack uses.

Secondary Outbound Proxy Port: The port of an outbound Proxy the

SIP Stack uses

Secondary Outbound Proxy User: The User ID of an outbound Proxy

the SIP Stack uses.

Secondary Outbound Proxy Password: The password of an outbound

Proxy the SIP Stack uses.

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

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

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Figure 7.6-2

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.

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

your 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

your 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 your expect the SIP Stack to handle simultaneously.

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 from 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.

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Application Pool Page Size: The size of page in the application pool

Retransmission T1: T1 determines several timer 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 inviteLingerTimer 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 generalLingerTimer milliseconds.

(Default Value: 32000)

Provisional Timer: When a client receives a provisional response, it

continues to retransmit the request, but with an interval of

provisionalTimer milliseconds.

Cancel General No Response Timer: When sending a CANCEL

request on a General transaction, the User Agent waits

cancelGeneralNoResponseTimer milliseconds before timeout

termination if there is no response for the cancelled transaction.

Cancel Invite No Response Timer: When sending a CANCEL request

on a Invite transaction, the User Agent waits

cancelInviteNoResponseTimer 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 generalRequestTimeoutTimer

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

AutoSend 183: VIP-2100 always send Call Progress Indicator (SIP 183)

to VoIP party. It can be used for CAS protocol to enable early media.

Behind NAT: Does VIP-2100 is located behind NAT or not

Public Signal IP: The static mapped IP for SIP signal

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Public Signal Port: The static mapped Port for RTP stream

Public RTP IP: The static mapped RTP IP

Public RTP Port: The static mapped RTP starting port

Public RTP Port Interval: The VIP-2100 has at least 30 RTP channels.

Each channel needs 3 ports mapping on NAT Server. The interval is

used to caculate the right port for each channel.

Overload Redirect: SIP overload redirect when VIP-2100 is not able for

service the call

Redirect Host: Redirect host URI (format: user@siphost, siphost)

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 VIP-2100 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 hold tone locally.

On Hold music: Hold tone music file name

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Behind NAT Example 1:

VIP-2100

NAT Server Setting

210.59.163.11

One-by-One Static

IP Mapping

192.168.111.112

Static Port Mapping

192.168.111.111:5060

210.59.163.10:10000

VIP-2100 NAT Enable Setting:

Public Signal IP: 210.59.163.10

Public Signal Port: 10000

Public RTP IP: 210.59.163.11

Public RTP base port: 4000 (same as “Interface→Advance’s Config”)

Public RTP Port Interval: 10

Behind NAT Example 2:

VIP-2100

192.168.111.111:5060

192.168.111.112:4000

4001

NAT Server Setting

210.59.163.10:5060

210.59.163.10:10000

10001

Static Port Mapping

RTP Channel 01

4002

10002

RTP Channel 02

192.168.111.112:4010

4011

210.59.163.10:10003

10004

4012

10005

RTP Channel 30

192.168.111.112:4310

210.59.163.10:10357

10358

4311

4312

10359

VIP-2100 NAT Enable Setting:

Public Signal IP: 210.59.163.10

Public Signal Port: 5060

Public RTP IP: 210.59.163.10

Public RTP base port: 10000 (same as “Interface→Advance’s Config”)

Public RTP Port Interval: 0

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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

Start Path: Configuration→Access Control→IP ACL

Figure 7.7-1

Parameters:

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.

☻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

Start Path: Configuration→Access Control→ANI ACL

Figure 7.7-2

Parameters:

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.

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DNIS ACL

Start Path: Configuration→Access Control→DNIS ACL

Figure 7.7-3

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.

User ACL

User ACL is used to store subscriber information when internal AAA is

enabled.

Start Path: Configuration→Access Control→User ACL

Figure 7.7-4

Parameters:

User: User ID (0~9, *#)

Password: Password (0~9, *#)

Prepaid Point: Allowed prepaid point (When prepaid point is used, the

system will deduct it automatically base on the rate setting.)

o Postpaid: postpaid user

Status:

o Active: User is activeled

o Inactive: User is inactived

☻Note: 1. IP Authentication method must be set to “ internal AAA” to

talk effect.

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New a Calling Rate: The calling rate will have different appearance for

different calling rate policy set in Radiu configuration.

Click Calling Rate button to add a new calling rate as figure 7.7-5.

Figure 7.7-5

Point per Second calling rate:

Calling rate (point per second) is used to convert prepaid point into

prepaid time in second. For example, you can set calling rate to 5 for “100”

prefix. When a caller, which has 200 prepaid point, calls “100xxxx”, the max

talk time will be 200/5=40 seconds. If a calling rate is set to “0”, it means free

charge.

New a Calling Rate (Second per Point):

Click Calling Rate button to add a new calling rate as figure 7.7-6.

Figure 7.7-6

Second per Point calling rate:

. Calling rate (Second per point) is used to convert prepaid point into

prepaid second in time. For example, you can set calling rate (Second) to 6,

charge point to 1 for “113” prefix. It means that every 6 seconds charge 1

point. When a caller, which has 200 prepaid point, calls “113xxxx”, the max

talk time, will be 200*6/1=1200 seconds.

J Note: Tel prefix * is used as a default rate, you need to create it to

work.

Search Condition:

You can search a user by User ID, Prepaid or Postpaid condition as figure

7.7-7.

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Figure 7.7-7

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.

Figure 7.8-1

Field Description:

Group ID: 1 (Number Replace Group identify)

Description: SIP in

Step 2: Click the New created NR and Detail button to add digits setting as

figure 7.8-2.

Figure 7.8-2

Field Description:

Original Number: Original number filter

Target Type: ANI or DNIS

Target Number: The ANI or DNIS are change to target nubmer

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

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Figure 7.9-1

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 7.9-2

Parameters:

Group: 4 Description: FET Trunk 02 and 03

Interface ID: Interface ID

Trunk ID: trunk id for group 4

Priority: Trunk priority

Channel Mask: Channel mask for incoming or outgoing calls (refer

T1/E1 Trunk Configuration)

J 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.

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Call Routing

The call routing can be used for hunting a PSTN trunk by prefix.

Start Path: Configuration→Routing Plan→Call Routing

Figure 7.9-3

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 7.10-1

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 Time (sec): The maximum wait for response time from

RADIUS Server

Auth Retry Interval (sec): The internal to resend the Authentication

packet to RADIUS Server.

Acc 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.

Auto Inactive: Auto inactive an unused or not

o Disable: Don’t auto inactive

o Prepaid User: Auto inactive prepaid user only

o Postpaid User: Auto inactive postpaid user only

o All User: Auto inactive all unused user

Inactive prepaid: The minimum credit point threshold for a prepaid user

to be inactived

Inactive Period: The max unaccess days for a postpaid user to be

inactived

Charge Method: Billing charge method selection

o Point per Second: Point / calling rate = seconds

o Second per Point: Point * calling rate / charge point = seconds

Auto Clean: Auto clean the inactive user

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o Disable: Don’t auto clean inactive user

o Prepaid User: Auto clean prepaid user only

o Postpaid User: Auto clean postpaid user only

o All User: Auto clean inactive user

Clean Filter: Auto clean filter

o None: Auto clean users exceed clean period without access the

network

o Inactive: Auto clean only to inactive users

Clean Period: The maximum unaccess days to clean up. When the

clean filter is set Inactive, the unaccess day is start counting when the

user is inacived

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 7.11-1

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 7.11-2

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Chapter 8 System Control

System

Start path: Click Control→System

Figure 8.1-1

Parameter:

Soft Reset: Soft Reset at VIP-2100

Restart: Restart the VIP-2100

Shutdown: Shutdown the VIP-2100

System Time

Timezone Setting

Step 1: If you would like to use timezone, click Timezone button to setup the

system timezone as figure 8.2-1.

Figure 8.2-1

Stardand:

Step 2: Select the Standard option to setup the system predefined time zone

as figure 8.2-2

Figure 8.2-2

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Parameter:

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 Daylinght Saving: Auto adjust daylinght saving time or not

User defined timezone :

Step 3: Select the Customized option and enter the time zone bias to set a

user defined timezone 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)

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:

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Community Name: Community name for network manager system

accessing

Access Rights: Giving access right to the community

Start path: Click Control→SNMP→Trap

Figure 8.4-2

Parameter:

Trap Community: Trap community name for NMS

Trap Host: Trap host IP address

JNote: It takes around 1-minute to update SNMP configuration and

display successful message.

Prompt Manager

Start path: Click Control→Prompt Manager

Figure 8.4-1

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☻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.

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

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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

8.4-2.

Figure 8.4-2

Stop,

Pause,

Play:

Step 2: Click Stop or Pause button to stop record, and click Play button to

listen the voice prompt as figure 8.4-3.

Figure 8.4-3

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 8.4-4.

Figure 8.4-4

Save Remote File:

Step 4: Click Save Remote File to saving the voice file at VIP-2100, and the

screen shows “please input the file path and file name!!” (i.e.

9999.raw) as figure 8.4-5

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Figure 8.4-5

☻Note: The file name must be “ .raw” file format.

Open Remote File:

Step 5: Click Open Remote File button to open voice file at VIP-2100 and

screen shows Voice File List as figure 8.4-6.

Figure 8.4-6

Open:

Step 6: Click Open button to open local host voice file and screen shows

Choose File as figure 8.4-7.

Figure 8.4-7

Close:

Step 7: Click Close button to close the voice file as figure 8.4-8.

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Figure 8.4-8

Copy:

Step 8: Select the desired voice range and click Copy button as figure 8.4-9

8.4-9

Paste:

Step 9: Click Paste button to paste the voice range as figure 8.4-10.

Figure 8.4-10

Cut:

Step 10: Select the desired voice range and click Cut button as figure 8.4-11.

Figure 8.4-11

Save As: Refer the Section “Save”

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Save Remote As: Refer the Section “Save Remote File”

Undo:

Step 13: Click Undo button to return modification, you can see the

configuration that haven’t be changed as figure 8.4-12.

Figure 8.4-12

Redo: Refer Section “Undo”

Zoom

Zoom In

Zoom Out:

Step 14: Select the desired voice range click Zoom button as figure 8.4-13.

Figure 8.4-13

Step 15: The screen shows the zoom out voice file range as figure 8.4-14.

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Figure 8.4-14

Delete Remote file:

Step 16: Click Delete Remote file button to delete remote voice file as figure

8.4-15.

Figure 8.4-15

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 7.5-1.

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Figure 7.5-1

Field Description:

File Name: Upload the software file name

Upload: Remote Upload the software at VIP-2100

Apply: Remote apply the upload at VIP-2100

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

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Line Detail

Show detail channel status.

Start Path: Monitor→Line Detail

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

Event Log

Show system log status.

Start Path: Configuration→Event Log

Figure 9.3-1

Field Description:

Type: Event Log type

o Information

o Warring

o Error

Date: Event created date

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Time: Event created time

Source: Executable program

Category: Event type (none, welltech Sys…)

Event ID: Event Log

☻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 9.3-2

Event Description:

Event ID

Event Description

Description

[GK]: [xxx.xxx.xxx.xxx:xxxx] not found or

registered failure

Failed to register to H323 Gatekeeper

8003

[SIP Register]: [xxx.xxx.xxx.xxx:xxxx] not found

or registered failure

Failed to registered to SIP Registratar

Server

8700

8703

VoIP Gateway application on the fly change

On the fly change (system change)

[0]: evt: D CHANNEL STATUS: runkId=3,

Status=1, Comment='', LOS=14, LOF=0,

RAI=108, AIS=145, RAI_CRC=-1

D Channel and Trunk ID (ID: 0) not available

9500

9501

9502

Gateway application started

AAA Mgr application started

TelnSvr application started

VoIP Board (0) started

VoIP Gateway program start

AAA Manager program start

Telnet Server program start

Interface (ID:0) start

H323 stack started

H323 stack start

H323 GK [xxx.xxx.xxx.xxx:xxxx] found &

registered

[SIP Register]: [xxx.xxx.xxx.xxxx] Found &

Registered.

Registered to H323 Gatekeeper

9503

Registered to SIP Registratar Server

Connect to PSTN

9504

9505

9600

PSTN trunk (0) alarm clear

[0]: evt: D CHANNEL STATUS: TrunkId=3,

Status=0, Comment='', LOS=29, LOF=67,

RAI=31, AIS=1, RAI_CRC=-1

D Channel and Trunk ID (ID:0) available

Failed / Success to connect SNTP server

SNTP client application started

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Debug Info

Start Path: Click “Monitor→Debug Info”

Figure 9.4-1

Get Log: Get debug log (-1~999)

Search: Search debug logs

Clear: Clear log

Filed Description:

Ping

You can use the “Ping” to check an IP is active or not.

Start Path: Configuration→Ping

Figure 9.5-1

Field Description:

Host IP Address: The IP address to ping

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Chapter 10 Telnet & RS-232 Configuration

VIP-2100 also can support to be managed by Telnet or Console port

(RS-232) for basic operations.

Interface:

v Network: TCP/IP Telnet

v RS232:

- Connect using: COM1

- Baud Rate: 9600

- Data bits: 8

- Parity: None

- Stop bits: 1

- Flow Control: None

- Wire: Null modem line (crossed)

Logon VIP-2100 by Telnet

Use Windows build-in Hyper Terminal or other telnet terminal emulator

to login (e.g. telnet 192.168.111.111:10086). User ID & password will be

required for login (default login user id: admin, password: admin & user id:

root, password: root).

Command List:

Command

Description

echo

Auto echo on or off

eventlog

exit

ipconfig

ping

reboot

reset

Clean or show system log message

Quit the current session

Configure or show network information

Check an IP address is available or not

Reboot

Soft-reset

shutdown Shutdown

time

timezone

Reset or show system time.

Setup or show system timezone

useradmin Manage user account.

help & ? View command list

Echo: auto echo on or not

Command

Purpose

Usage: echo on/off

Example: echo on

[root#]echo ?

[root#]echo on

[root#]echo off

Echo is on

Echo is off (default ralue)

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Eventlog: show system log message

Command

Purpose

Usage: eventlog [-clear]

Example: eventlog

eventlog -clear

[root#]eventlog ?

[root#]eventlog

Show system eventlog:

Eventlog example:

Time: 2003-06-19 20:15:17

Type: Warning

Event ID: 8700

Source : wellgate5x00

Description: [0]: evt: TRUNK ALARM: TrunkId=3

Time: 2003-06-19 20:15:17

Type: Warning

Event ID: 8700

Source : wellgate5x00

Description: [0]: evt: TRUNK ALARM: TrunkId=2

Time: 2003-06-19 20:15:14

Type: Information

Event ID: 9501

Source : wellgate5x00

Description: [0]: evt: BOARD STARTED: SLOT:8

Press any key to continue or press 'Q' to quit

[root#]eventlog -

clear

Clear all event log

Exit: Quit the current session

Command

Purpose

[root#]exit

Quit the current session

Ipconfig: Configuration or show network information

Command

[root#] ipconfig ?

Purpose

Usage: ipconfig [-delete dns] [-dhcp] [-dns IPAddress1

IPAddress2 ] [-ip IPAddress -mask Mask -gateway

Gateway]

Example : ipconfig -ip 192.168.111.111 -mask 255.255.0.0

-gateway 192.168.1.254

: ipconfig -dhcp

: ipconfig -dns 192.168.1.1

: ipconfig -delete dns

[root#]ipconfig

Show current network configuration

USE FIXED IP (or DHCP)

IP Address

: 192.168.5.113

: 255.255.0.0

: 192.168.1.254

: 192.168.5.1

168.95.1.1

Subnet Mask

Default Gateway

DNS Servers

[root#]ipconfig –delete

dns

Delete the DNS servers setting

USE FIXED IP

IP Address

: 192.168.5.113

: 255.255.0.0

: 192.168.1.254

Subnet Mask

Default Gateway

DNS Servers

[root#]ipconfig –dchp

Enable DHCP

USE DHCP

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IP Address

: 192.168.5.10

: 255.255.0.0

: 192.168.1.254

: 192.168.5.1

Subnet Mask

Default Gateway

DNS Servers

168.95.1.1

[root#]ipconfig –ip

61.220.126 28 –mask

255.255.0.224 –gateway

61.220.126.1

Use fixed network configuration

USE FIXED IP

IP Address

: 61.220.126.28

: 255.255.255.1

: 61.220.126.254

Subnet Mask

Default Gateway

DNS Servers

[root#]ipconfig –ip

61.220.126.115

Changes IP address only.

USE FIXED IP

IP Address

: 61.220.126.115

: 255.255.255.1

Subnet Mask

Default Gateway : 61.220.126.254

DNS Servers

[root#]ipconfig –dns

210.59.126.53

Changes DNS configuration only.

USE FIXED IP

IP Address

: 61.220.126.115

: 255.255.255.1

Subnet Mask

Default Gateway : 61.220.126.254

DNS Servers : 210.59.126.53

Ping: Check an IP address is available or not

Command

Purpose

[root#] ping ?

Usage: ping IP.

Example: ping 127.0.0.1

[root#]ping 61.220.126.1

Ping result

Reply from 61.220.126.1 bytes=64 time=1ms TTL=29

Reboot:

Command

Purpose

[root#] reboot ?

Reboot System

Are You Sure? (Y/N)

[root#]reboot

VIP-2100 are rebooting

Are You Sure?(Y/N)y

Shutdown:

Command

Purpose

[root#] shutdown ?

Shutdown System

Are You Sure? (Y/N)

[root#]shutdown

VIP-2100 are shutting down

Are You Sure?(Y/N)y

Reset:

Command

Purpose

[root#] reset ?

Soft reset System

Are You Sure? (Y/N)

[root#]reset

Are You Sure?(Y/N)y

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Time: Reset or show system time

Command

Purpose

Usage : time YYYY-MM-DD HH:NN:SS

Example : Time 2002-01-01 12:00:00

[root#] time ?

[root#]time

Show current time

The current time is 2003-06-20 15:17:30

[root#]time 2003-07-29

23:14:53

Change system bios time

Timezone: Setup or show system timzone

Command

Purpose

Fixed Zone List:

01. Afghanistan Standard Time

03. Arab Standard Time

[root#] timezone ?

02. Alaskan Standard Time

04. Arabian Standard Time

06. Atlantic Standard Time

08. AUS Eastern Standard Time

10. Canada Central Standard

Time

05. Arabic Standard Time

07. AUS Central Standard Time

09. Azores Standard Time

11. Cape Verde Standard Time

13. Cen. Australia Standard Time 12. Caucasus Standard Time

15. Central Asia Standard Time

17. Central European Standard

Time

19. Central Standard Time

21. Dateline Standard Time

23. E. Australia Standard Time

25. E. South America Standard

Time

14. Central America Standard

Time

16. Central Europe Standard

Time

18. Central Pacific Standard

Time

20. China Standard Time

22. E. Africa Standard Time

24. E. Europe Standard Time

26. Eastern Standard Time

28. Ekaterinburg Standard Time

30. FLE Standard Time

32. Greenland Standard Time

34. GTB Standard Time

36. India Standard Time

38. Israel Standard Time

40. Mexico Standard Time

42. Mid-Atlantic Standard Time

44. Myanmar Standard Time

46. Nepal Standard Time

48. Newfoundland Standard

Time

27. Egypt Standard Time

29. Fiji Standard Time

31. GMT Standard Time

33. Greenwich Standard Time

35. Hawaiian Standard Time

37. Iran Standard Time

39. Korea Standard Time

41. Mexico Standard Time 2

43. Mountain Standard Time

45. N. Central Asia Standard

Time

47. New Zealand Standard Time

49. North Asia East Standard

Time

51. Pacific SA Standard Time

53. Romance Standard Time

55. SA Eastern Standard Time

57. SA Western Standard Time

59. SE Asia Standard Time

61. South Africa Standard Time

63. Taipei Standard Time

65. Tokyo Standard Time

67. US Eastern Standard Time

69. Vladivostok Standard Time

71. W. Central Africa Standard

Time

50. North Asia Standard Time

52. Pacific Standard Time

54. Russian Standard Time

56. SA Pacific Standard Time

58. Samoa Standard Time

60. Singapore Standard Time

62. Sri Lanka Standard Time

64. Tasmania Standard Time

66. Tonga Standard Time

68. US Mountain Standard Time

70. W. Australia Standard Time

72. W. Europe Standard Time

74. West Pacific Standard Time

73. West Asia Standard Time

75. Yakutsk Standard Time

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Usage1 : timezone Zone (1 to 75) AutoDaylight (Y or N)

Example1 : timezone 1 Y

Usage2 : timezone -custom Bias DaylightBias DaylightStart

StandardStart

Bias

: -12:00 to +13:00

DaylightBias : -12:00 to +13:00

DaylightStart :

MM (Month: 01 to 12) ;

WD (Day of week: 00 to 06)

DD (Day:01 to 05 ;Specifies the occurrence of day in the

month;

01 = First occurrence of day,

02 = Second occurrence of day, ..., 05 = Last occurrence of

day

HH (Hour:00 to 23)

StandardStart :

MM (Month: 01 to 12) ;

WD (Day of week: 00 to 06)

DD (Day:01 to 05 ;Specifies the occurrence of day in the

month;

01 = First occurrence of day,

02 = Second occurrence of day, ..., 05 = Last occurrence of

day

HH (Hour:00 to 23)

Example2 : timezone -custom +08:00 -01:00 04-00-01-02 10-00-

05-02

[root#]timezone

Show current timezone info

Time Zone

: (40) Mexico Standard Time (GMT -06:00)

Daylight Bias : -01:00

Daylight Start : 05-00-01 02:00

Standard Start : 09-00-05 02:00

Auto Daylight : Y

[root#]timezone 40 n Use pre-defined timezone

Time Zone

: (40) Mexico Standard Time (GMT -06:00)

Daylight Bias : -01:00

Daylight Start : 05-00-01 02:00

Standard Start : 09-00-05 02:00

Auto Daylight : n

[root#]timezone -

custom +08:00 -

01:00 05-00-01-03

09-00-05-03

Use customized timezone

Time Zone

: (99) Customized (GMT 08:00)

Daylight Bias : -01:00

Daylight Start : 05-00-01 03:00

Standard Start : 09-00-05 03:00

Auto Daylight : Y

Please refer Timezone to Country Mapping List

Useradmin: Manager user account

Command

Purpose

Usage: useradmin [-add User] [-delete User] [-

password User]

[root#] useradmin ?

Example: useradmin -add irene

[root#]useradmin

Show the current login user account

root

[root#]useradmin -list

Show the current user account list

rdmin

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root

irene

[root#] useradmin -add irene Add the new user account: irene

Password : irene

Confirm : irene

Add user Success.

[root#] useradmin -delete

1111

Delete the user: 1111

Are You Sure?(Y/N)y

[root#] useradmin -password Change the user: root’s password.

root

New Password : 1234

Confirm : 1234

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Chapter 11 LCD Display Configuration

VIP-2100 provides a front panel LCD for basic operations.

Button List:

Button List

Description

When the VIP-2100 is ready, the LCD screen shows as blow

Ready |

04-03-03

16:40

Press Enter to select command

Enter

Event Log

IP Config

ESC

▲

▼

Quit the current command

Up or previous edit mode

Next or previous edit mode

Command Tree:

Main Menu

Event Log

Show system log message

Show IP Info

Use DHCP

IP Config

Use Fixed IP

Yes

Reboot

Yes

Reset PWD

Soft Reset

Shut Down

Yes

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Event Log:

Configure

LCD Display

▲

▼

Enter

▲

Previous event log

Next event log

Show detail event log

Previous line

▼

Next line

ESC

Quit detail event log viewing

Quit to main menu

IP Config:

Configure

LCD Display

▲

▼

Enter

▲

Select Network configuration

Configure Network

Increase the digit apply to network setting

Decrease the digit apply to network setting

Apply change to network information

Quit network setting

▼

Enter

ESC

Quit to main menu

Reboot:

Configure

LCD Display

Select Reboot or not

▲

▼

Select Reboot or not

Enter

ESC

Reset user: root’s (or admin) user password

Quit Reboot configure

Quit to main menu

Reset:

Configure

LCD Display

Select user to change password

Change user password

Increase the alphabet apply to user password setting

Decrease the alphabet apply to user password setting

Quit Reset configure

▲

▼

Enter

▲

▼

ESC

Quit to main menu

Soft Reset:

Configure

LCD Display

Select Reset or not

▲

▼

Select Reset or not

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Enter

ESC

Reset or not

Quit Reset configure

Quit to main menu

Shutdown:

Configure

LCD Display

▲

▼

Enter

ESC

Select Shutdown or not

Shutdown or not

Quit Shutdown configure

Quit to main menu

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Appendix 1 Call Flow Example

One Stage Dialing (Gatekeeper Mode)

Example Description:

Components

Contents

Start

Component ID:

1000

Next Component: 1001

PSTN To: 1004

H.323 To: 1007

SIP To: 1005

CTB

Component ID:

1001

1007 Route for H.323Gatekeeper call

Route Mode: PSTN

Finish To: 1005

Failed Other To: 1005

MakeCall

Component ID:

1007

Reason: PSTN normal call clear

Next Component: 1006

Disc

Component ID:

1005

Quit

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Component: 1006

1004 Route for PSTN call

Route Mode: Gatekeeper

Finish To: 1005

Failed Other To: 1005

MakeCall

Component ID:

1004

Next Component: 1006

Disc

Component ID:

1005

Quit

Component: 1006

Example Used Call Flow:

Start:1000

CTB:1001

PSTN to:1004

H323 to: 1007

SIP to: 1005

1007

H.323 in

1004

PSTN in

Make Call: 1007

Call to PSTN

1005

SIP in

Make Call: 1004

Make Gatekeeper

call to H.323

Finish to

Disc: 1005

Success to

Failed other to

Disconnect

Quit: 1006

Disconnect

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One Stage Dialing (SIP Proxy Mode)

Example Description:

Components

Contents

Start

Component ID:

1000

Next Component: 1001

PSTN To: 1004

H.323 To: 1005

SIP To: 1007

CTB

Component ID:

1001

1007 Route for SIP Proxy call

Route Mode: PSTN

Finish To: 1005

Failed Other To: 1005

MakeCall

Component ID:

1007

Reason: PSTN normal call clear

Next Component: 1006

Disc

Component ID:

1005

Quit

Component ID:

1006

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1004 Route for PSTN call

Route Mode: SIP Proxy Call

Finish To: 1005

Failed Other To: 1005

MakeCall

Component ID:

1004

Next Component: 1006

Disconnect

Component ID:

1005

Quit

Component ID:

1006

Example Used Call Flow:

Start:1000

CTB:1001

PSTN to:1004

H323 to: 1005

SIP to: 1007

1007

SIP in

1004

PSTN in

Make Call: 1007

Call to PSTN

1005

H.323 in

Make Call: 1004

Make SIP Proxy call

to SIP

Finish to

Disc: 1005

Success to

Failed other to

Disconnect

Quit: 1006

Disconnect

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One Stage Dialing (Peer to Peer Mode)

Example Description:

Components

Contents

Start

Component ID:

1000

Next Component: 1001

PSTN To: 1004

H.323 To: 1007

SIP To: 1007

CTB

Component ID:

1001

1007 Route for SIP Proxy or H.323 Gatekeeper call

Route Mode: PSTN

Finish To: 1005

Failed Other To: 1005

MakeCall

Component ID:

1007

Next Component: 1006

Disc

Component ID:

1005

Quit

Component ID:

1006

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1004 Route for PSTN call

Route Mode: P2P Call

Finish To: 1005

Failed Other To: 1005

MakeCall

Component ID:

1004

Next Component: 1006

Disc

Component ID:

1005

Quit

Component ID:

1006

Example Used Call Flow:

Start:1000

CTB:1001

PSTN to:1004

H323 to: 1007

SIP to: 1007

1007

H.323/ SIP in

1004

PSTN in

Make Call: 1007

Call to PSTN

Make Call: 1004

Make Peer to Peer

call to SIP and H.323

Finish to

Disc: 1005

Success to

Failed other to

Disconnect

Quit: 1006

Disconnect

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Two Stage Dialing (VoIP, PSTN mixed call)

Example Description:

Components

Contents

Call route from PSTN to IP Side

Start

Component ID:

1000

Next Component: 1001

PSTN To: 1002

H.323 To: 1006

SIP To: 1006

CTB

Component ID:

1001

1001 route for SIP Proxy and H.323 Gatekeeper call

Info Type: DNIS

CIB

Component ID:

1006

Prefix: 5 goto: 1010

Prefix: 7 goto: 1008

Other goto: 1003

Route Mode: Gatekeeper Call

Finish To: 1004

Failed Other To: 1004

MakeCall

Component ID:

1010

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Route Mode: SIP Proxy Call

Finish To: 1004

Failed Other To: 1004

MakeCall

Component ID:

1008

Route Mode: PSTN

Finish To: 1004

Failed Other To: 1004

MakeCall

Component ID:

1003

Next Component: 1005

Disc

Component ID:

1004

Quit

Component ID:

1005

1001 Route for PSTN call

Route Mode: SIP Proxy Call

Finish To: 1004

Failed Other To: 1004

MakeCall

Component ID:

1002

Next Component: 1005

Disc

Component ID:

1004

Quit

Component ID:

1005

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Example Used Call Flow:

H.323 & SIP in

Start:1000

CIB 1006 Type DNIS

Prefix: 5 to 1010

Prefix: 7 to 1008

Other to 1003

CTB 1001

SIP in to 1006

H.323 in to 1006

PSTN in to1002

PSTN in

Other

Make Call: 1003

Call to PSTN

Make Call: 1008

Make Call: 1010

Make SIP Proxy

call to SIP

Make Gatekeeper

call to H.323

Finish to

Failed other to

Finish to

Failed other to

Make Call: 1002

Make SIP Proxy

call to SIP

Finish to

Failed other to

Disc: 1004

Disconnect

Finish to

Failed other to

Quit: 1005

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Two Stage Dialing with AAA (IP Side AAA)

Example Description:

Components

Contents

Start

Component ID:

1000

Next Component: 1001

PSTN To: 1012

H.323 To: 1013

SIP To: 1019

CTB

Component ID:

1001

1012 Route for PSTN call

Route Mode: SIP Proxy

Finish To: 1013

Failed Other To: 1013

MakeCall

Component ID:

1012

Next Component: 1014

Disc

Component ID:

1013

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Quit

Component ID:

1014

1012 Route for H.323 call

Next Component: 1014

Disc

Component ID:

1013

Quit

Component ID:

1014

1012 Route for SIP call

Next Component: 1015

Anser

Component ID:

1019

Assign to: User ID / Password

Use SysParam: Yes

Value: ANI

Next Component: 1016

Type: Authentication

Prepaid User to: 1017

Postpaid User to: 1009

Failed to: 1020

Set Data

Component ID:

1015

AAA

Component ID:

1016

Failed Reason:

- Invalid Account

- Account InUse

- Zero Balance

- Account Expired

- Over Credit Limit

- Number of Retries Exceeded

- Insufficient Balance

- Route for prepaid user call

Voice File: 0004.raw

Language: English

Interrupted: No

Next Component: 1018

Play Type: Voice or dial tone

Voice File: 0001.raw

Max DTMF: 30

Result Append To: DNIS

End of DTMF: #

Component ID:

1017

PCUI

Component ID:

1018

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Next Component: 1006

Type: Authorization

Success to: 1021

Failed to: 1022

Failed Reason:

AAA

Component ID:

1006

- Invalid Account

- Account InUse

- Zero Balance

- Account Expired

- Over Credit Limit

- Number of Retries Exceeded

- Insufficient Balance

Voice File: 0004.raw

Language: English

Interrupted: No

Component: 1021

Next Component: 1010

Route Mode: PSTN Call

Finish To: 1013

MakeCall

Failed Other To: 1013

Component ID:

1010

- Route for failed user call

Dynamic Play: Disable

Voice File: 0005.raw

Language: English

Interrupted: No

Component: 1022

Next Component: 1013

Next Component: 1014

Disc

Component ID:

1013

Quit

Component ID:

1014

- Route for postpaid user call

Route Mode: SIP Proxy Call

Finish To: 1013

Failed Other To: 1013

MakeCall

Component ID:

1012

Next Component: 1014

Disc

Component ID:

1013

Quit

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Component ID:

1014

Example Used Call Flow:

Start:1000

CTB: 1001

PSTN to: 1012

H.323 to: 1013

SIP to: 1019

SIP in

Answer: 1019

Set Data: 1015

AAA: 1016

1025

PSTN in

Type: Authentication

Prepaid User to: 1021

Postpaid to: 1009

Prepaid User

Failed to

Failed to: 1020

PB: 1021

Voice: You have ..balance

xxx dollars ..

PA: 1020

Voice: User ID or

Postpaid User

Password is invalid,

Please try later.

PCUI: 1018

Voice: Please enter

distination

PCUI: 1018

Voice: Please enter

distination

H.323 in

Make Call: 1012

SIP Proxy

Make Call: 1010

PSTN

AAA: 1006

Type: Authorization

Success to: 1017

Failed to: 1022

Failed to

call to SIP

call to PSTN

PA: 1022

Voice: no balance,

please contact your

sales...

PD: 1021

Voice: the call can go…

xx minutes and xx seconds

Success/ Failed to

Make Call: 1010

PSTN

call to PSTN

Success/ Failed to

Disc: 1013

Disconnect

Success/ Failed to

Quit: 1014

Quit

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Appendix 2 Java plug-in Installation

You need to install Java Plug-in before using call flow editor, prompt

manager and upgrade. Please confirm you JRE version is 1.4.1_02 or above

if your PC has already installed Java.

After downloaded the java runtime version (1.3.1 or later) from Sun, you

just follow the wizard to install the Java runtime. When you see the display

shows “Select Browsers”, do not select any option item, press Next button

to continue.

You also need to set newer versions of stored pages. Click Tool→Internet

Option→General→Setting.

After success, restart your browser to take effect.

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Appendix 3 Retrieve CDR Information

J Retrieve method example (stop20040305.log) by ftp:

C:\>ftp 192.168.19.117

Connected to 192.168.19.117.

220 Server ready

User (192.168.19.117:(none)): root

331 Password required for root.

Password:

230 User root logged in.

ftp> cd planet\cdr

250 CWD command successful. "D:/planet/cdr/" is current directory.

ftp> dir

200 Port command successful.

150 Opening data connection for directory list.

drw-rw-rw- 1 ftp

-rw-rw-rw- 1 ftp

226 File sent ok

ftp

0 Mar 06 00:02 .

0 Mar 06 00:02 ..

ftp 53998192 Mar 05 23:57 STOP20040305.log

ftp 20222855 Mar 05 23:50 STRT20040305.log

ftp: 403 bytes received in 0.25Seconds 1.61Kbytes/sec

ftp> bin

200 Type set to I.

ftp> lcd

Local directory now C:\.

ftp> get stop20040305.log

200 Port command successful.

150 Opening data connection for stop20040305.log.

226 File sent ok

ftp: 20222855 bytes received in 4.43Seconds 4569.10Kbytes/sec.

ftp>bye

221 Goodbye

Billing Start CDR:

File name: STRTyyyymmdd.log

Field delimit: ,

Field description:

NAS-IP-Address : VoIP gateway IP address

NAS-Port-Type

Asynchronous

User-Name

: (Network Access Server Port Type)

: User ID

Calling-Station-Id

Acct-Status-Type

Service-Type

Gateway-Name

Conf-ID

: Calling station number

: Message type (1: start)

: 1: login

: VoIP gateway aliases

: GUID

Call-Type

: Telephony or VOIP

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Call-Originate

Setup-Time

: originate or answer

: Call initiate time (UTC time)

Acct-Session-Id

Acct-Delay-Time

: N/A

Billing Stop CDR:

File name: STOPyyyymmdd.log

Field delimit: ,

Field description:

NAS-IP-Address

NAS-Port-Type

: VoIP gateway IP address

: (Network Access Server Port Type)

Asynchronous

User-Name

: User ID

Called-Station-Id

Calling-Station-Id

Acct-Status-Type

Service-Type

: Called station number

: Calling station number

: Message type (1: Start , 2: Stop)

: 1: login

Gateway-Name

Conf-ID

: VoIP gateway aliases

: GUID

Call-Type

Call-Originate

Setup-Time

Connect-Time

Disconnect-Time

Disconnect-Cause

Voice-Quality

: Telephony or VOIP

: originate or answer

: Setup Time (UTC time)

: Connect Time (UTC time)

: Disconnect Time (UTC time)

: Disconnect cause code

: Voice Quality

Gateway-ID

: Remote gateway IP address

: N/A

: Talk time

: N/A

Acct-Session-Id

Acct-Input-Octets

Acct-Output-Octets

Acct-Input-Packets

Acct-Output-Packets

Acct-Session-Time

Acct-Delay-Time

Charge rate

: Internal AAA prepaid user charge rate

: Internal AAA prepaid user available

Available Balance

balance

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Appendix 4 Interface LED Description

Interface Real Panel:

Ethernet LED:

Trunk LED:

Trunk RJ48 Wiring:

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Appendix 5 Build-in Voice Prompt Index

File Name

Description

0001.raw

0002.raw

0003.raw

0004.raw

0005.raw

Please enter the destination

Please enter your user ID

Please enter your password

You have

User ID or password is invalid. Please try later.

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Appendix 6 Timezone to Country Mapping List

Greenwich Mean Time & Country List

(GMT-12:00) International Date Line West

(GMT-11:00) Midway Island, Samoa

(GMT-10:00) Hawaii

Time Zone

21. Dateline Standard Time

58. Samoa Standard Time

35. Hawaiian Standard Time

02. Alaskan Standard Time

(GMT-09:00) Alaska

(GMT-08:00) Pacific Time (US & Canada);

Tijuana

52. Pacific Standard Time

(GMT-07:00) Mountain Time (US & Canada)

(GMT-07:00) Chihuahua, La Paz, Mazatlan

(GMT-07:00) Arizona

43. Mountain Standard Time

41. Mexico Standard Time 2

68. US Mountain Standard Time

10. Canada Central Standard Time

(GMT-06:00) Saskatchewan

(GMT-06:00) Guadalajara, Mexico City,

Monterrey

40. Mexico Standard Time

(GMT-06:00) Central Time (US & Canada)

(GMT-06:00) Central America

(GMT-05:00) Indiana (East)

(GMT-05:00) Eastern Time (US & Canada)

(GMT-05:00) Bogota, Lima, Quito

(GMT-04:00) Santiago

19. Central Standard Time

14. Central America Standard Time

67. US Eastern Standard Time

26. Eastern Standard Time

56. SA Pacific Standard Time

51. Pacific SA Standard Time

57. SA Western Standard Time

06. Atlantic Standard Time

(GMT-04:00) Caracas, La Paz

(GMT-04:00) Atlantic Time (Canada)

(GMT-03:30) Newfoundland

(GMT-03:00) Greenland

48. Newfoundland Standard Time

32. Greenland Standard Time

55. SA Eastern Standard Time

25. E. South America Standard Time

42. Mid-Atlantic Standard Time

11. Cape Verde Standard Time

09. Azores Standard Time

(GMT-03:00) Buenos Aires, Georgetown

(GMT-03:00) Brasilia

(GMT-02:00) Mid-Atlantic

(GMT-01:00) Cape Verde Is.

(GMT-01:00) Azores

(GMT) Greenwich Mean Time: Dublin, Edinburgh,

Lisbon, London

31. GMT Standard Time

(GMT) Casablanca, Monrovia

33. Greenwich Standard Time

(GMT+01:00) West Central Africa

71. W. Central Africa Standard Time

(GMT+01:00) Sarajevo, Skopje, Warsaw, Zagreb 17. Central European Standard Time

(GMT+01:00) Brussels, Copenhagen, Madrid,

53. Romance Standard Time

Paris

(GMT+01:00) Belgrade, Bratislava, Budapest,

16. Central Europe Standard Time

Ljubljana, Prague

(GMT+01:00) Amsterdam, Berlin, Bern, Rome,

72. W. Europe Standard Time

Stockholm, Vienna

(GMT+02:00) Jerusalem

38. Israel Standard Time

(GMT+02:00) Helsinki, Kyiv, Riga, Sofia, Tallinn,

Vilnius

30. FLE Standard Time

(GMT+02:00) Harare, Pretoria

(GMT+02:00) Cairo

61. South Africa Standard Time

27. Egypt Standard Time

(GMT+02:00) Bucharest

24. E. Europe Standard Time

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(GMT+02:00) Athens, Istanbul, Minsk

(GMT+03:00) Nairobi

34. GTB Standard Time

22. E. Africa Standard Time

(GMT+03:00) Moscow, St. Petersburg, Volgograd 54. Russian Standard Time

(GMT+03:00) Kuwait, Riyadh

(GMT+03:00) Baghdad

03. Arab Standard Time

05. Arabic Standard Time

37. Iran Standard Time

(GMT+03:30) Tehran

(GMT+04:00) Baku, Tbilisi, Yerevan

(GMT+04:00) Abu Dhabi, Muscat

(GMT+04:30) Kabul

12. Caucasus Standard Time

04. Arabian Standard Time

01. Afghanistan Standard Time

73. West Asia Standard Time

28. Ekaterinburg Standard Time

(GMT+05:00) Islamabad, Karachi, Tashkent

(GMT+05:00) Ekaterinburg

(GMT+05:30) Chennai, Kolkata, Mumbai, New

Delhi

36. India Standard Time

(GMT+05:45) Kathmandu

46. Nepal Standard Time

(GMT+06:00) Sri Jayawardenepura

(GMT+06:00) Astana, Dhaka

(GMT+06:00) Almaty, Novosibirsk

(GMT+06:30) Rangoon

62. Sri Lanka Standard Time

15. Central Asia Standard Time

45. N. Central Asia Standard Time

44. Myanmar Standard Time

50. North Asia Standard Time

59. SE Asia Standard Time

63. Taipei Standard Time

(GMT+07:00) Krasnoyarsk

(GMT+07:00) Bangkok, Hanoi, Jakarta

(GMT+08:00) Taipei

(GMT+08:00) Perth

70. W. Australia Standard Time

60. Singapore Standard Time

49. North Asia East Standard Time

(GMT+08:00) Kuala Lumpur, Singapore

(GMT+08:00) Irkutsk, Ulaan Bataar

(GMT+08:00) Beijing, Chongqing, Hong Kong,

Urumqi

20. China Standard Time

(GMT+09:00) Yakutsk

75. Yakutsk Standard Time

39. Korea Standard Time

(GMT+09:00) Seoul

(GMT+09:00) Osaka, Sapporo, Tokyo

(GMT+09:30) Darwin

65. Tokyo Standard Time

07. AUS Central Standard Time

13. Cen. Australia Standard Time

69. Vladivostok Standard Time

64. Tasmania Standard Time

74. West Pacific Standard Time

08. AUS Eastern Standard Time

23. E. Australia Standard Tim

(GMT+09:30) Adelaide

(GMT+10:00) Vladivostok

(GMT+10:00) Hobart

(GMT+10:00) Guam, Port Moresby

(GMT+10:00) Canberra, Melbourne, Sydney

(GMT+10:00) Brisbane

(GMT+11:00) Magadan, Solomon Is., New

Caledonia

18. Central Pacific Standard Time

(GMT+12:00) Fiji, Kamchatka, Marshall Is.

(GMT+12:00) Auckland, Wellington

(GMT+13:00) Nuku'alofa

29. Fiji Standard Time

47. New Zealand Standard Time

66. Tonga Standard Time

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Appendix 7 IP Bandwidth Requirement

1 voice

paths

30 voice

paths

60 voice

paths

120 voice

paths

Packet

duration

Compression

Bandwidth Bandwidth Bandwidth Bandwidth

(kbps)

960

640

534

1024

704

598

1440

960

800

(kbps)

1920

1280

1068

2048

1408

1196

2880

1920

1600

9600

8640

8320

(kbps)

3840

2560

2134

4096

2816

2390

5760

3840

3200

19200

17280

16640

30 ms

60 ms

90 ms

30 ms

60 ms

90 ms

20 ms

40 ms

60 ms

20 ms

40 ms

60 ms

7.231.1

(5.3kbps)

21.2

17.8

23.4

19.8

7.231.1

(6.4kbps)

G.729A

(8kbps)

26.6

160

144

138.6

4800

4320

4160

G.711 (PCM)

(64kbps)

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Appendix 8 Release Complete Cause Code

H.225 Release Complete Reason to cause IE mapping

RelaseCompleteReason code

Corresponding Q.931/Q.850 cause

vale

noBandwidth

34 - No circuit/channel available

47 – Resource Unavailable

3 – No route to destination

16 – Normal call clearing

88 – Incompatible destination

111 – Interworking, unspecified

38 – Network out of order

42 – Switching equipment comgestion

28 – Invalid number format

41 – Temporary Failure

gatekeeperResources

unreachableDestination

destinationRejection

invaliRevision

noPermission

unreachableGatekeeper

Gateway Resources

badFormatAddress

adaptiveBusy

inConf

17 – User busy

undefineReason

FacilityCallDeflection

securityDenied

calledPartyNotRegistered

callerNotRegistered

31 – Normal, unspecified

16 – Normal call clearing

31 – Normal, unspecified

20 – Subscriber absent

31 – Normal, unspecified

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PSTN to SIP Cause Code Mapping

PSTN

Cause

Code

Description

SIP Event

Unallocated number

No route to specified transit network

No route to destination

User busy

404 Not found

486 User here

No user response

No answer from the user

Subscriber absent

480 Temporarily

unavailable

Call Rejected

Number changed

403 Forbidden

410 Gone

Non-selected user clearing

Destination out of order

404 Not found

484 Address

incomplete

501 Not

implemented

404 Not found

503 Service

unavailable

503 Service

unavailable

503 Service

unavailable

503 Service

unavailable

503 Service

unavailable

Address incomplete

Facility rejected

Normal, unspecified

No, circuit available

Network out of order

Temporary failure

Switching equipment congestion

Resource unavailable

Incoming calls barred within Closed User

Group(CUG)

Bearer capability not presently available

403 Forbidden

501 Not

implemented

Bearer capability not implemented

Service or option not implemented

501 Not

implemented

503 Service

Unavailable

400 Bad request

408 Request

timeout

400 Bad request

500 Internal server

error

User not member of Closed User Group

(CUG)

Incompatible destination

Invalid message

102

111

Recovery on Expires timeout

Protocol error

Any code other than those listed above:

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SIP to PSTN Cause Code Mapping

PSTN Cause

IP Event

Description

Code

404 Not found

486 User here

No route to destination

User busy

480 Temporarily

unavailable

No user response

403 Forbidden

410 Gone

Call Rejected

Number changed

Unallocated number

404 Not found

484 Address

incomplete

501 Not

Address incomplete

Service or option not implemented

Unallocated number

implemented

404 Not found

503 Service

unavailable

503 Service

unavailable

503 Service

unavailable

503 Service

unavailable

503 Service

unavailable

403 Forbidden

501 Not

Service or option unavailable

Bearer Capability not authorized

Service or option not implemented

Service or option unavailable

implemented

501 Not

implemented

503 Service

Unavailable

400 Bad request Interworking, unspecified

408 Request

Recovery on Expires timeout

timeout

102

111

127

400 Bad request Protocol error

500 Internal

server error

Any code other than those listed

above:

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Appendix 9 RADIUS Format Attributes

RADIUS Format V2.0

Start Accounting Request Attributes

ttri-bute Name

Description

Format Sample

NAS-IP-Address

NAS-Port-Type

User-Name

IP Address of the In-Bound gateway Numeric 4 bytes unsigned long

Physical port type

Numeric 0: Asynchronous

Account number(with 4 digit pin

number on postfix)

String

5500033440

Calling-Station-Id

Called-Station-Id

Acct-Status-Type

Calling Party Number (ANI)

Destination phone number

Accounting Request Type

String

886282265699

86258765432

Numeric 1: Start Accounting

2: Stop Accounting

Service-Type

Type of service requested

Name of the Gateway (IP address)

GUID

Protocol type or family used on this

leg of the call (Telephony or VOIP)

‘Originate’ or ‘Answer’

Numeric 5: Outbound

h323-gw-id -33

h323-conf-id -24

h323-call-type -27

String

h323-gw-id =VIP2100

h323-conf-id=xxxx

h323-call-type=VOIP

h323-call-origin - 26

h323-setup-time -25

String

h323-call-origin

=Originate

h323-setup-time=

23:24:19.810 UTC Sun

Sep 26 2001

Setup time in NTP format

Acct-Session-Id

Acct-Delay-Time

A unique accounting identifier -

match start & stop

No of seconds tried in sending a

particular record

String

8 bytes, like 00012345

Numeric

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Stop Accounting Request Attributes

Attribute NAME

Description

Format

Sample

NAS-IP-Address

NAS-Port-Type

User-Name

IP Address of the In-Bound gateway Numeric

4 bytes unsigned long

0: Asynchronous

5500033440

Physical port type

Numeric

String

Account number (with 4 digit pin

number on postfix)

Called-Station-Id

Calling-Station-Id

Acct-Status-Type

Destination phone number

Calling Party Number (ANI)

Account Request Type

String

Numeric

862587654321

886282265699

1: Start Accounting

2: Stop Accounting

5: Outbound

h323-gw-id =VIP2100

h323-conf-id =xxxx

h323-call-type=VOIP

Service-Type

Type of service requested

Gateway IP address

GUID

Protocol type used on this leg of the String

call - Telephony or VOIP

Numeric

String

h323-gw-id -33

h323-conf-id -24

h323-call-type -27

h323-setup-time -25

Setup time in NTP format

String

h323-setup-

time=23:24:19.810 UTC

Sun Sep 26 2001

h323-connect-

time=23:24:19.810 UTC

Sun Sep 26 2001

h323-disconnect-

time=23:24:19.810 UTC

Sun Sep 26 2001

h323-disconnect-

cause=16

h323-call-origin

=Originate

h323-remote-

address=192.168.19.15

h323-connect-time

–28

Connect time in NTP format

Disconnect time in NTP format

h323-disconnect-

time –29

h323-disconnect-

cause –30

h323-call-origin - 26

Q.931 disconnect cause code

String

‘Originate’ or ‘Answer’

h323-remote-

address-23

IP address of the Out-Bound

gateway

Acct-Session-Id

Acct-Session-Time

Acct-Delay-Time

A unique accounting identifier-

match start & stop

For how many second the user

receive the service

No of seconds tried in sending a

particular record

String

8 bytes, like 00012345

Numeric

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Authentication Request Attributes

Attribute

NAME

Description

Format Sample

NAS-IP-

Address

IP Address of the In-Bound

gateway

Numeric 4 bytes

unsigned long

Numeric 0:

Asynchronous

Numeric 8:

Authentication

NAS-Port-Type Physical port type

Service-Type

User-Name

Type of service requested

Only

5500033440

Account number (with 4 digit pin String

number on postfix)

Calling-Station- Calling Party Number (ANI)

h323-conf-id - GUID

String

886282265699

h323-conf-

id=xxx

User-Password 16 octets user password

Authentication Response Attribute

Attribute NAME

Description

Sample

Format

h323-return-

code -103

The reason for failing

authentication

String

h323-return-code=0

0: Authenticated

1: Invalid Account

2: Invalid pin number

3: Account in use

5: Account Expired

6. Over Credit Limit

7: Denied User

10: Number of Retries

Exceeded

11: Insufficient Balance

h323-credit-

amount -101

Amount of credit

(currency) remaining in

the account

Type of billing service for String

a specific call.

String

h323-credit-amount=13.25

h323-billing-

model -109

h323-billing-model=1

0:Credit (Post Paid)

1:Debit (Prepaid)

h323-currency-type=USD

ISO 4217

h323-currency- Currency for use with

type -110 h323-credit-amount

String

USD America,

Dollars

EUR Euro

GBP U.K., Pounds

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Authorization Request Attributes

Attribute

NAME

Description

Format Sample

NAS-IP-

Address

IP Address of the In-Bound

gateway

Numeric 4 bytes

unsigned long

Numeric 0:

Asynchronous

Numeric 5: Outbound

NAS-Port-Type Physical port type

Service-Type

User-Name

Type of service requested

Account number (with 4 digit

pin number on postfix)

String

5500033440

Called-Station- Destination phone number

Calling-Station- Calling Party Number (ANI)

h323-conf-id - GUID

User-Password 16 octets user password

862587654321

886282265699

h323-conf-id

=xxxx

Authorization Response Attributes

Attribute NAME

Description

Format Sample

h323-return-

code -103

The reason for failing

authentication

Sting

h323-return-code=0

0: Authenticated

1: Invalid Account

2: Invalid pin

number

3: Account in use

4: Zero Balance

5: Account Expired

6. Over Credit Limit

7: Denied User

9: Called Number

Blocked

10: Number of

Retries Exceeded

11: Insufficient

Balance

h323-credit-time Number of seconds for which

String

h323-credit-

-102

the call is authorized

time=360

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Appendix 10 Quick Start Check List

Host Network:

IP Address: _____. _____. _____. _____

Sub-Mask: _____. _____. _____. _____

Default-Gateway:_____. _____. _____. _____

Interface Network:

IP Address: _____. _____. _____. _____

Sub-Mask: _____. _____. _____. _____

Default-Gateway: _____. _____. _____. _____

►H.323 Call:

VoIP Configuration:

pRegister to Gatekeeper

GK IP Address: _____. _____. _____. _____

Phone

pPeer To Peer

Refer to User Guide-

GK RAS Port: ______

Book setting

H.245 tunneling: pEnable pDisable

Fast Connect: pEnable Fast Start pEarly H.245 pDisable

Separate H.245 after Fast Start: pYes pNo

Fast Start Enabled Codec:

pG.711 a-law

pG.711 u-law

pG.729

pG.729 A/B

pG.723.1 (5.3K)

pG.723.1 (6.3A)

Codec Select Priority:

__ G.711 a-law

__ G.711 u-law

__ G.729

__ G.729 A/B

__ G.723.1 (5.3K)

__ G.723.1 (6.3A)

►SIP Call:

VoIP Configuration: pPeer To Peer

pRegister to SIP Proxy Server

Phone

Refer to User Guide-

Book setting

Registar Proxy Server: _____. _____. _____. _____

Registar Proxy Port: ______

Registar User ID: _________

Registar Password: _________

Outbound Proxy Server: _____. _____. _____. _____

Outbound Proxy Port: ______

Outbound User: ________

Outbound Port: ________

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180 SDP: pYes pNo

183 SDP: pYes pNo

Local Codec Codec:

pG.711 a-law

pG.711 u-law

pG.729

pNone

pG.723.1 (5.3K)

pG.723.1 (6.3A)

Accept Proxy Call Only: pYes pNo

PSTN Interface:

PCM encoding: pA-law pMu-law

PCM Idle Pattern:

Default (-1): p0x55 A-law p0xff u-law pspecified: ______

Clock Source: pExternal pInternal

pE1

Framing Method:

pAutomatic CRC4 or Double Frame selection

pDouble Frame Format

pCRC4 multi-frame

pCRC4 extend multi-frame

Line Code: pHDB3 pAMI

ISDN/PRI:

Termination Site: pNetwork pUser site

Variance:

pEuro ISDN

pAustralia ISDN

pHong Kong ISDN

pKorea ISDN

CAS:

CAS Idle ABCD signal:

pE1 MFC R2

pE1 CAS R2

Variance:

Default (-1): specified: ______

pE1 R2 MF Aregntina ANI

pE1 R2 MF Aregntina ANI 7digits

pE1 R2 MF Aregntina no ANI

pE1 R2 MF Aregntina no ANI 7 digits

pE1 R2 MF Bolivia ANI

pE1 R2 MF Bolivia ANI 7digits

pE1 R2 MF Bolivia no ANI

pE1 R2 MF Bolivia no ANI 7 digits

pE1 R2 MF Brazil ANI

pE1 R2 MF Brazil ANI 7digits

pE1 R2 MF Brazil no ANI

pE1 R2 MF Brazil no ANI 7 digits

pE1 R2 MF Chile ANI

pE1 R2 MF Chile ANI 7digits

pE1 R2 MF Chile no ANI

pE1 R2 MF Chile no ANI 7 digits

pE1 R2 MF China ANI

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pE1 R2 MF China ANI 7digits

pE1 R2 MF China no ANI

pE1 R2 MF China no ANI 7 digits

pE1 R2 MF Czech-Republic ANI

pE1 R2 MF Czech-Republic ANI 7digits

pE1 R2 MF Czech-Republic no ANI

pE1 R2 MF Czech-Republic no ANI 7 digits

pE1 R2 MF Egypt -ANI

pE1 R2 MF Egypt -ANI 7digits

pE1 R2 MF Egypt - no ANI

pE1 R2 MF Egypt - no ANI 7 digits

pE1 R2 MF India – 10 Digits no ANI

pE1 R2 MF India – 10 Digits with ANI

pE1 R2 MF India – Type 1 No ANI 10

pE1 R2 MF India – Type 2 Orig ANI 10

pE1 R2 MF India – Type 2 Term ANI 10

pE1 R2 MF India – Type 2 Term No ANI 10

pE1 R2 MF India – Type 2 Orig ANI 10

pE1 R2 MF India – Type 3 ANI 10

pE1 R2 MF India – Type 3 NoANI 10

pE1 R2 MF Indonesia - ANI

pE1 R2 MF Indonesia - ANI 7digits

pE1 R2 MF Indonesia - no ANI

pE1 R2 MF Indonesia - no ANI 7 digits

pE1 R2 MF Israel(Bezeq) - ANI

pE1 R2 MF Israel(Bezeq) - ANI 7digits

pE1 R2 MF Israel(Bezeq) -c no ANI

pE1 R2 MF Israel(Bezeq) - no ANI 7 digits

pE1 R2 MF ITU - ANI

pE1 R2 MF ITU - ANI 7digits

pE1 R2 MF ITU - no ANI

pE1 R2 MF ITU - no ANI 7 digits

pE1 R2 MF KOREA - ANI

pE1 R2 MF KOREA - ANI 7digits

pE1 R2 MF KOREA - no ANI

pE1 R2 MF KOREA - no ANI 7 digits

pE1 R2 MF Malaysia - ANI

pE1 R2 MF Malaysia - ANI 7digits

pE1 R2 MF Malaysia - no ANI

pE1 R2 MF Malaysia - no ANI 7 digits

pE1 R2 MF Mexico - ANI

pE1 R2 MF Mexico - ANI 7digits

pE1 R2 MF Mexico - no ANI

pE1 R2 MF Mexico - no ANI 7 digits

pE1 R2 MF Philippines - ANI

pE1 R2 MF Philippines - ANI 7digits

pE1 R2 MF Philippines - no ANI

pE1 R2 MF Philippines - no ANI 7 digits

pE1 R2 MF Thailand -Republic ANI

pE1 R2 MF Thailand - ANI 7digits

pE1 R2 MF Thailand - no ANI

pE1 R2 MF Thailand - no ANI 7 digits

pE1 R2 MF Uruguay - ANI

pE1 R2 MF Uruguay - ANI 7digits

pE1 R2 MF Uruguay - no ANI

pE1 R2 MF Uruguay - no ANI 7 digits

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pE1 R2 MF Venezuela - ANI

pE1 R2 MF Venezuela - ANI 7digits

pE1 R2 MF Venezuela - no ANI

pE1 R2 MF Venezuela - no ANI 7 digits

pT1

Framing Method:

psuper frame

p4-frame multi-frame

p12 frame multi-frame (D4)

pextend super frame without CRC6

pextend super frame with CRC6

p72-Frame Multi-Frame

Line Code: pAMI pB8ZS

ISDN/PRI:

Termination Site: pNetwork pUser site

Variance:

pNI2 ISDN

p5ESS 9 ISDN

p5ESS 10 ISDN

pDMS100 ISDN

pNTT ISDN (INS1500)

CAS:

CAS Idle ABCD signal:

pT1 CAS

Variance:

Default (-1): specified: ______

pT1 E&M BellCore Feature Group D Wink Start

pT1 E&M Delay Start

pT1 E&M Feature Group A Immediate Start

pT1 E&M Feature Group B Wink Start

pT1 E&M Feature Group D Wink Start(ANI B4 ADDR)

pT1 E&M Feature Group D Wink Start

pT1 E&M FGAImmediate

pT1 E&M FGB Wink

pT1 E&M FGB Wink(ANI B4 ADDRESS)

pT1 E&M FGD Wink

pT1 E&M Immediate

pT1 E&M Immediate Start

pT1 E&M Wink

pT1 E&M WinkStart A-Bit Only FXO

pT1 E&M WinkStart A-Bit Only FXS

pT1 E&M Wink Start

pT1 GroundStart FXO

pT1 GroundStart FXS

pT1 LoopStart FXO

pT1 LoopStart FXS

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VIP-2100 FAQ

Q1. Forgotten user password to logon VIP-2100.

Answer:

a. Logon by a user has Administrator right to reset the user’s password

b. Use the LCD control panel to change the user id: admin or root’s

password.

Q2. In H.323 Mode: Cannot hear ring back tone for PSTN caller.

Answer:

Normally, the ring back tone is generated by the nearest PABX

connected to VIP-2100. If a caller from PSTN site cannot hear the ring back

tone, please check:

a. Consult to PABX/PSTN vender to clarify the PABX/PSTN will generate

ring back tone.

b. For Fast Start mode, make sure the far end VoIP end point will have

ring tone generated. For example, a PSTN subscriber calls a VoIP

H.323 IP Phone. When it is on Fast Start mode, VIP-2100 will cut

through the voice path after receive Fast Start Ack. Please make sure

the Far End VoIP Endpoint will generate ring back tone over RTP

media path to VIP-2100.

c. If you really need VIP-2100 to generate PSNT ring back tone, please

do the following setting:

Turn on “local ring back” from “Interface -> Trunk” for each

trunk required local ring back tine generation.

Disable Fast Start for H.323 outgoing call (set to disable or

H.323 incoming call only.)

Q3. In H.323 Mode: VIP-2100 cannot keep registering to Gatekeeper after

Gatekeeper restarted.

Answer:

a. Check whether VIP2100’s register time to live is too long or not. If yes,

make it shorter from “H.323 -> Register Time to Live”. If we make it

longer, it means it might need take long time to re-register to

Gatekeeper after Gatekeeper failed or restart. If it is very short, will

cause more IP traffic.

b. Check whether Gatekeeper has a preset TTL setup or not. If so, the

GK TTL will overwrite the VIP-2100’s TTL request by using the default

value.

Q4. In SIP Mode: VIP-2100 cannot keep registering to SIP Register

Server after Register Server restarted.

Answer:

a. Check whether VIP-2100’s registar IP address, port, user id and

password are correct.

b. Check whether Register Server has a preset TTL setup or not. If so,

the Register Server TTL will overwrite the VIP-2100’s TTL request by

using the default value.

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Q5. VIP-2100 cannot make a success call.

Answer:

a. Check PSTN trunk ready to work or not. You need to have the

following event generated - “9504: trunk alarm clear (trunk #)”

b. Check VIP-2100 is registered to H.323: Gatekeeper /SIP: Register

Server or not. You need to have the following event generated - “9503:

H323 GK/ SIP Register [xxx.xxx.xxx.xxx] found & registered”

c. Check the digit manipulation setting is correct or not. Make sure you

have DM put into call flow editor, interface or VoIP.

d. For P2P call, make sure you have the address book setting for dialed

number.

Q6. Cannot hear voice after the calls connect.

Answer:

a. Make sure the interface and host Ethernet are well connected.

b. Ping each related IP to see network is working or not.

c. The voice codec priority should be matched both side.

Q7. In H.323 Mode: Failed to setup a fast start call.

Answer:

a. Make sure the far end “Fast Start” is enabled.

b. Check weather Gatekeeper can support Fast Start or not. (Some

Gatekeepers are not.)

c. If you cannot hear early announcement, make sure the far and H.323

end point can listen RTP port before connect.

Q8. In SIP Mode: Failed to setup a normal call.

Answer:

a. Make sure the voice codec priority should be matched both side.

b. Make sure the VIP-2100 accept proxy call only or not.

Q9. In SIP Mode: Failed to hear the early media before call connected.

Answer:

a. Make sure the 180 SDP or 183 SDP are enabled.

b. Make sure the remote SIP end point can cut through voice before call

connected.

Q10. Cannot send or receive the DTMF to/from far end VoIP end point.

Answer:

a. In H.323 Mode: Make sure VIP-2100 and other endpoint use same

DTMF relay mode (e.g. H.245 Alphanumeric.)

b. In SIP Mode: Make sure VIP-2100 and other endpoint use same

DTMF relay mode (e.g. SIP Info or RFC2833-payload type.)

c. In SIP Mode: Make sure the remote SIP end point’s RTP payload type

is supported or not.

d. If use Q.931 UUI DTMF Relay mode, make sure Gatekeeper can

correctly forward Q.931 UUI when registering to Gatekeeper is set to

true.

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Q8. Does VIP-2100 can cooperate with the Cisco VoIP products?

Answer:

The short answer is “ yes”. The following configuration example can be

used for normal and fax call.

Example for H.323 Mode

voice service voip

fax protocol t38 ls-redundancy 1 hs-redundancy 1

voice class codec 100

codec preference 1 g723r63

codec preference 2 g729r8

codec preference 3 g711ulaw

codec preference 4 g711alaw

dial-peer voice 100 pots

application session

destination-pattern 8001

progress_ind progress enable 8

port 1/1/0

dial-peer voice 200 voip

destination-pattern 2T

voice-class codec 100

session target ras

dtmf-relay h245-signal h245-alphanumeric

fax rate 14400

fax-relay ecm disable

fax protocol t38 ls-redundancy 1 hs-redundancy 1

no vad

Example for SIP Mode

voice service voip

fax protocol t38 ls-redundancy 1 hs-redundancy 1

dial-peer voice 300 voip

destination-pattern 20T

rtp payload-type nte 110

voice-class codec 88

session protocol sipv2

session target ipv4:192.168.5.205

dtmf-relay rtp-nte

fax-relay ecm disable

fax rate 14400

fax protocol t38 ls-redundancy 2 hs-redundancy 2 fallback cisco

no vad

dial-peer voice 250 voip

application session

destination-pattern 2T

voice-class codec 2

session protocol sipv2

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session target sip-server

fax rate 14400

fax protocol t38 ls-redundancy 1 hs-redundancy 1 fallback cisco

sip-ua

line con 0

speed 115200

line aux 0

line vty 0 4

Q9. VIP-2100 cannot register to Cisco gatekeeper.

Answer:

a. Make sure GK IP and port number is correct.

b. If the gatekeeper can only allow predefined endpoint, make sure VIP-

2100 has it defined.

c. If you need prefix support, set it on GK.

Q10. External Radius server does not work.

Answer:

a. Make sure “VoIP Authentication method” is set to “ external AAA”.

b. Make sure “AAA” component is used in the call flow editor to take

effect.

c. Make sure Radius server IP and port for authentication & billing are

correct.

Q11. Internal Radius server does not work.

Answer:

a. Make sure “VoIP Authentication method” is set to “ internal AAA”.

b. Make sure “AAA” component is used in the call flow editor to take

effect.

c. Make sure only debit user is used for VoIP caller.

Q12. PSTN hunting Group does not work.

Answer:

a. Make sure “PSTN hunting Group” component is used in the call flow

editor to take effect.

b. Make sure “prefix” is met your target dialed number.

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