3com VCX V7122 User manual

3com VCX V7122 User manual

3Com

®

VCX

V7122 VoIP SIP

Gateway User Manual

Version 4.4 http://www.3com.com

Part Number 900-0255-01

Published June 2005

3Com Corporation, 350 Campus Drive, Marlborough MA 01752-3064

Copyright © 2004, 2005, 3Com Corporation. All rights reserved. No part of this documentation may be reproduced in any form or by any means or used to make any derivative work (such as translation, transformation, or adaptation) without written permission from 3Com Corporation.

3Com Corporation reserves the right to revise this documentation and to make changes in content from time to time without obligation on the part of 3Com Corporation to provide notification of such revision or change.

3Com Corporation provides this documentation without warranty, term, or condition of any kind, either implied or expressed, including, but not limited to, the implied warranties, terms, or conditions of merchantability, satisfactory quality, and fitness for a particular purpose. 3Com may make improvements or changes in the product(s) and/or the program(s) described in this documentation at any time.

If there is any software on removable media described in this documentation, it is furnished under a license agreement included with the product as a separate document, in the hardcopy documentation, or on the removable media in a directory file named

LICENSE.TXT or !LICENSE.TXT. If you are unable to locate a copy, please contact 3Com and a copy will be provided to you.

UNITED STATES GOVERNMENT LEGEND

If you are a United States government agency, then this documentation and the software described herein are provided to you

subject to the following:

All technical data and computer software are commercial in nature and developed solely at private expense. Software is delivered as “Commercial Computer Software” as defined in DFARS 252.227-7014 (June 1995) or as a “commercial item” as defined in FAR 2.101(a) and as such is provided with only such rights as are provided in 3Com’s standard commercial license for the Software. Technical data is provided with limited rights only as provided in DFAR 252.227-7015 (Nov 1995) or FAR

52.227-14 (June 1987), whichever is applicable. You agree not to remove or deface any portion of any legend provided on any licensed program or documentation contained in, or delivered to you in conjunction with, this guide.

Unless otherwise indicated, 3Com registered trademarks are registered in the United States and may or may not be registered in other countries.

3Com, the 3Com logo, NBX, and SuperStack are registered trademarks of 3Com Corporation. NBX NetSet, pcXset, and VCX are trademarks of 3Com Corporation.

Adobe is a trademark and Adobe Acrobat is a registered trademark of Adobe Systems Incorporated. Microsoft, Windows,

Windows 2000, Windows NT, and Microsoft Word are registered trademarks of Microsoft Corporation.

All other company and product names may be trademarks of the respective companies with which they are associated.

2 3Com VCX V7122 SIP VoIP Gateway User Manual

C

ONTENTS

A

BOUT

T

HIS

G

UIDE

9

How to Use This Guide 9

Conventions 10

Related Documentation 10

Documentation Comments 10

C

HAPTER

1: VCX V7122 SIP O

VERVIEW

13

Available Configurations 14

SIP Overview 15

VCX V7122 VoIP SIP Supported Features 15

General Features 15

Hardware Features 15

PSTN-to-SIP Interworking 16

Supported SIP Features 17

C

HAPTER

2: VCX V7122 SIP P

HYSICAL

D

ESCRIPTION

19

General 19

The VCX V7122 Chassis 20

Power Supply 20

The TP-1610 Board 20

Board Hot-Swap Support 21

TP-1610 Front Panel LED Indicators 22

Rear Transition Module 23

Optional CPU Board 24

C

HAPTER

3: I

NSTALLING THE

VCX V7122 25

Unpacking 25

Package Contents 26

Mounting the VCX V7122 26

Mounting the VCX V7122 on a Desktop 26

Installing the VCX V7122 in a 19-inch Rack 26

Cabling the VCX V7122 28

Connecting the E1/T1 Trunk Interfaces 30

Installing the Ethernet Connection 31

Connecting the Power Supply 31

C

HAPTER

4: G

ETTING

S

TARTED

35

3Com VCX V7122 SIP VoIP Gateway User Manual 3

Assigning the VCX V7122 IP Address 35

Assigning an IP Address Using HTTP 35

Assigning an IP Address Using BootP 36

Restoring Networking Parameters to Their Initial State 37

Configuring the VCX V7122 Basic Parameters 37

C

HAPTER

5: W

EB

M

ANAGEMENT

41

Configuration Concepts 41

Overview of the Embedded Web Server 41

Computer Requirements 41

Password Control 42

Embedded Web Server Username and Password 42

Configuring the Web Interface via the ini File 42

Limiting the Embedded Web Server to Read-Only Mode 42

Disabling the Embedded Web Server 42

Accessing the Embedded Web Server 42

Using Internet Explorer to Access the Embedded Web Server 43

Getting Acquainted with the Web Interface 43

Main Menu Bar 44

Saving Changes 45

Entering Phone Numbers in Various Tables 45

Protocol Management 45

Protocol Definition Parameters 46

Advanced Parameters 47

Number Manipulation Tables 47

Configuring the Routing Tables 52

Configuring the Profile Definitions 58

Configuring the Trunk Group Table 62

Configuring the Trunk Group Settings 63

Advanced Configuration 65

Configuring the Network Settings 65

Configuring the Channel Settings 68

Configuring the Trunk Settings 69

Configuring the TDM Bus Settings 72

Restoring and Backing Up the Gateway Configuration 72

Regional Settings 73

Changing the VCX V7122 Username and Password 75

Status and Diagnostic 75

Gateway Statistics 75

Monitoring the VCX V7122 Trunks and Channels 77

Activating the Internal Syslog Viewer 79

System Information 80

Software Update Menu 81

Software Upgrade Wizard 81

4 3Com VCX V7122 SIP VoIP Gateway User Manual

Auxiliary Files 86

Updating the Software Upgrade Key 88

Save Configuration 88

Resetting the VCX V7122 89

C

HAPTER

6:

INI

F

ILE

C

ONFIGURATION OF THE

VCX V7122 91

Secured ini File 91

Modifying an ini File 91

The ini File Content 92

The ini File Structure 92

The ini File Structure Rules 92

The ini File Example 93

Basic, Logging, Web, and RADIUS Parameters 93

SNMP Parameters 101

SIP Configuration Parameters 102

ISDN and CAS Interworking-Related Parameters 114

Number Manipulation and Routing Parameters 119

E1/T1 Configuration Parameters 127

Channel Parameters 132

Dynamic Jitter Buffer Operation 136

Configuration Files Parameters 137

C

HAPTER

7: C

ONFIGURATION

F

ILES

139

Configuring the Call Progress Tones 139

Format of the Call Progress Tones Section in the ini File 139

Prerecorded Tones (PRT) File 141

PRT File Format 141

Voice Prompts File 142

CAS Protocol Configuration Files 142

C

HAPTER

8: G

ATEWAY

C

APABILITIES

D

ESCRIPTION

143

Proxy or Registrar Registration Example 143

Redirect Number and Calling Name (Display) 143

ISDN Overlap Dialing 144

Using ISDN NFAS 144

NFAS Interface ID 145

Working with DMS-100 Switches 146

Configuring the DTMF Transport Types 146

Configuring the Gateway’s Alternative Routing (based on Connectivity and QoS) 150

Alternative Routing Mechanism 150

Determining the Availability of Destination IP Addresses 150

PSTN Fallback as a Special Case of Alternative Routing 150

Relevant Parameters 151

Working with Supplementary Services 151

3Com VCX V7122 SIP VoIP Gateway User Manual 5

Call Hold and Retrieve Features 151

Call Transfer 151

TDM Tunneling 152

Implementation 152

Call Detail Report 154

Trunk to Trunk Routing Example 156

SIP Call Flow Example 157

SIP Authentication Example 160

C

HAPTER

9: D

IAGNOSTICS

163

VCX V7122 Self-Testing 163

Syslog Support 163

Syslog Servers 164

Operation 164

C

HAPTER

10: B

OOT

P/DHCP S

UPPORT

167

Startup Process 167

DHCP Support 169

BootP Support 169

Upgrading the VCX V7122 169

Vendor Specific Information Field 170

C

HAPTER

11: SNMP-B

ASED

M

ANAGEMENT

173

About SNMP 173

SNMP Message Standard 173

SNMP MIB Objects 174

SNMP Extensibility Feature 175

Carrier Grade Alarm System 175

Active Alarm Table 175

Alarm History 176

Cold Start Trap 176

Third-Party Performance Monitoring Measurements 176

Supported MIBs 177

SNMP Interface Details 180

SNMP Community Names 180

Trusted Managers 181

SNMP Ports 183

Multiple SNMP Trap Destinations 183

SNMP Manager Backward Compatibility 185

Element Management System 185

C

HAPTER

12: S

ELECTED

T

ECHNICAL

S

PECIFICATIONS

187

A

PPENDIX

A: VCX V7122 SIP S

OFTWARE

K

IT

191

6 3Com VCX V7122 SIP VoIP Gateway User Manual

A

PPENDIX

B: T

HE

B

OOT

P/TFTP C

ONFIGURATION

U

TILITY

193

When to Use the BootP/TFTP 193

An Overview of BootP 193

Key Features 193

Specifications 194

Installation 194

Loading the cmp File, Booting the Device 194

BootP/TFTP Application User Interface 195

Function Buttons on the Main Screen 195

Log Window 196

Setting the Preferences 197

BootP Preferences 198

TFTP Preferences 198

Configuring the BootP Clients 199

Adding Clients 199

Deleting Clients 200

Editing Client Parameters 200

Testing the Client 201

Setting Client Parameters 201

Using Command Line Switches 202

Managing Client Templates 203

A

PPENDIX

C: RTP/RTCP P

AYLOAD

T

YPES AND

P

ORT

A

LLOCATION

205

Payload Types Defined in RFC 1890 205

Defined Payload Types 205

Default RTP/RTCP/T.38 Port Allocation 206

A

PPENDIX

D: F

AX AND

M

ODEM

T

RANSPORT

M

ODES

209

Configuring Fax Relay Mode 209

Configuring Fax/Modem ByPass Mode 209

Supporting V.34 Faxes 210

A

PPENDIX

E: VCX V7122 C

LOCK

S

ETTINGS

211

A

PPENDIX

F: C

USTOMIZING THE

VCX V7122 W

EB

I

NTERFACE

213

Replacing the Main Corporate Logo 213

Replacing the Main Corporate Logo with an Image File 214

Replacing the Main Corporate Logo with a Text String 215

Replacing the Background Image File 215

Customizing the Product Name 216

Modifying ini File Parameters via the Web AdminPage 217

A

PPENDIX

G: A

CCESSORY

P

ROGRAMS AND

T

OOLS

219

3Com VCX V7122 SIP VoIP Gateway User Manual 7

TrunkPack Downloadable Conversion Utility 219

Converting a CPT ini File to a Binary dat File 220

Creating a Loadable Voice Prompts File 221

Encoding/Decoding an ini File 223

Creating a Loadable Prerecorded Tones File 224

PSTN Trace Utility 226

Operation 226

A

PPENDIX

H: S

OFTWARE

U

PGRADE

K

EY

229

Loading the Software Upgrade Key 229

Loading the Software Upgrade Key Using the Embedded Web Server 229

Loading the Software Upgrade Key Using BootP/TFTP 231

Verifying That the Key was Successfully Loaded 231

Troubleshooting an Unsuccessful Loading of a Key 231

A

PPENDIX

I: R

ELEASE

R

EASON

M

APPING

233

A

PPENDIX

J: RADIUS B

ILLING AND

C

ALLING

C

ARD

A

PPLICATION

237

Benefits 237

Features 237

Supported Architecture 238

Implementation 239

Basic Calling Card IVR Scenario 239

Call Flow Description 240

Operation and Configuration 241

Configuration Parameters 242

Supported RADIUS Attributes 243

RADIUS Server Messages 246

Authentication 246

Authorization 247

Accounting 247

Voice XML Interpreter 247

Features 248

Supported Elements and Attributes 249

Provided Calling Card System 255

Voice Prompts 255

VXML Flow Chart 257

VXML Script Example 261

8 3Com VCX V7122 SIP VoIP Gateway User Manual

A

BOUT

T

HIS

G

UIDE

This User Manual describes the 3Com

®

VCX

V7122 Gateway.

This product is supported by software version 4.4, and enables you to send voice, fax, and data over the same IP network.

Information contained in this document is believed to be accurate and reliable at the time of printing. However, because of on-going product improvements and revisions, 3Com cannot guarantee the accuracy of printed material after the Date Published nor can it accept responsibility for errors or omissions. Updates to this document and other documents can be viewed by registered Technical Support customers. See Appendix K: Obtaining Support for

Your 3Com Products for details on how to register your product and get support from 3Com.

How to Use This Guide

This book covers these topics:

Chapter 1: VCX V7122 SIP Overview

Chapter 2: VCX V7122 SIP Physical Description

Chapter 3: Installing the VCX V7122

Chapter 4: Getting Started

Chapter 5: Web Management

Chapter 6: ini File Configuration of the VCX V7122

Chapter 7: Configuration Files

Chapter 8: Gateway Capabilities Description

Chapter 9: Diagnostics

Chapter 10: BootP/DHCP Support

Chapter 11: SNMP-Based Management

Chapter 12: Selected Technical Specifications

Appendix A: VCX V7122 SIP Software Kit

Appendix B: The BootP/TFTP Configuration Utility

Appendix C: RTP/RTCP Payload Types and Port Allocation

Appendix D: Fax and Modem Transport Modes

3Com VCX V7122 SIP VoIP Gateway User Manual 9

Appendix E: VCX V7122 Clock Settings

Appendix F: Customizing the VCX V7122 Web Interface

Appendix G: Accessory Programs and Tools

Appendix H: Software Upgrade Key

Appendix I: Release Reason Mapping

Appendix J: RADIUS Billing and Calling Card Application

Appendix K: Obtaining Support for Your 3Com Products

Conventions

Table 1 lists conventions that are used throughout this guide.

Table 1

Notice Icons

Information note

Description

Information that describes important features or instructions.

Caution

Warning

Information that alerts you to potential loss of data or potential damage to an application, device, system, or network.

Information that alerts you to potential personal injury or death.

Related Documentation

The following documents are available on the 3Com Partner Access website for the 3Com

VCX V7111 Gateway:

3Com VCX V7122 SIP VoIP Gateway Release Notes

3Com VCX V7122 SIP VoIP Gateway Installation Guide

Where “network” appears in this manual, it means LAN, WAN, etc. accessed via the gateway’s Ethernet interface.

Documentation Comments

Your suggestions are important to us because we want to make our documentation more useful to you.

Please send e-mail comments about this guide or any of the VCX 7111 documentation and

Help systems to:

[email protected]

Please include the following information with your comments:

Document part number (usually found on the front page)

10 3Com VCX V7122 SIP VoIP Gateway User Manual

Your name and organization (optional)

Example:

3Com VCX V7112 VoIP SIP Gateway User Manual

Page 25

Part Number 900-0255-01

3Com VCX V7122 SIP VoIP Gateway User Manual 11

12 3Com VCX V7122 SIP VoIP Gateway User Manual

C

HAPTER

1: VCX V7122 SIP O

VERVIEW

The VCX V7122 SIP (Session Initialization Protocol) Voice over IP

(

VoIP) gateway enables voice, fax, and data traffic to be sent over the same IP network. The VCX V7122 gateway provides excellent voice quality and optimized packet voice streaming over IP networks.

The VCX V7122 uses the award-winning, field-proven DSP voice compression technology used in other TrunkPack

TM

series products.

The VCX V7122 incorporates 1, 2, 4, 8 or 16 E1 or T1 spans for connection, directly to

Public Switched Telephone Network (PSTN) / Private Branch Exchange (PBX) telephony trunks, and includes one or two 10/100 Base-TX Ethernet ports for connection to the network.

The VCX V7122 supports up to 480 simultaneous VoIP or Fax over IP (FoIP) calls, supporting various Integrated Services Digital Network (ISDN) Primary Rate Interface (PRI) protocols such as EuroISDN, North American NI2, Lucent 5ESS, Nortel DMS100 and others.

In addition, it supports different variants of Channel Associated Signaling (CAS) protocols for

E1 and T1 spans, including MFC R2, E&M immediate start, E&M delay dial/start, loop start and ground start.

The VCX V7122 gateway, best suited for large and medium-sized VoIP applications, is a compact device, comprising a 19-inch 1U chassis with optional dual AC or single DC power supplies.

The deployment architecture can include several VCX V7122 gateways in branch or departmental offices, connected to local PBXs. Call routing is performed by the gateways themselves or by SIP Proxy(s).

The VCX V7122 gateway enables users to make low cost long distance or international telephone/fax calls between distributed company offices, using their existing telephones/fax.

These calls are routed over the existing network ensuring that voice traffic uses minimum bandwidth.

The VCX V7122 can also route calls over the network using SIP signaling protocol, enabling the deployment of "Voice over Packet" solutions in environments where access is enabled to

PSTN subscribers by using a trunking media gateway. This provides the ability to transmit voice and telephony signals between a packet network and a TDM network. Routing of the calls from the PSTN to a SIP service node (e.g., Call Center) is performed by the VCX

V7122 internal routing feature or by a SIP Proxy.

Figure 1 below illustrates typical VCX V7122 gateway applications over VoIP Network.

3Com VCX V7122 SIP VoIP Gateway User Manual 13

Figure 1

Typical VCX V7122 Gateway Application

PSTN

Telephone

Router

LAN

LAN

E1/T1 PRI/CAS

Mediant 2000

SIP Proxy

Router

SIP

Service

Node

Mediant 2000

LAN

E1/T1 PRI/CAS

Router

IP Netw ork

Router

LAN

Mediant 2000

E1/T1 PRI/CAS

PBX - Branch A

Available Configurations

The VCX V7122 is provided in the following configurations.

E1 Available Configurations:

30 Channels on 1 E1 span with gateway-1 only

60 Channels on 2 E1 spans with gateway-1 only

120 Channels on 4 E1 spans with gateway-1 only

240 Channels on 8 E1 spans with gateway-1 only

480 Channels on 16 E1 spans with gateway-1 and gateway-2

14

PBX - Branch B

3Com VCX V7122 SIP VoIP Gateway User Manual

T1 Available Configurations:

24 Channels on 1 T1 span with gateway-1 only

48 Channels on 2 T1 spans with gateway-1 only

96 Channels on 4 T1 spans with gateway-1 only

192 Channels on 8 T1 spans with gateway-1 only

384 Channels on 16 T1 spans with gateway-1 and gateway-2

SIP Overview

SIP is an application-layer control (signaling) protocol used on the VCX V7122 for creating, modifying, and terminating sessions with one or more participants. These sessions can include Internet telephone calls, media announcements and conferences.

SIP invitations are used to create sessions and carry session descriptions that enable participants to agree on a set of compatible media types. SIP uses elements called proxy servers to help route requests to the user's current location, authenticate and authorize users for services, implement provider call-routing policies and provide features to users.

SIP also provides a registration function that enables users to upload their current locations for use by proxy servers. SIP, on the VCX V7122, complies with the IETF (Internet

Engineering Task Force) RFC 3261 (see http://www.ietf.org

).

VCX V7122 VoIP SIP Supported Features

This section provides a high-level overview of some of the many VCX V7122 supported features.

General Features

Superior, high quality PSTN gateway for Voice and Fax over IP calls.

Up to 16 E1/T1/J1 digital spans supporting various PRI and CAS protocols.

Compliant with SIP (RFC 3261).

Coders include: G.711, G.723.1, G.726, G.729A and NetCoder at 6.4 to 8.8 kbps, negotiable per channel.

Echo Canceler with up to 128 msec tail length.

Silence suppression with Comfort Noise Generation.

Web management for easy configuration and installation.

Simple Network Management Protocol (SNMP) and Syslog support.

Simple Network Time Protocol (SNTP) support, the time-of-day can be obtained from a standard SNTP server.

Hardware Features

Two 10/100 Base-TX Ethernet interface connections to the network, providing network redundancy.

Compact, rugged 19-inch rack mount unit, one U high (1.75" or 44.5 mm), with two compactPCI (cPCI) slots.

Optional cPCI slot for third-party CPU board.

3Com VCX V7122 SIP VoIP Gateway User Manual 15

Optional dual redundant AC or a single DC power supply.

PSTN-to-SIP Interworking

The VCX V7122 gateway performs interworking between ISDN and CAS via E1/T1/J1 digital spans and SIP IETF signaling protocol. 16 E1, T1 or J1 spans are supported (480 channels) in a two modules gateway.

The VCX V7122 gateway supports various ISDN PRI protocols such as EuroISDN, North

American NI2, Lucent 5ESS, Nortel DMS100, Meridian 1 DMS100, Japan J1, as well as

QSIG (basic call). PRI support includes User Termination or Network Termination side.

ISDN-PRI protocols can be defined on an E1/T1 basis (i.e., different variants of PRI are allowed on different E1/T1 spans).

In addition, it supports numerous variants of CAS protocols for E1 and T1 spans, including

MFC R2, E&M wink start, E&M immediate start, E&M delay dial/start, loop-start, and ground start. CAS protocols can be defined on an E1/T1 basis (i.e., different variants of CAS are allowed on different E1/T1 spans).

PSTN to SIP and SIP to PSTN Called number can be optionally modified according to rules that are defined in gateway ini file.

Supported Interworking Features

Definition and use of Trunk Groups for routing IP PSTN calls.

B-channel negotiation for PRI spans.

ISDN Non Facility Associated Signaling (NFAS).

Supports SIP2QSIG IETF draft-ietf-sipping-qsig2sip-04.txt, including interworking between 180/183 responses with Session Description Protocol (SDP) and Q.931

Progress message.

PRI to SIP Interworking of Q.931 Display (Calling name) information element.

PRI (NI-2) to SIP interworking of Calling Name using Facility IE in Setup and Facility messages.

Configuration of Numbering Plan and Type for IP ISDN calls.

Interworking of PSTN to SIP release causes.

Interworking of ISDN redirect number to SIP diversion header (according to IETF draftlevy-sip-diversion-05.txt).

Optional change of redirect number to called number for ISDN IP calls.

Interworking of ISDN calling line Presentation & Screening indicators using RPID header

<draft-ietf-sip-privacy-04.txt>.

Interworking of Q.931 Called and Calling Number Type and Number Plan values using the RPID header.

Supports ISDN en-block or overlap dialing for incoming Tel IP calls.

Supports routing of IP Tel calls to predefined trunk groups.

Supports a configurable channel select mode per trunk group.

Supports various number manipulation rules for IP Tel and Tel IP, called and calling numbers.

16 3Com VCX V7122 SIP VoIP Gateway User Manual

Option to configure ISDN Transfer Capability (per Gateway).

Supported SIP Features

The VCX V7122 SIP VoIP main features are:

Reliable User Datagram Protocol (UDP) transport, with retransmissions.

T.38 real time Fax (using SIP).

Note: If the remote side includes the fax maximum rate parameter in the SDP body of the Invite message, the gateway returns the same rate in the response SDP.

Works with Proxy or without Proxy, using an internal routing table.

Fallback to internal routing table if Proxy is not responding.

Supports up to four Proxy servers. If the primary Proxy fails, the VCX V7122 automatically switches to a redundant Proxy.

Supports Proxy server discovery using Domain Name Server (DNS) SRV records.

Proxy and Registrar Authentication (handling 401 and 407 responses) using Basic or

Digest methods.

Supported methods: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, INFO,

REFER, UPDATE, NOTIFY, PRACK and SUBSCRIBE.

Modifying connection parameters for an already established call (re-INVITE).

Working with a Redirect server and handling 3xx responses.

Early Media (supporting 183 Session Progress).

PRACK reliable provisional responses <RFC 3262>.

Call Hold and Transfer Supplementary services using REFER, Refer-To, Referred-By,

Replaces and NOTIFY messages.

Supports RFC 3327 – Adding “Path” to Supported header.

Supports RFC 3581 – Symmetric Response Routing.

Session Timer <draft-ietf-sip-session-timer-10.txt>.

RFC 2833 Relay for Dual Tone Multi Frequency (DTMF) digits, including payload type negotiation.

DTMF out-of-band transfer using:

INFO method <draft-choudhuri-sip-info-digit-00.txt>

INFO method, compatible with Cisco gateways

NOTIFY method <draft-mahy-sipping-signaled-digits-01.txt>.DTMF out-of-band transfer using INFO method (draft-choudhuri-sip-info-digit-00.txt)

Can negotiate coder from a list of given coders.

G.711 A-law 64 kbps (10, 20, 30, 40, 50, 60, 80, 100, 120 msec)

(10, 20, 30, 40, 50, 60, 80, 100, 120 msec)

G.723.1 5.3, 6.3 kbps

G.726 32 kbps

(30, 60, 90, 120, 150 msec)

(10, 20, 30, 40, 50, 60, 80, 100, 120 msec)

G.729A 8 kbps (10, 20, 30, 40, 50, 60, 80, 100, 120 msec)

G.729B is supported if Silence Suppression is enabled.

3Com VCX V7122 SIP VoIP Gateway User Manual 17

NetCoder 6.4, 7.2, 8.0 and 8.8 kbps (20, 40, 60, 80, 100, 120 msec).

Transparent

For more updated information on the gateway’s supported features, see the latest VCX

V7122 & TP-1610 SIP Release Notes.

18 3Com VCX V7122 SIP VoIP Gateway User Manual

C

HAPTER

2: VCX V7122 SIP P

HYSICAL

D

ESCRIPTION

This section provides detailed information on the VCX V7122 hardware components, the location and functionality of the LEDs, buttons and connectors on the front and rear panels.

General

The VCX V7122 gateway comprises the following hardware components:

A 19-inch 1U high rack mount chassis (see The VCX V7122 Chassis on page 20 ).

A single compactPCI™ TP-1610 board (see The TP-1610 Board on page 20 ).

A single TP-1610 Rear Transition Module (RTM) (see Rear Transition Module on page 23 ).

A single available cPCI slot for an optional third-party CPU board (see Optional CPU

Board on page 24 ).

Figure 2 shows the front view of the VCX V7122 media gateway.

Figure 2

VCX V7122 Front View

3

4 11

1 2 5 6 7 8

Table 2

VCX V7122 Front View Component Descriptions

Item # Label

1 FAULT

2

3

4

Component Description

Dual AC Power LED. cPCI board locking screws.

TP-1610 cPCI board, 16-trunk configuration.

6

9 10

3Com VCX V7122 SIP VoIP Gateway User Manual 19

5

6

7 ETH

8

9

T1/E1 STATUS

10

11

Status LED Indicators.

E1/T1 Trunk Status LED Indicators.

Ethernet LED Indicators. cPCI LED Indicators.

Power and Fan LEDs.

An available cPCI slot for an optional third-party CPU board.

The VCX V7122 Chassis

The VCX V7122 chassis is an industrial platform, 19” wide, 1U high and 12” deep that houses the TP-1610 board in its front cage, slot #1 (the lower slot) and the TP-1610 RTM in its rear cage, slot #1 (the lower slot).

Slot # 2 in the VCX V7122 chassis’ front and rear cages can optionally be used by customers for a CPU board.

See Table 3 for detailed description of the chassis’ LED indicators.

Table 3

Chassis LED Indicators

Right side of front panel

Right side of front panel

Green

Red

Left side of front panel

Red

The power is on.

Fan failure - indicates that any of the internal fans has significantly reduced its speed or has frozen.

Power supply failure - indicates that one of the two AC redundant power supplies is faulty or disconnected from the

AC/mains outlet. (This LED is only relevant for the dual AC power supply).

Power Supply

The VCX V7122 power supply is available in three configuration options:

Single universal 100-240 VAC 1 A max, 50-60 Hz.

Dual-redundant 100-240 VAC 1.5 A max, 50-60 Hz.

-48 VDC power supply suitable for field wiring applications.

The TP-1610 Board

The VCX V7122 is populated by a single compactPCI board, the TP-1610 (see Figure 3 on page 21 ). The TP-1610 is a high-density, hot-swappable, cPCI resource board with a capacity of up to 480 ports, supporting all necessary functions for voice, data and fax streaming over IP networks. The TP-1610 is composed of one or two identical media gateways modules: Gateway-1 and Gateway-2, each containing 240 DSP channels. These media gateways are fully independent, each gateway having its own MAC (Media Access

Control) and IP addresses and LED indicators. The TP-1610 board is supplied with a rear I/O

20 3Com VCX V7122 SIP VoIP Gateway User Manual

configuration in which both PSTN trunks and Ethernet interface are located on a passive rear

I/O module (for information on the RTM see Rear Transition Module on page 23 ).

Figure 3

Front and Upper View of the TP-1610 cPCI Board

1

2

3

4

5

6

7

Table 4

Front and Upper View of the TP-1610 cPCI Board Component Descriptions

Component Description

6

7

Item # Label

1

2 ETH

3

4

5

T1 / E1 STATUS

T1 / E1 STATUS

T1/E1 Trunk Status LEDs (for each of trunks 1 to 8)

T1/E1 Trunk Status LEDs (for each of trunks 9 to 16)

Board Hot-Swap Support

The TP-1610 cPCI board is hot-swappable and therefore can be inserted and removed when the VCX V7122 chassis is under power.

When inserting the board into a cPCI chassis, the blue LED is normally lit for a second and then turns off. An unlit blue LED indicates that the board has been inserted correctly and power supply to the board is functioning correctly. If the board has any abnormal physical or

3Com VCX V7122 SIP VoIP Gateway User Manual 21

electrical condition, then the blue LED is lit, indicating a fault, and the board is NOT powered up.

When removing the board from a cPCI chassis, press the red latches at both ends of the boards and wait for the blue LED to light, indicating that the board can be removed.

TP-1610 Front Panel LED Indicators

The functionality of the front panel LEDs for the TP-1610 is described in the following four tables and illustrated in Figure 3 on page 21 . Note that there is a choice of front panels according to the number of channels.

Table 5

Status LED Indicators

Label

FAIL

ACT

LED Color LED Function

Red Normally OFF; Red indicates gateway failure (fatal error)

Green Gateway initialization sequence terminated OK

Yellow N/A

Bi-color LED

Table 6

E1/T1 Trunk Status LED Indicators

Label

T1/E1 Status 1 to 8 and

T1/E1 Status 9 to 16

LED Color Signal Description

Green Trunk is synchronized (normal operation)

Red Loss due to any of the following 4 signals:

LOS

LFA

Loss of Signal

Loss of Frame Alignment

AIS

RAI

Bi-color LED

Alarm Indication Signal (the Blue Alarm)

Remote Alarm Indication (the Yellow Alarm)

On the front panel 16 LEDs are provided for 16-span units and 8 LEDs are provided for 1-span, 2-span, 4-span, and 8-span units. In the case of 1-span,

2-span and 4-span units, the extra LEDs are unused.

Table 7

Ethernet LED Indicators

Label

LINK

ACT

LED Color LED Function

Green Link all OK

Yellow Transmit / receive activity

22 3Com VCX V7122 SIP VoIP Gateway User Manual

Table 8

cPCI LED Indicators

Label

PWR

SWAP READY

LED Color LED Function

Green Power is supplied to the board

Blue

The cPCI board can now be removed.

The cPCI board was inserted successfully.

For detailed information on the Swap-Ready LED see

Board Hot-Swap Support on page

21

.

During correct VCX V7122 operation, the ACT LED is lit green, the FAIL LED is off.

Changing of the FAIL LED to red indicates a failure.

Rear Transition Module

The VCX V7122 RTM includes a PSTN trunks and an Ethernet interfaces.

The Ethernet interface features dual 10/100 Base-TX, RJ-45 shielded connectors for (an active / standby) redundancy scheme providing protection against the event of a failure.

The PSTN interface is provided with a choice of rear panels (1-span, 2-span, 4-span, 8-span or 16-span).

Rear panel with two 50-pin female Telco connectors (DDK 57AE-40500-21D) (see Figure 4 ) is required for a gateway equipped with up to 16 E1/T1 spans. Rear panel with RJ-48c connectors (see Figure 5 on page 24 ) is required for a gateway equipped with 1, 2, 4, or 8

E1/T1 spans. The physical difference between the 1-Span, 2-Span and 4-Span RTMs, and the 8-span RTM is that the RJ-48c ports are depopulated correspondingly.

Figure 4

Rear Panel with two 50-pin Connectors for 16 Trunks

1

2

3Com VCX V7122 SIP VoIP Gateway User Manual

3

23

Table 9

Rear Panel with two 50-pin Connectors for 16 Trunks Component Descriptions

Item # Label

1

2

3

ETHERNET

TRUNKS

TRUNKS

Component Description

2 Ethernet Ports. 2 RJ-45 network connectors.

E1/T1 trunks 9 to 16. 50-pin male Telco connector.

E1/T1 trunks 1 to 8. 50-pin male Telco connector.

Figure 5

Rear Panel with 8 RJ-48c Connectors for 8 Trunks

1 2

Table 10

Rear Panel with 8 RJ-48c Connectors for 8 Trunks Component Descriptions

Item # Label

1 ETHERNET

2 TRUNKS

Component Description

2 Ethernet Ports. 2 RJ-45 network connectors

8 E1/T-1 Spans. 8 RJ-48c trunk connectors

Optional CPU Board

The VCX V7122 provides an optional second cPCI slot that can be optionally used for customer’s CPU board. This CPU board can be used for general applications such as a

Gatekeeper, Softswitch, Application Server or other. The following CPU boards were tested for compliancy with the VCX V7122 chassis:

Sun CP2080 + PMC-233 (Ramix disk on board) + Rear Transition Module (RTM).

Intel ZT5515B-1A with 40GB on-board disk plus RTM (ZT4807).

24 3Com VCX V7122 SIP VoIP Gateway User Manual

C

HAPTER

3: I

NSTALLING THE

VCX V7122

This section describes the hardware installation procedures for the VCX V7122. For information on how to start using the gateway, see Chapter 4: Getting Started on page 35 .

For detailed information on the VCX V7122 connectors, LEDs and buttons, see Chapter 2:

VCX V7122 SIP Physical Description on page 19 .

CAUTION Electrical Shock: The equipment must only be installed or serviced by qualified service personnel.

To install the VCX V7122, follow these steps:

1 Unpack the VCX V7122 (see Unpacking below).

2 Check the package contents (see

Package Contents on page 26 ).

3 Mount the VCX V7122 (see Mounting the VCX V7122 on page 26 ).

4 Cable the VCX V7122 (see

Cabling the VCX V7122 on page 28 ).

After powering-up the VCX V7122, the Ready and LAN LEDs on the front panel turn to green

(after a self-testing period of about 3 minutes). Any malfunction changes the Ready LED to red (see TP-1610 Front Panel LED Indicators on page 22 for details on the VCX V7122

LEDs).

When you have completed the above relevant sections you are then ready to start configuring the gateway (see Chapter 4: Getting Started on page 35 ).

Unpacking

CAUTION Electrical Component Sensitivity: Electronic components on printed circuit boards are extremely sensitive to static electricity. Normal amounts of static electricity generated by clothing can damage electronic equipment. To reduce the risk of damage due to electrostatic discharge when installing or servicing electronic equipment, it is recommended that anti-static grounding straps and mats be used.

To unpack the VCX V7122, follow these steps:

1 Open the carton and remove packing materials.

2 Remove the VCX V7122 gateway from the carton.

3 Check that there is no equipment damage.

4 Check, retain and process any documents.

5 Notify 3Com or your local supplier of any damage or discrepancies.

6 Retain any diskettes or CDs.

3Com VCX V7122 SIP VoIP Gateway User Manual 25

Package Contents

Ensure that in addition to the VCX V7122, the package contains:

For the dual AC power supply version two AC power cables are supplied; for the single

AC power supply version one AC power cable is supplied.

For the DC power supply version, one connectorized DC power cable (crimp connection type) and one DC adaptor (screw connection type) connected to the rear panel of the

VCX V7122 are supplied; use only one type.

CD (software and documentation).

Small plastic bag containing (see Figure 6 ):

Two brackets and four bracket-to-device screws for 19-inch rack installation option.

Four anti-slide bumpers for desktop / shelf installation option.

Figure 6

19-inch Rack & Desktop Accessories

Mounting the VCX V7122

The VCX V7122 can be mounted on a desktop, or installed in a standard 19-inch rack. See

Cabling the VCX V7122 on page 28 for cabling the VCX V7122.

Mounting the VCX V7122 on a Desktop

No brackets are required. Optionally, attach the four (supplied) anti-slide bumpers to the base of the VCX V7122 and place it on the desktop in the position you require.

Installing the VCX V7122 in a 19-inch Rack

Users can install the device in a standard 19-inch rack either by placing the device on a shelf preinstalled in the rack (preferred method), or by attaching the device directly to the rack’s frame via integral brackets.

Before rack mounting the chassis, attach the two (supplied) brackets to the front sides of the device (see Figure 7 ).

26 3Com VCX V7122 SIP VoIP Gateway User Manual

To attach the two front side brackets, follow these steps:

1 Remove the 2 screws nearest the front panel on either side of the device.

2 Align a bracket over 2 holes on one side (so that the bracket’s larger holes face front) and with the 2 supplied replacement screws, screw in the bracket.

3 Perform the same procedure on the other side.

Figure 7

VCX V7122 Front View with 19-inch Rack Mount Brackets

CAUTION Electrical Component Sensitivity

When installing the chassis in a rack, be sure to implement the following Safety instructions recommended by Underwriters Laboratories:

Elevated Operating Ambient - If installed in a closed or multi-unit rack assembly, the operating ambient temperature of the rack environment may be greater than room ambient. Therefore, consideration should be given to installing the equipment in an environment compatible with the maximum ambient temperature

(Tma) specified by the manufacturer.

Reduced Air Flow - Installation of the equipment in a rack should be such that the amount of air flow required for safe operation on the equipment is not compromised.

Mechanical Loading - Mounting of the equipment in the rack should be such that a hazardous condition is not achieved due to uneven mechanical loading.

Circuit Overloading - Consideration should be given to the connection of the equipment to the supply circuit and the effect that overloading of the circuits might have on overcurrent protection and supply wiring. Appropri-ate consideration of equipment nameplate ratings should be used when addressing this concern.

Reliable Grounding - Reliable grounding of rack-mounted equipment should be maintained. Particular attention should be given to supply connections other than direct connections to the branch circuit (e.g., use of power strips.)

To attach the device to a 19-inch rack, follow these steps:

1 Position the device in your 19-inch rack and align the left-hand and right-hand bracket holes to holes (of your choosing) in the vertical tracks of the 19-inch rack.

3Com VCX V7122 SIP VoIP Gateway User Manual 27

2 Use standard 19-inch rack bolts (not provided) to fasten the device to the frame of the rack.

3Com recommends using two additional (not supplied) rear mounting brackets to provide added support.

Users assembling the rear brackets by themselves should note the following. The distance between the screws on each bracket is 26.5 mm. To attach the brackets, use 4-40 screws with a maximal box penetration length of 3.5 mm.

To place the device on a 19-inch rack’s shelf, follow these steps:

1 Place the device on the preinstalled shelf.

2 You’re now recommended to take the optional steps of fastening the device to the frame of the rack (as described above) while it is placed on the shelf, so preventing it from sliding when inserting cables into connectors on the rear panel.

Cabling the VCX V7122

See Chapter 2: VCX V7122 SIP Physical Description on page 19 for detailed information on the VCX V7122 rear panel connectors and LEDs.

Note that the VCX V7122 is available in many configurations, i.e., AC or DC, in the 16-trunk,

8-trunk, 4-trunk, 2-trunk or 1-trunk device. The 16-trunk dual AC (see Figure 8 ) and the

8-trunk DC (see Figure 9 ) configurations are illustrated here as representative products.

Figure 8

VCX V7122 Rear Panel Cabling (16 Trunks, Dual AC Power)

1

2 3

2 1 4 5

28 3Com VCX V7122 SIP VoIP Gateway User Manual

Table 11

VCX V7122 Rear Panel Cabling (16 Trunks, Dual AC Power) Component Descriptions

3

4

5

Item # Label

1

ETHERNET

2

TRUNKS

100-240~1.5A

Component Description

RTM locking screws.

Two Category 5 network cables, connected to the 2 Ethernet RJ-45 ports.

Two 50-pin Telco connector cables, each supporting 8 trunks.

Protective grounding screw.

Dual AC power cables.

Figure 9

VCX V7122 Rear Panel Cabling (8 Trunks, DC Power)

1 2 3 3 1

4

5

Table 12

VCX V7122 Rear Panel Cabling (8 Trunks, DC Power) Component Descriptions

5

2

3

4

Item # Label

1

ETH

PSTN

48V 4A max

Component Description

A Category 5 network cable, connected to the Ethernet 1 RJ-45 port.

8 RJ-48c ports, each supporting a trunk.

Protective grounding screw.

2-pin connector for DC.

CAUTION Electrical Grounding: The unit must be permanently connected to ground via the screw provided at the back on the unit. Use 14-16 AWG wire and a proper ring terminal for the grounding.

To cable the VCX V7122, follow these steps:

1 Permanently connect the device to a suitable ground with the protective grounding screw on the rear connector panel, using 14-16 AWG wire.

2 Connect the E1/T1 trunk interfaces (see

Connecting the E1/T1 Trunk Interfaces below).

3Com VCX V7122 SIP VoIP Gateway User Manual 29

3 Install the Ethernet connection (see Installing the Ethernet Connection on page 31 ).

4 Connect the power supply (see Connecting the Power Supply on page 31 ).

Connecting the E1/T1 Trunk Interfaces

Connect the VCX V7122 E1/T1 Trunk Interfaces using either Telco or RJ-48 connectors:

With 50-pin Telco connectors (16-trunk device), follow these steps:

1 Attach the Trunk cable with a 50-pin male Telco connector to the 50-pin female Telco connector labeled “Trunks 1 8” on the Rear Transition Module (RTM).

2 Connect the other end of the Trunk cable to the PBX/PSTN switch.

3 Repeat steps 1 and 2 for the other Trunk cable but this time connect it to the connector labeled “Trunks 9 16”.

The 50-pin male Telco cable connector must be wired according to the pinout in Table 13 below, and to mate with the female connector illustrated in Figure 10 .

Figure 10

50-pin Female Telco Board-Mounted Connector

25

Pin Numbers

1

50

26

Table 13

E1/T1 Connections on each 50-pin Telco Connector

E1/T1 Number

1 to 8 9 to 16

1 9

2 10

3 11

4 12

5 13

6 14

7 15

8 16

Tx Pins (Tip/Ring)

27/2

29/4

31/6

33/8

35/10

37/12

39/14

41/16

Rx Pins (Tip/Ring)

26/1

28/3

30/5

32/7

34/9

36/11

38/13

40/15

With RJ-48c Connectors, follow these steps:

1 Connect the E1/T1 trunk cables to the ports labeled “Trunks 1 to 8” (in the case of the 8trunk device) on the VCX V7122 RTM.

2 Connect the other ends of the Trunk cables to the PBX/PSTN switch.

RJ-48c trunk connectors are wired according to Figure 11 below.

30 3Com VCX V7122 SIP VoIP Gateway User Manual

Figure 11

Pinout of RJ-48c Trunk Connectors

RJ-48c Connector and Pinout

1 2 3 4 5 6 7 8

1 = Rx Ring

2 = Rx Tip

4 = Tx Ring

5 = Tx Tip

3, 6, 7, 8 not connected body = shield

Installing the Ethernet Connection

When initializing (connecting the VCX V7122 to the network for the first time) use a standard

Ethernet cable to connect the network interface on your computer to a port on a network hub/switch. Use a second standard Ethernet cable to connect the VCX V7122 to another port on the same network hub/switch.

For normal use, connect a standard Category 5 network cable to the Ethernet RJ-45 port

(and the other as optional redundancy/backup). Connect the other end of the Category 5 network cables to your IP network. The Ethernet connectors (labeled Ethernet 1 and

Ethernet 2) are wired according to Figure 12 .

Note that for redundant operation it is recommended to connect each of the Ethernet connectors to a different Switch.

Figure 12

Pinout of RJ-45 Connectors

RJ-45 LAN Connector and Pinout

1 2 3 4 5 6 7 8

1 = Tx+

2 = Tx-

3 = Rx+

6 = Rx-

4, 5, 7, 8 not connected

Connecting the Power Supply

Connect the VCX V7122 to the power supply using one of the following methods:

Connecting the AC Power Supply

When using a single AC power cable:

Attach one end of the supplied 100/240 VAC power cable to the rear AC socket and connect the other end to the correct grounded AC power supply.

When using a dual AC power cable:

Attach one end of the supplied 100/240 VAC power cables to the rear AC sockets and connect the other end to a separate grounded mains circuits (for power source redundancy).

3Com VCX V7122 SIP VoIP Gateway User Manual 31

CAUTION For the dual AC power supply note the following:

The LED on the left side of the chassis is only connected when the dual AC is used. It is not relevant to the single AC power connection.

If only a single socket is connected to the AC power, (while the other plug is left unconnected) the chassis’ LED (on the left side) is lit Red, indicating that one of the dual power inlets is disconnected.

When both the AC power cables are connected, one of the plugs can be disconnected under power without affecting operation, in which case the chassis’ left LED is lit Red.

UPS can be connected to either (or both) of the AC connections.

The dual AC connections operate in a 1 + 1 configuration and provide load-sharing redundancy.

Each of the dual power cables can be connected to different AC power phases.

Connecting the DC Power Supply

To connect the VCX V7122 to a DC power supply use one of these two options:

DC Terminal block with a screw connection type.

DC Terminal block with a crimp connection type.

When using a DC terminal block screw connector, follow these steps:

1 Create a DC cable by inserting two 14-16 AWG insulated wires into the supplied adaptor

(see Figure 13 ) and fasten the two screws, each one located directly above each wire.

2 Connect the two insulated wires to the correct DC power supply. Ensure that the connections to the DC power supply maintain the correct polarity.

3 Insert the terminal block into the DC inlet located on the VCX V7122.

Figure 13

DC Terminal Block Screw Connector

32 3Com VCX V7122 SIP VoIP Gateway User Manual

When using a DC terminal block crimp connector, follow these steps:

1 Remove the DC adaptor (screw connection type) that is attached to the VCX V7122 rear panel.

2 Connect the two insulated wires to the correct DC power supply. Ensure that the connections to the DC power supply maintain the correct polarity (see Figure 14 ).

3 Insert the terminal block into the DC inlet located on the VCX V7122.

Figure 14

DC Terminal Block Crimp Connector

2 screws for connecting the crimp terminal block to the VCX

V7122 rear panel

3Com VCX V7122 SIP VoIP Gateway User Manual 33

34 3Com VCX V7122 SIP VoIP Gateway User Manual

C

HAPTER

4: G

ETTING

S

TARTED

The VCX V7122 is supplied with application software already resident in its flash memory

(with factory default parameters).

Assigning the VCX V7122 IP Address below describes how to assign IP addresses to the

VCX V7122, while Assigning an IP Address Using HTTP describes how to set up the VCX

V7122 with basic parameters using a standard Web browser (such as Microsoft Internet

Explorer).

For detailed information on how to fully configure the gateway see the Web Interface, described in Chapter 5: Web Management on page 41 .

Assigning the VCX V7122 IP Address

The VCX V7122 is composed of one or two identical media gateway modules. These media gateways are fully independent, each gateway having its own MAC and IP addresses

( Table 15 shows the default IP addresses of the VCX V7122). To assign an IP address to each of the VCX V7122 modules use one of the following methods:

HTTP using a Web browser (see Assigning an IP Address Using HTTP below).

(see on page 36 ).

Dynamic Host Control Protocol (DHCP) (see DHCP Support on page 169 ).

You can use the ‘Reset’ button to restore the VCX V7122 networking parameters to their factory default values (see Restoring Networking Parameters to their Initial State on page 37 ).

Table 14

VCX V7122 Default Networking Parameters

VCX V7122 Version

Single module (up to 8 Trunks)

Double module (up to 16 Trunks)

Default Value

10.1.10.10

10.1.10.10 (Trunks 1-8) and 10.1.10.11 (Trunks 9-16)

Default subnet mask is 255.255.0.0, default gateway IP address is 0.0.0.0

Assigning an IP Address Using HTTP

To assign an IP address using HTTP, follow these steps:

1 Connect your computer to the VCX V7122. Either connect the network interface on your computer to a port on a network hub/switch (see Installing the Ethernet Connection on page 31 ), or use an Ethernet cross-over cable to directly connect the network interface on your computer to one of the RJ-45 jacks on the VCX V7122.

2 Change your PC’s IP address and subnet mask to correspond with the VCX V7122 factory default IP address and subnet mask, shown in Table 15 on page 48 . For details

3Com VCX V7122 SIP VoIP Gateway User Manual 35

on changing the IP address and subnet mask of your PC, see Windows Online Help

(Start>Help).

3 Access the VCX V7122 first module’s Embedded Web Server (see Accessing the

Embedded Web Server on page 42 ).

4 In the ‘Quick Setup’ screen (see

Figure 15 on page 38 ), set the VCX V7122 ‘IP Address’,

‘Subnet Mask’ and ‘Default Gateway IP Address’ fields under ‘IP Configuration’ to correspond with your network IP settings. If your network doesn’t feature a default gateway, enter a dummy value in the ‘Default Gateway IP Address’ field.

5 Click the Reset button and click OK in the prompt. The VCX V7122 applies the changes and restarts. This takes approximately 3 minutes to complete. When the VCX V7122 has finished restarting, the Ready and LAN LEDs on the front panel are lit green.

Record and retain the IP address and subnet mask you assign the VCX V7122. Do the same when defining new username or password. If the Embedded Web Server is unavailable (for example, if you’ve lost your username and password), use the

BootP/TFTP (Trivial File Transfer Protocol) configuration utility to access the device, “reflash” the load and reset the password (see Appendix B: The

BootP/TFTP Configuration Utility on page 191 for detailed information on using a

BootP/TFTP configuration utility to access the device).

6 Repeat steps 3 to 5 for the VCX V7122 second module (if used).

7 Disconnect your computer from the VCX V7122 or from the hub / switch (depending on the connection method you used in step 1 ).

8 Reconnect the VCX V7122 and your PC (if necessary) to the network.

9 Restore your PC’s IP address and subnet mask to what they originally were. If necessary, restart your PC and re-access the VCX V7122 via the Embedded Web

Server with its new assigned IP address.

Assigning an IP Address Using BootP

BootP procedure can also be performed using any standard compatible BootP server.

You can also use BootP to load the auxiliary files to the VCX V7122 (see Dynamic

Jitter Buffer Operation on page 136 ).

To assign an IP address using BootP, follow these steps:

1 Open the BootP application (supplied with the VCX V7122 software package).

2 Add client configuration for the gateway that you want to initialize, see

Adding Clients on page 199 .

3 Reset the gateway physically causing it to use BootP; the VCX V7122 changes its network parameters to the values provided by the BootP.

4 Repeat steps 2 and 3 for the VCX V7122 second module (if used).

36 3Com VCX V7122 SIP VoIP Gateway User Manual

Restoring Networking Parameters to Their Initial State

You can use the ‘Reset’ button to restore the VCX V7122 networking parameters to their factory default values (described in Table 14 on page 35 ) and to reset the username and password.

Note that this process also restores the VCX V7122 parameters to their factory settings, therefore you must load your previously backed-up ini file, or the default ini file (received with the software kit) to set them to their correct values.

This option is currently supported on one media gateway module (trunks 1-8) only.

To restore networking parameters to their initial state, follow these steps:

1 Disconnect the VCX V7122 from the power and network cables.

2 Reconnect the power cable; the gateway is powered up. After approximately 45 seconds, the ACT LED blinks for about 4 seconds.

3 While the ACT LED is blinking, press shortly on the reset button (located on the front panel); the gateway resets a second time and is restored with factory default parameters

(username: “Admin”, password: “Admin”).

4 Reconnect the network cable.

5 Load your previously backed-up ini file, or the default ini file (received with the software kit). To load the ini file via the Embedded Web Server, see Restoring and Backing Up the

Gateway Configuration on page 72 .

Configuring the VCX V7122 Basic Parameters

To configure the VCX V7122 basic parameters use the Embedded Web Server’s ‘Quick

Setup’ screen (see Figure 15 below). See Accessing the Embedded Web Server on page 42 for information on accessing the ‘Quick Setup’ screen.

3Com VCX V7122 SIP VoIP Gateway User Manual 37

Figure 15

VCX V7122 Quick Setup Screen

To configure basic SIP parameters, follow these steps:

1 If the VCX V7122 is behind a router with NAT (Network Address Translation) enabled, perform the following procedure. If it isn’t, leave the ‘NAT IP Address’ field undefined.

Determine the “public” IP address assigned to the router (by using, for instance, router Web management). Enter this public IP address in the ‘NAT IP Address’ field.

Enable the DMZ (Demilitarized Zone) configuration on the residential router for the

LAN port where the VCX V7122 gateway is connected. This enables unknown packets to be routed to the DMZ port.

2 Under ‘SIP Parameters’, enter the VCX V7122 domain name in the field ‘Gateway

Name’. If the field is not specified, the VCX V7122 IP address is used instead (default).

3 When working with a Proxy server, set ‘Working with Proxy’ field to ‘Yes’ and enter the IP address of the primary Proxy server in the field ‘Proxy IP address’. When no Proxy is used, the internal routing table is used to route the calls.

4 Enter the Proxy name in the field ‘Proxy Name’. If Proxy name is used, it replaces the

Proxy IP address in all SIP messages. This means that messages is still sent to the physical Proxy IP address but the SIP URI contains the Proxy name instead.

5 Configure ‘Enable Registration’ to ‘Yes’ or ‘No’:

‘No’ = the VCX V7122 does not register to a Proxy server/Registrar (default).

‘Yes’ = the VCX V7122 registers to a Proxy server/Registrar at power up and every

‘Registration Time’ seconds. For detailed information on the parameter ‘Registration

Time’, see Table 26 on page 103 .

6 Select the coder (i.e., vocoder) that best suits your VoIP system requirements. The default coder is: G.7231 30 msec. To program the entire list of coders you want the VCX

V7122 to use, click the button on the left side of the ‘1st Coder’ field; the drop-down lists

38 3Com VCX V7122 SIP VoIP Gateway User Manual

for the 2nd to 5th coders appear. Select coders according to your system requirements.

Note that coders higher on the list take precedence over coders lower on the list.

The preferred coder is the coder that the VCX V7122 uses as a first choice for all connections. If the far end gateway does not use this coder, the VCX V7122 negotiates with the far end gateway to select a coder that both sides can use.

7 To program the Tel to IP Routing Table, press the arrow button next to ‘Tel to IP Routing

Table’. For information on how to configure the Tel to IP Routing Table, see Tel to IP

Routing Table on page 52 .

8 To program the E1/T1 B-channels, press the arrow button next to ‘Trunk Group Table’.

For information on how to configure the Trunk Group Table, see Configuring the Trunk

Group Table on page 62 .

9 Click the Reset button and click OK in the prompt; The VCX V7122 applies the changes and restarts, taking approximately 3 minutes to complete. When the VCX V7122 has finished restarting, the Ready and LAN LEDs on the front panel are lit green.

10 After the gateway was reset, access the Advanced Configuration>Trunk Settings page, and select the gateway’s E1/T1 protocol type and Framing method that best suits your system requirements. Note that for E1 spans, the framing method must always be set to

‘Extended Super Frame’. For information on how to configure the Trunk Settings see

Configuring the Trunk Settings on page 69 .

You are now ready to start using the gateway. To prevent unauthorized access to the VCX

V7122, it is recommended that you change the username and password that are used to access the Web Interface. See Changing the VCX V7122 Username and Password on page 75 for details on how to change the username and password.

Once the gateway is configured correctly, back up your settings by making a copy

of the VoIP gateway configuration (ini file) and store it in a directory on your

computer to restore configuration settings at a future time. See Restoring and

Backing Up the Gateway Configuration on page 72 .

3Com VCX V7122 SIP VoIP Gateway User Manual 39

40 3Com VCX V7122 SIP VoIP Gateway User Manual

C

HAPTER

5: W

EB

M

ANAGEMENT

Configuration Concepts

Customers can use the VCX V7122 in a wide variety of applications, enabled by its parameters and configuration files (e.g., Call Progress Tones (CPT), etc.). The parameters can be configured and configuration files can be loaded using:

A standard Web Browser (described and explained in this section).

A configuration file referred to as the ini file. For information on how to use the ini file see

Chapter 6: ini File Configuration of the VCX V7122 on page 91 .

An SNMP browser software (see Chapter 11: SNMP-Based Management on page 173 ).

3Com’s Element Management System (EMS) (see Element Management System on page 185 ).

To upgrade the VCX V7122 (load new software or configuration files onto the gateway) use the Software Upgrade wizard, available through the Web Interface (see Software Upgrade

Wizard on page 81 ), or alternatively use the BootP/TFTP configuration utility (see Upgrading the VCX V7122 on page 169 ).

For information on the configuration files see Chapter 7: Configuration Files on page 139 .

Overview of the Embedded Web Server

The Embedded Web Server is used both for gateway configuration, including loading of configuration files, and for run-time monitoring. The Embedded Web Server can be accessed from a standard Web browser, such as Microsoft™ Internet Explorer, Netscape™ Navigator, etc. Specifically, Users can employ this facility to set up the gateway configuration parameters. Users also have the option to remotely reset the gateway and to permanently apply the new set of parameters.

Computer Requirements

To use the Web Interface, the following is needed:

A computer capable of running your Web browser.

A network connection to the VoIP gateway.

One of the following compatible Web browsers:

Microsoft Internet Explorer (version 6.0 and higher).

Netscape Navigator (version 7.0 and higher).

Some Java-script applications are not supported in Netscape.

3Com VCX V7122 SIP VoIP Gateway User Manual 41

Password Control

The Embedded Web Server is protected by a unique username and password combination.

The first time a browser request is made, the User is requested to provide his username and password to obtain access. Subsequent requests are negotiated by the browser on behalf of the User, so that the User doesn’t have to re-enter the username and password for each request, but the request is still authenticated (the Embedded Web Server uses the MD5 authentication method supported by the HTTP 1.1 protocol).

An additional level of protection is obtained by a restriction that no more than three IP addresses can access the Embedded Web Server concurrently. With this approach, a fourth

User is told that the server is busy, even if the correct username and password were provided.

Embedded Web Server Username and Password

The default username and password for all gateways are:

Username (case-sensitive)

Password = “Admin” (case-sensitive)

For details on changing the username and password, see Changing the VCX V7122

Username and Password on page 75 . Note that the password and username can be a maximum of 7 case-sensitive characters.

The User can reset the Web username and password (to the default values) by enabling an

ini file parameter called ‘ResetWebPassword’. The Web password is automatically the default password.

Configuring the Web Interface via the ini File

Two additional security preferences can be configured using ini file parameters. These security levels provide protection against unauthorized access (such as Internet hacker attacks), particularly to Users without a firewall. For information on the ini file see Chapter 6: ini File Configuration of the VCX V7122 on page 91 .

Limiting the Embedded Web Server to Read-Only Mode

Users can limit the Web Interface to read-only mode by changing the ini file parameter

‘DisableWebConfig’ to 1. In this mode all Web screens are read-only and cannot be modified. In addition, the following screens cannot be accessed: ‘Quick Setup’, ‘Change

Password’, ’Reset‘, ‘Save Configuration‘, ‘Software Upgrade Wizard’, ‘Load Auxiliary Files’,

‘Configuration File’ and ‘Regional Settings’.

Disabling the Embedded Web Server

To deny access to the gateway through HTTP protocol, the User can disable the Embedded

Web Server task. To disable the Web task, use the ini file parameter ‘DisableWebTask = 1’.

The default is to Web task enabled.

Accessing the Embedded Web Server

To access the Embedded Web Server, follow these steps:

42 3Com VCX V7122 SIP VoIP Gateway User Manual

1 Open a standard Web-browsing application such as Microsoft™ Internet Explorer™

(Version 6.0 and higher) or Netscape™ Navigator™ (Version 7.0 and higher).

2 In the Uniform Resource Locator (URL) field, specify the IP address of the VCX V7122

(e.g., http://10.1.10.10); the Embedded Web Server’s ‘Enter Network Password’ screen appears, shown in Figure 16 .

Figure 16

Embedded Web Server Login Screen

3 In the ‘User Name’ and ‘Password’ fields, enter the username (default: “Admin”) and password (default: “Admin”). Note that the username and password are case-sensitive.

4 Click the OK button; the ‘Quick Setup’ screen is accessed (see

Figure 15 on page 38 ).

Using Internet Explorer to Access the Embedded Web Server

Internet Explorer’s security settings may block access to the gateway’s Web browser if they’re configured incorrectly. In this case, the following message is displayed:

Unauthorized

Correct authorization is required for this area. Either your browser does not perform authorization or your authorization has failed. RomPager server.

To troubleshoot blocked access to Internet Explorer, follow these steps:

1 Delete all cookies from the Temporary Internet files. If this does not clear up the problem, the security settings may need to be altered (see Step 2).

2 In Internet Explorer, Tools, Internet Options select the Security tab, and then select

Custom Level. Scroll down until the Logon options are displayed and change the setting to Prompt for username and password and then restart the browser. This fixes any issues related to domain use logon policy.

Getting Acquainted with the Web Interface

Figure 17 shows the general layout of the Web Interface screen.

3Com VCX V7122 SIP VoIP Gateway User Manual 43

Figure 17

VCX V7122 Web Interface

Main Menu

Bar

Submenu

Bar

Title Bar

MG Module

Corporate

Logo

Main Action

Frame

Control

Protocol

The Web Interface screen features the following components:

Title bar - contains three configurable elements: corporate logo, a background image and the product’s name. For information on how to modify these elements see

Appendix F: Customizing the VCX V7122 Web Interface on page 213 .

Main menu bar - always appears on the left of every screen to quickly access parameters, submenus, submenu options, functions and operations.

Submenu bar - appears on the top of screens and contains submenu options.

Main action frame - the main area of the screen in which information is viewed and configured.

Corporate logo –3Com’s corporate logo. For information on how to remove this logo see Appendix F: Customizing the VCX V7122 Web Interface on page 213 .

Control Protocol – the VCX V7122 control protocol.

MG Module – the VCX V7122 media gateway module (Module 1 or Module 2).

Main Menu Bar

The main menu bar of the Web Interface is divided into the following 7 menus:

Quick Setup – Use this menu to configure the gateway’s basic settings; for the full list of configurable parameters go directly to ‘Protocol Management’ and ‘Advanced

Configuration’ menus. An example of the Quick Setup configuration is described in

Restoring Networking Parameters to their Initial State on page 37 .

Protocol Management – Use this subdivided menu to configure the gateway’s control protocol parameters and tables (see Protocol Management on page 45 ).

44 3Com VCX V7122 SIP VoIP Gateway User Manual

Advanced Configuration – Use this subdivided menu to set the gateway’s advanced configuration parameters (for advanced users only) (see Advanced Configuration on page 65 ).

Status & Diagnostics – Use this subdivided menu to view and monitor the gateway’s channels, Syslog messages, hardware / software product information, and to assess the gateway’s statistics and IP connectivity information (see Status & Diagnostic on page 75 ).

Software Update – Use this subdivided menu when you want to load new software or configuration files onto the gateway (see Software Update Menu on page 81 ).

Save Configuration – Use this menu to save configuration changes to the non-volatile flash memory (see Save Configuration on page 88 ).

Reset – Use this menu to remotely reset the gateway. Note that you can choose to save the gateway configuration to flash memory before reset (see Save Configuration on page 88 ).

When positioning your curser over a parameter name (or a table) for more than 1 second, a short description of this parameter is displayed. Note that those parameters that are preceded with an exclamation mark (!) are Not changeable on-the-fly and require reset.

Saving Changes

To save changes to the volatile memory (RAM) press the Submit button (changes to parameters with on-the-fly capabilities are immediately available, other parameter are updated only after a gateway reset). Parameters that are only saved to the volatile memory revert to their previous settings after hardware reset. When performing a software reset (i.e., via Web or SNMP) you can choose to save the changes to the non-volatile memory. To save changes so they are available after a power fail, you must save the changes to the nonvolatile memory (flash). When Save Configuration is performed, all parameters are saved to the flash memory.

To save the changes to flash, follow these steps:

1 Click the Save Configuration button; the ‘Save Configuration to Flash Memory’ screen appears.

2 Click the Save Configuration button in the middle of the screen; a confirmation message appears when the save is complete.

Note: When you reset the VCX V7122 from the Web Interface, you can choose to save the configuration to flash memory.

Entering Phone Numbers in Various Tables

Phone numbers entered into various tables on the gateway, such as the Tel to IP routing table, must be entered without any formatting characters. For example, if you wish to enter the phone number 555-1212, it must be entered as 5551212 without the hyphen (-). If the hyphen is entered, the entry does not work. The hyphen character is used in number entry only, as part of a range definition. For example, the entry [20-29] means “all numbers in the range 20 to 29”.

Protocol Management

Use this subdivided menu to configure the gateway’s SIP parameters and tables.

3Com VCX V7122 SIP VoIP Gateway User Manual 45

Protocol Definition Parameters

Use this submenu to configure the following gateway’s specific SIP protocol parameters:

Proxy & Registration Parameters

(see below)

DTMF & Dialing Parameters

Coders

From the Coders screen you can configure the first to fifth preferred coders (and their corresponding ptimes) for the gateway. The first coder is the highest priority coder and is used by the gateway whenever possible. If the far end gateway cannot use the coder assigned as the first coder, the gateway attempts to use the next coder and so forth.

To configure the gateway’s coders, follow these steps:

1 Open the ‘Coders’ screen (Protocol Management menu > Protocol Definition submenu > Coders option); the ‘Coders’ screen is displayed.

Figure 18

Coders Screen

2 From the coder drop-down list, select the coder you want to use. For the full list of available coders and their corresponding ptimes see the ini file parameter ‘CoderName’

(described in Table 27 on page 103 ).

Note: Each coder can appear only once.

3 From the drop-down list to the right of the coder list, select the size of the Voice Packet

(ptime) used with this coder in milliseconds. Selecting the size of the packet determines how many coder payloads are combined into one RTP (Real-Time Transport Protocol)

(voice) packet.

Note 1: The ptime packetization period depends on the selected coder name.

Note 2: If not specified, the ptime gets a default value.

Note 3: The ptime specifies the maximum packetization time the gateway can receive.

4 Repeat steps 2 and 3 for the second to fifth coders (optional).

5 Click the Submit button to save your changes.

6 To save the changes so they are available after a power fail see Save Configuration on page 88 .

Only the ptime of the first coder in the defined coder list is declared in Invite/200

OK SDP, even if multiple coders are defined.

46 3Com VCX V7122 SIP VoIP Gateway User Manual

Advanced Parameters

Use this submenu to configure the following gateway’s advanced control protocol parameters.

Disconnect and Answer Supervision

CDR and Debug

Number Manipulation Tables

The VoIP gateway provides four Number Manipulation tables for incoming and outgoing calls. These tables are used to modify the destination and source telephone numbers so that the calls can be routed correctly.

The Manipulation Tables are:

Destination Phone Number Manipulation Table for IP Tel calls

Destination Phone Number Manipulation Table for Tel IP call

Source Phone Number Manipulation Table for IP Tel calls

Source Phone Number Manipulation Table for Tel IP calls

Number manipulation can be performed either before or after a routing decision is made. For example, you can route a call to a specific trunk group according to its original number, and then you can remove/add a prefix to that number before it is routed. To control when number manipulation is done, set the ‘RouteModeIP2Tel’ and the ‘RouteModeTel2IP’ parameters. For information on these parameters, see

Table 28 on page 119 .

Possible uses for number manipulation can be as follows:

To strip/add dialing plan digits from/to the number. For example, a user could dial 9 in front of each number to indicate an external line. This number (9) can be removed here before (after) the call is setup.

Assignment of NPI/TON to IP Tel calls. The VoIP gateway can use a single global setting for NPI/TON classification or it can use the setting in this table on a call by call basis. Control for this is done using “Protocol Management>Protocol

Definition>Destination/Source Number Encoding Type”.

Allow / disallow Caller ID information to be sent according to destination / source prefixes.

To configure the Number Manipulation tables, follow these steps:

1 Open the Number Manipulation screen you want to configure (Protocol Management menu > Manipulation Tables submenu); the relevant Manipulation table screen is displayed. Figure 19 shows the ‘Source Phone Number Manipulation Table for Tel IP calls’.

3Com VCX V7122 SIP VoIP Gateway User Manual 47

Figure 19

Source Phone Number Manipulation Table for Tel IP Calls

2 In the ‘Table Index’ drop-down list, select the range of entries that you want to edit (up to

20 entries can be configured for Source Number Manipulation and 50 entries for

Destination Number Manipulation).

3 Configure the Number Manipulation table according to Table 15 .

4 Click the Submit button to save your changes.

5 To save the changes so they are available after a power fail see

Save Configuration on page 88 .

Table 15

Number Manipulation Parameters

Parameter Description

Destination Prefix

Each entry in the Destination Prefix fields represents a destination telephone number prefix.

Source Prefix

Source IP

Each entry in the Source Prefix fields represents a source telephone number prefix.

Each entry in the Source IP fields represents the source IP address of the call.

This column only applies to the ‘Destination Phone Number Manipulation

Table for IP to Tel’.

Note: The source IP address can include the “x” wildcard to represent single digits. For example: 10.8.8.xx represents all the addresses between

10.8.8.10 to 10.8.8.99.

48 3Com VCX V7122 SIP VoIP Gateway User Manual

Table 15

Number Manipulation Parameters

Parameter Description

The manipulation rules are applied to any incoming call whose:

Destination number prefix matches the prefix defined in the ‘Destination Number’ field.

Source number prefix matches the prefix defined in the ‘Source Prefix’ field.

Source IP address matches the IP address defined in the ‘Source IP’ field (if applicable).

Note that number manipulation can be performed using a combination of each of the above criteria, or using each criterion independently.

Note: For available notations that represent multiple numbers see Dialing Plan Notation on page 50 .

Num of stripped digits

Prefix / Suffix to add

Enter the number of digits that you want to remove from the left of the telephone number prefix. For example, if you enter 3 and the phone number is 5551234, the new phone number is 1234.

Enter the number of digits (in brackets) that you want to remove from the right of the telephone number prefix.

Note: A combination of the two options is allowed (e.g., 2(3)).

Prefix - Enter the number / string you want to add to the front of the phone number. For example, if you enter 9 and the phone number is 1234, the new number is 91234.

Suffix - Enter the number / string (in brackets) you want to add to the end of the phone number. For example, if you enter (00) and the phone number is 1234, the new number is 123400.

Note: You can enter a prefix and a suffix in the same field (e.g., 9(00)).

Enter the number of digits that you want to leave from the right. Number of digits to leave

Note: The manipulation rules are executed in the following order:

1.

Num of stripped digits

2.

Number of digits to leave

3.

Prefix / suffix to add

Figure 19 on the previous page exemplifies the use of these manipulation rules in the ‘Source Phone

Number Manipulation Table for Tel IP Calls’:

When destination number equals 035000 and source number equals 20155, the source number is changed to 97220155.

When source number equals 1001876, it is changed to 587623.

Source number 1234510012001 is changed to 20018.

Source number 3122 is changed to 2312.

NPI Select the Number Plan assigned to this entry.

You can select Unknown [0], Private [9] or E.164 Public [1].

The default is Unknown.

For a detailed list of the available NPI/TON values see Numbering Plans on page 51 .

3Com VCX V7122 SIP VoIP Gateway User Manual 49

Table 15

Number Manipulation Parameters

Parameter Description

TON Select the Number Type assigned to this entry.

If you selected Unknown as the Number Plan, you can select Unknown

[0].

If you selected Private as the Number Plan, you can select Unknown [0],

Level 2 Regional [1], Level 1 Regional [2], PSTN Specific [3] or Level 0

Regional (Local) [4].

If you selected E.164 Public as the Number Plan, you can select Unknown

[0], International [1], National [2], Network Specific [3], Subscriber [4] or

Abbreviated [6].

The default is Unknown.

Presentation Select ‘Allowed’ to send Caller ID information when a call is made using these destination / source prefixes.

Select ‘Restricted’ if you want to restrict Caller ID information for these prefixes.

Dialing Plan Notation

The dialing plan notation applies, in addition to the four Manipulation tables, also to Tel IP

Routing table and to IP Trunk Group Routing table.

When entering a number in the ‘Prefix’ column, you can create an entry that represents multiple numbers using the following notation:

[n-m] represents a range of numbers

[n,m] represents multiple numbers. Note that this notation only supports single digit numbers. x represents any single digit

# represents the end of a number

For example:

[5551200-5551300]# represents all numbers from 5551200 to 5551300

[2,3,4] represents all numbers that start with the numbers 2, 3 and 4

54324 represents any number that starts with 54324

54324xx# represents a 7 digit number that starts with 54324

123[100-200]# represents all numbers from 123100 to 123200.

The VoIP gateway matches the rules starting at the top of the table. For this reason, enter more specific rules above more generic rules. For example, if you enter 551 in entry 1 and

55 in entry 2, the VoIP gateway applies rule 1 to numbers that starts with 551 and applies rule 2 to numbers that start with 550, 552, 553, 554, 555, 556, 557, 558 and 559. However if you enter 55 in entry 1 and 551 in entry 2, the VoIP gateway applies rule 1 to all numbers that start with 55 including numbers that start with 551.

50 3Com VCX V7122 SIP VoIP Gateway User Manual

Numbering Plans

Numbers are classified by their Numbering Plan Indication (NPI) and their Type of Number

(TON). The VCX V7122 supports all NPI/TON classifications used in the standard. The most important NPI/TON values are as follows:

Table 16

NPI/TON Values

Unknown [0]

E.164 Public [1]

(ISDN)

Private [9]

Unknown [0]

Unknown [0]

International [1]

National [2]

Subscriber [4]

Unknown [0]

Level 1 Regional [2]

Level 0 Regional [4]

A valid classification, but one that has no information about the numbering plan.

A public number in E.164 format, but no information on what kind of E.164 number.

A public number in complete international E.164 format. For example: 16135551234

A public number in complete national E.164 format.

For example: 6135551234

A public number in complete E.164 format representing a local subscriber. For example:

5551234

A private number, but with no further information about the numbering plan

A private number with a location. For example:

3932200

A private local extension number. For example: 2200

3Com VCX V7122 SIP VoIP Gateway User Manual 51

Configuring the Routing Tables

Use this submenu to configure the gateway’s IP Tel and Tel IP routing tables and their associated parameters.

Tel to IP Routing Table

The Tel to IP Routing Table is used to route incoming Tel calls to IP addresses. This routing table associates a called / calling telephone number’s prefixes with a destination IP address or with an FQDN (Fully Qualified Domain Name). When a call is routed through the VoIP gateway (Proxy isn’t used), the called and calling numbers are compared to the list of prefixes on the IP Routing Table (up to 50 prefixes can be configured); Calls that match these prefixes are sent to the corresponding IP address. If the number dialed does not match these prefixes, the call is not made.

When using a Proxy server, you do not need to configure the Telephone to IP Routing Table.

However, if you want to use fallback routing when communication with Proxy is lost, or to use the ‘Filter Calls to IP’ and IP Security features, or to obtain different SIP URI host names (per called number), you need to configure the IP Routing Table.

Possible uses for Telephone to IP Routing can be as follows:

Can fallback to internal routing table if there is no communication with the Proxy.

Call Restriction – (when Proxy isn’t used), reject all outgoing Tel IP calls that are associated with the destination IP address: 0.0.0.0.

IP Security – When the IP Security feature is enabled (SecureCallFromIP = 1), the VoIP gateway accepts only those IP Tel calls with a source IP address identical to one of the

IP addresses entered in the Telephone to IP Routing Table.

Filter Calls to IP – When a Proxy is used, the gateway checks the Tel IP routing table before a telephone number is routed to the Proxy. If the number is not allowed (number isn’t listed or a Call Restriction routing rule was applied), the call is released.

Always Use Routing Table – When this feature is enabled (AlwaysUseRouteTable = 1), even if a Proxy server is used, the SIP URI host name in the sent INVITE message is obtained from this table. Using this feature users are able to assign a different SIP URI host name for different called and/or calling numbers.

Assign Profiles to destination address (also when a Proxy is used).

Alternative Routing – (When Proxy isn’t used) an alternative IP destination for telephone number prefixes is available. To associate an alternative IP address to called telephone number prefix, assign it with an additional entry (with a different IP address), or use an

FQDN that resolves to two IP addresses. Call is sent to the alternative destination when one of the following occurs:

No ping to the initial destination is available, or when poor Quality of Service (QoS)

(delay or packet loss, calculated according to previous calls) is detected, or when a

DNS host name is not resolved. For detailed information on Alternative Routing, see

Configuring the Gateway’s Alternative Routing (based on Connectivity and QoS) on page 150 .

When a release reason that is defined in the ‘Reasons for Alternative Tel to IP

Routing’ table is received. For detailed information on the ‘Reasons for Alternative

Routing Tables’ see Reasons for Alternative Routing on page 57 .

52 3Com VCX V7122 SIP VoIP Gateway User Manual

Alternative routing (using this table) is commonly implemented when there is no response to an Invite message (after Invite retransmissions). The gateway then issues an internal 408

‘No Response’ implicit release reason. If this reason is included in the ‘Reasons for

Alternative Routing’ table, the gateway immediately initiates a call to the redundant destination using the next matched entry in the ‘Tel to IP Routing’ table. Note that if a domain name in this table is resolved to two IP addresses, the timeout for Invite retransmissions can be reduced by using the parameter ‘Number of RTX Before Hotswap’.

Note: If the alternative routing destination is the gateway itself, the call can be configured to be routed back to PSTN. This feature is referred to as "PSTN Fallback", meaning that if sufficient voice quality is not available over the IP network, the call is routed through legacy telephony system (PSTN).

Tel to IP routing can be performed either before or after applying the number manipulation rules. To control when number manipulation is done, set the

RouteModeTel2IP parameter. For information on this parameter, see Table 28 on page 119 .

To configure the Tel to IP Routing table, follow these steps:

1 Open the ‘Tel to IP Routing’ screen (Protocol Management menu > Routing Tables submenu > Tel to IP Routing option); the ‘Tel to IP Routing’ screen is displayed (shown in Figure 20 ).

2 In the ‘Tel to IP Routing Mode’ field, select the Tel to IP routing mode (see

Table 28 on page 119 ).

3 In the ‘Routing Index' drop-down list, select the range of entries that you want to edit.

4 Configure the Tel to IP Routing table according to Table 17 .

5 Click the Submit button to save your changes.

6 To save the changes so they are available after a power fail see Save Configuration on page 88 .

Figure 20

Tel to IP Routing Table Screen

3Com VCX V7122 SIP VoIP Gateway User Manual 53

Table 17

Tel to IP Routing Table

Parameter Description

Destination Phone Prefix Each entry in the Destination Phone Prefix fields represents a called telephone number prefix. The prefix can be 1 to 19 digits long. An asterisk

(*) represents all numbers.

Source Phone Prefix

Any telephone number whose destination number matches the prefix defined in the ‘Destination Phone

Prefix’ field and its source number matches the prefix defined in the adjacent ‘Source Phone Prefix‘ field, is sent to the IP address entered in the ‘IP Address’ field.

Note that Tel to IP routing can be performed according to a combination of source and destination phone prefixes, or using each independently.

Note 1: An additional entry of the same prefixes can be assigned to enable alternative routing.

Note 2: For available notations that represent multiple numbers see Dialing Plan Notation on page 50 .

Destination IP Address

Each entry in the Source Phone Prefix fields represents a calling telephone number prefix. The prefix can be 1 to 19 digits long. An asterisk

(*) represents all numbers.

Profile ID

Status

In each of the IP Address fields, enter the IP address that is assigned to these prefixes. Domain names, such as domain.com, can be used instead of IP addresses.

To discard outgoing IP calls, enter 0.0.0.0 in this field.

Note: When using domain names, you must enter a DNS server IP address, or alternatively define these names in the ‘Internal DNS Table’.

Enter the number of the IP profile that is assigned to the destination IP address defined in the ‘Destination IP Address’ field.

A read only field representing the quality of service of the destination IP address.

N/A = Alternative Routing feature is disabled.

OK = IP route is available

Ping Error = No ping to IP destination, route is not available

QoS Low = Bad QoS of IP destination, route is not available

DNS Error = No DNS resolution (only when domain name is used instead of an IP address).

IP to Trunk Group Routing Table

The IP to Trunk Group Routing Table is used to route incoming IP calls to groups of E1/T1

B-channels called trunk groups. Calls are assigned to trunk groups according to any combination of the following three options (or using each independently):

Destination phone prefix

Source phone prefix

Source IP address

The call is then sent to the VoIP gateway channels assigned to that trunk group. The specific channel, within a trunk group, that is assigned to accept the call is determined according to the trunk group’s channel selection mode which is defined in the Trunk Group Settings table

(see Configuring the Trunk Group Settings on page 63 ), or according to the global parameter

‘ChannelSelectMode’ (see Table 28 on page 119 ).

54 3Com VCX V7122 SIP VoIP Gateway User Manual

Note: When a release reason that is defined in the ‘Reasons for Alternative IP to Tel

Routing’ table is received for a specific IP Tel call, an alternative trunk group for that call is available. To associate an alternative trunk group to an incoming IP call, assign it with an additional entry in the ‘IP to Trunk Group Routing’ table (repeat the same routing rules with a different trunk group ID). For detailed information on the ‘Reasons for Alternative Routing

Tables’ see Reasons for Alternative Routing on page 57 .

To use trunk groups you must also do the following:

You must assign a trunk group ID to the VoIP gateway E1/T1 B-channels on the Trunk

Group Table. For information on how to assign a trunk group ID to a B-channel, see

Configuring the Trunk Group Table on page 62 .

You can configure the Trunk Group Settings table to determine the method in which new calls are assigned to channels within the trunk groups (a different method for each trunk group can be configured). For information on how to enable this option, see Configuring the Trunk Group Settings on page 63 . If a Channel Select Mode for a specific trunk group isn’t specified, then the global ‘Channel Select Mode’ parameter (defined in

‘General Parameters’ screen under ‘Advanced Parameters’) applies.

To configure the IP to Trunk Group Routing table, follow these steps:

1 Open the ‘IP to Trunk Group Routing’ screen (Protocol Management menu > Routing

Tables submenu > IP to Trunk Group Routing option); the ‘IP to Trunk Group Routing’ table screen is displayed.

Figure 21

IP to Trunk Group Routing Table

2 In the ‘IP to Tel Routing Mode’ field, select the IP to Tel routing mode (see Table 28 on page 119 ).

3 In the ‘Routing Index’ drop-down list, select the range of entries that you want to edit (up to 24 entries can be configured).

4 Configure the IP to Trunk Group Routing table according to

Table 18 .

5 Click the Submit button to save your changes.

6 To save the changes so they are available after a power fail, see

Save Configuration on page 88 .

3Com VCX V7122 SIP VoIP Gateway User Manual 55

Table 18

IP to Trunk Group Routing Table

Parameter Description

Destination Phone Prefix Each entry in the Destination Phone Prefix fields represents a called telephone number prefix. The prefix can be 1 to 49 digits long. An asterisk (*) represents all numbers.

Source Phone Prefix Each entry in the Source Phone Prefix fields represents a calling telephone number prefix. The prefix can be 1 to 49 digits long. An asterisk (*) represents all numbers.

Source IP Address Each entry in the Source IP Address fields represents the source IP address of an IP Tel call.

Note: The source IP address can include the “x” wildcard to represent single digits. For example: 10.8.8.xx represents all the addresses between 10.8.8.10 to 10.8.8.99.

Any SIP incoming call whose destination number matches the prefix defined in the ‘Destination Phone

Prefix’ field and its source number matches the prefix defined in the adjacent ‘Source Phone Prefix‘ field and its source IP address matches the address defined in the ‘Source IP Address’ field, is assigned to the trunk group entered in the field to the right of these fields.

Note that IP to trunk group routing can be performed according to any combination of source / destination phone prefixes and source IP address, or using each independently.

Note: For available notations that represent multiple numbers (used in the prefix columns), see Dialing Plan

Notation on page 50 .

Trunk Group ID

Profile ID

In each of the Trunk Group ID fields, enter the trunk group ID to which calls that match these prefixes are assigned.

Enter the number of the IP profile that is assigned to the routing rule.

Internal DNS Table

The internal DNS table, similar to a DNS resolution, translates hostnames into IP addresses.

This table is used when hostname translation is required (e.g., ‘Tel to IP Routing’ table, etc.).

Two different IP addresses can be assigned to the same hostname. If the hostname isn’t found in this table, the gateway communicates with an external DNS server.

Assigning two IP addresses to hostname can be used for alternative routing (using the ‘Tel to

IP Routing’ table).

To configure the internal DNS table, follow these steps:

1 Open the ‘Internal DNS Table’ screen (Protocol Management menu > Routing Tables submenu > Internal DNS Table option); the ‘Internal DNS Table’ screen is displayed.

56 3Com VCX V7122 SIP VoIP Gateway User Manual

Figure 22

Internal DNS Table Screen

2 In the ‘DNS Name’ field, enter the hostname to be translated. You can enter a string up to 31 characters long.

3 In the ‘First IP Address’ field, enter the first IP address that the hostname is translated to.

4 In the ‘Second IP Address’ field, enter the second IP address that the hostname is translated to.

5 Repeat steps 2 to 4, for each Internal DNS Table entry.

6 Click the Submit button to save your changes.

7 To save the changes so they are available after a power fail, see Save Configuration on page 88 .

Reasons for Alternative Routing

The Reasons for Alternative Routing screen includes two tables (Tel IP and IP Tel). Each table enables you to define up to 4 different release reasons. If a call is released as a result of one of these reasons, the gateway tries to find an alternative route to that call. The release reason for IP Tel calls is provided in Q.931 notation. The release reason for Tel IP calls is provided in SIP 4xx, 5xx and 6xx response codes. For Tel IP calls an alternative IP address, for IP Tel calls an alternative trunk group.

See the ‘ Tel to IP Routing Table ’ on page 52 for information on defining an alternative IP address. See the ‘ IP to Trunk Group Routing Table ’ on page 54 for information on defining an alternative trunk group.

You can use this table for example:

For Tel IP calls, when there is no response to an Invite message (after Invite retransmissions), and the gateway then issues an internal 408 ‘No Response’ implicit release reason.

For IP Tel calls, when the destination is busy, and release reason #17 is issued or for other call releases that issue the default release reason (#3), see ‘DefaultReleaseCause’ in

Table 26 on page 103 .

Note: The reasons for alternative routing option for Tel IP calls only applies when Proxy isn’t used.

To configure the reasons for alternative routing, follow these steps:

1 Open the ‘Reasons for Alternative Routing’ screen (Protocol Management menu >

Routing Tables submenu > Reasons for Alternative Routing option); the ‘Reasons for

Alternative Routing’ screen is displayed.

3Com VCX V7122 SIP VoIP Gateway User Manual 57

Figure 23

Reasons for Alternative Routing Screen

2 In the ‘IP to Tel Reasons’ table, from the drop-down list select up to 4 different call failure reasons that invoke an alternative IP to Tel routing.

3 In the ‘Tel to IP Reasons’ table, from the drop-down list select up to 4 different call failure reasons that invoke an alternative Tel to IP routing.

4 Click the Submit button to save your changes.

5 To save the changes so they are available after a power fail, see Save Configuration on page 88 .

Configuring the Profile Definitions

Utilizing the Profiles feature, the VCX V7122 provides high-level adaptation when connected to a variety of equipment (from both Tel and IP sides) and protocols, each of which require a different system behavior. Using Profiles, users can assign different Profiles (behavior) on a per-call basis, using the Tel to IP and IP to Trunk Group Routing tables, or associate different Profiles to the gateway’s B-channels(s). The Profiles contain parameters such as

Coders, T.38 Relay, Voice and DTMF Gains, Silence Suppression, Echo Canceler, RTP

DiffServ and more. The Profiles feature allows users to tune these parameters or turn them on or off, per source or destination routing and/or the specific gateway or its ports. For example, specific E1/T1 spans can be designated for to have a profile which always uses

G.711.

Each call can be associated with one or two Profiles, Tel Profile and (or) IP Profile. If both IP and Tel profiles apply to the same call, the coders and other common parameters of the preferred Profile (determined by the Preference option) are applied to that call. If the

Preference of the Tel and IP Profiles is identical, the Tel Profile parameters are applied.

The default values of the parameters in the Tel and IP Profiles are identical to the

Web/ini file parameter values. If a value of a parameter is changed in the Web/ini

file, it is automatically updated in the Profiles correspondingly. After any parameter in the Profile is modified by the user, modifications to parameters in the Web/ini file no longer impact that Profile.

58 3Com VCX V7122 SIP VoIP Gateway User Manual

Coder Group Settings

Use the Coders Group Settings screen to define up to four different coder groups. These coder groups are used in the Tel and IP Profile Settings screens to assign different coders to

Profiles.

To configure the coder group settings, follow these steps:

1 Open the ‘Coder Group Settings’ screen (Protocol Management menu > Profile

Definitions submenu > Coder Group Settings option); the ‘Coder Group Settings’ screen is displayed.

Figure 24

Coder Group Settings Screen

2 In the ‘Coder Group ID’ drop-down list, select the coder group you want to edit (up to four coder groups can be configured).

3 From the coder drop-down list, select the coder you want to use. For the full list of available coders and their corresponding ptimes see the ini file parameter

‘CoderName_ID’ (described in Table 26 on page 103 ).

Note: Each coder can appear only once.

4 From the drop-down list to the right of the coder list, select the size of the Voice Packet

(ptime) used with this coder in milliseconds. Selecting the size of the packet determines how many coder payloads are combined into one RTP (voice) packet.

Note 1: The ptime packetization period depends on the selected coder name.

Note 2: If not specified, the ptime gets a default value.

Note 3: The ptime specifies the maximum packetization time the gateway can receive.

5 Repeat steps 3 and 4 for the second to fifth coders (optional).

6 Repeat steps 2 to 5 for the second to forth coder groups (optional).

7 Click the Submit button to save your changes.

8 To save the changes so they are available after a power fail see Save Configuration on page 88 .

In the current version, only the ptime of the first coder is sent in the SDP section of the Invite message.

Tel Profile Settings

Use the Tel Profile Settings screen to define up to four different Tel Profiles. These Profiles are used in the ‘Trunk Group’ table to associate different Profiles to gateway’s B-channels, thereby applying different behavior to different VCX V7122 B-channels.

3Com VCX V7122 SIP VoIP Gateway User Manual 59

To configure the Tel Profile settings, follow these steps:

1 Open the ‘Tel Profile Settings’ screen (Protocol Management menu > Profile

Definitions submenu > Tel Profile Settings option); the ‘Tel Profile Settings’ screen is displayed.

Figure 25

Tel Profile Settings Screen

2 In the ‘Profile ID’ drop-down list, select the Tel Profile you want to edit (up to four Tel

Profiles can be configured).

3 In the ‘Profile Preference’ drop-down list, select the preference (1-10) of the current

Profile. The preference option is used to determine the priority of the Profile. If both IP and Tel profiles apply to the same call, the coders and other common parameters of the preferred Profile are applied to that call. If the Preference of the Tel and IP Profiles is identical, the Tel Profile parameters are applied.

Note: If the coder lists of both IP and Tel Profiles apply to the same call, an intersection of the coders is performed (i.e., only common coders remain). The order of the coders is determined by the preference.

4 Configure the Profile’s parameters according to your requirements. For detailed information on each parameter, see the description of the screen in which it is configured as an individual parameter.

5 In the ‘Coder Group’ drop-down list, select the coder group you want to assign to that

Profile. You can select the gateway’s default coders (see Coders on page 46 ) or one of the coder groups you defined in the Coder Group Settings screen (see Coder Group

Settings on page 59 ).

6 Repeat steps 2 to 6 for the second to fifth Tel Profiles (optional).

60 3Com VCX V7122 SIP VoIP Gateway User Manual

7 Click the Submit button to save your changes.

8 To save the changes so they are available after a power fail, see Save Configuration on page 88 .

IP Profile Settings

Use the IP Profile Settings screen to define up to four different IP Profiles. These Profiles are used in the Tel to IP and IP to Trunk Group Routing tables to associate different Profiles to routing rules. IP Profiles can also be used when working with Proxy server (set

‘AlwaysUseRouteTable’ to 1).

To configure the IP Profile settings, follow these steps:

1 Open the ‘IP Profile Settings’ screen (Protocol Management menu > Profile

Definitions submenu > IP Profile Settings option); the ‘IP Profile Settings’ screen is displayed.

Figure 26

IP Profile Settings Screen

2 In the ‘Profile ID’ drop-down list, select the IP Profile you want to edit (up to four IP

Profiles can be configured).

3 In the ‘Profile Preference’ drop-down list, select the preference (1-10) of the current

Profile. The preference option is used to determine the priority of the Profile. If both IP and Tel profiles apply to the same call, the coders and other common parameters of the preferred Profile are applied to that call. If the Preference of the Tel and IP Profiles is identical, the Tel Profile parameters are applied.

Note: If the coder lists of both IP and Tel Profiles apply to the same call, an intersection of the coders is performed (i.e., only common coders remain). The order of the coders is determined by the preference.

3Com VCX V7122 SIP VoIP Gateway User Manual 61

4 Configure the Profile’s parameters according to your requirements. For detailed information on each parameter, see the description of the screen in which it is configured as an individual parameter.

5 In the ‘Coder Group’ drop-down list, select the coder group you want to assign to that

Profile. You can select the gateway’s default coders (see Coders on page 46 ) or one of the coder groups you defined in the Coder Group Settings screen (see Coder Group

Settings on page 59 ).

6 Repeat steps 2 to 6 for the second to fifth IP Profiles (optional).

7 Click the Submit button to save your changes.

8 To save the changes so they are available after a power fail, see Save Configuration on page 88 .

Configuring the Trunk Group Table

Use the Trunk Group table to assign trunk groups, profiles and logical telephone numbers to the gateway's E1/T1 B-channels. Trunk Groups are used for routing IP Tel calls with common rules. Channels that are not defined are disabled.

To configure the Trunk Group table, follow these steps:

1 Open the ‘Trunk Group Table’ screen (Protocol Management menu > Trunk Group); the ‘Trunk Group Table’ screen is displayed.

Figure 27

Trunk Group Table Screen

2 Configure the Trunk Group according to Table 19 .

3 Click the Submit button to save your changes.

4 To save the changes so they are available after a power fail, see Save Configuration on page 88 .

Table 19

Trunk Group Table

Parameter Description

Trunk ID The numbers (1-8) in the Trunk ID drop-down list represent the physical trunks on the back of the VoIP gateway.

Channels To enable the trunk’s B-channels, you must enter their number in this field.

[n-m] represents a range of channels.

For example, enter [1-24] to specify the channels from 1 to 24.

Note: The number of defined channels must not exceed the number of the trunk’s

B-channels (1-24 for T1 spans and 1-30 for E1 spans).

62 3Com VCX V7122 SIP VoIP Gateway User Manual

Table 19

Trunk Group Table

Parameter Description

Phone Number In each of the Phone Number fields, enter the first number in an ordered sequence that is assigned to the range of channels defined in the adjacent ‘Channels’ field.

Note: This field is optional. The logical numbers defined in this field are used when an incoming PSTN / PBX call doesn’t contain the calling number or called number

(the latter being determined by the parameter

‘ReplaceEmptyDstWithPortNumber’), these numbers are used to replace them.

These logical numbers are also used for B-channel allocation for IP to Tel calls, if the trunk group’s ‘Channel Select Mode’ is set to ‘By Phone Number’.

Trunk Group ID

Profile ID

In each of the Trunk Group ID fields, enter the trunk group ID (1-99) assigned to the channels. The same trunk group ID can be used for more than one group of channels.

Trunk group ID is used to define a group of common behavior channels that are used for routing IP to Tel calls. If an IP to Tel call is assigned to a trunk group, the call is routed to the channel or channels that correspond to the trunk group ID.

You can configure the Trunk Group Settings table to determine the method in which new calls are assigned to channels within the trunk groups (see Configuring the Trunk Group Settings below).

Note: You must configure the IP to Trunk Group Routing Table (assigns incoming

IP calls to the appropriate trunk group). If you do not configure the IP to Trunk

Group Routing Table, calls do not complete.

For information on how to configure this table, see IP to Trunk Group Routing

Table on page 54 .

Enter the number of the Tel profile that is assigned to the B-channels defined in the ‘Channels’ field.

Configuring the Trunk Group Settings

The Trunk Group Settings Table is used to determine the method in which new calls are assigned to B-channels within each trunk group. If such a rule doesn’t exist (for a specific

Trunk group), the global rule, defined by the Channel Select Mode parameter (Protocol

Definition > General Parameters), applies.

To configure the Trunk Group Settings table, follow these steps:

1 Open the ‘Trunk Group Settings’ screen (Protocol Management menu > Trunk Group

Settings); the ‘Trunk Group Settings’ screen is displayed.

3Com VCX V7122 SIP VoIP Gateway User Manual 63

Figure 28

Trunk Group Settings Screen

2 In the Routing Index drop-down list, select the range of entries that you want to edit (up to 24 entries can be configured).

3 In the Trunk Group ID field, enter the Trunk Group ID number.

4 In the Channel Select Mode drop-down list, select the Channel Select Mode that determines the method in which new calls are assigned to B-channels within the Trunk groups entered in the field to the right of this field. For information on available Channel

Select Modes see Table 20 .

5 Repeat steps 4 and 5, for each defined Trunk group.

6 Click the Submit button to save your changes.

7 To save the changes so they are available after a power fail, see

Save Configuration on page 88 .

Table 20

Channel Select Modes

Mode Description

By phone number Select the gateway port according to the called number (see the note below).

Cyclic Ascending

Ascending

Select the next available channel in an ascending cycle order.

Always select the next higher channel number in the Trunk Group.

When the gateway reaches the highest channel number in the

Trunk Group, it selects the lowest channel number in the Trunk

Group and then starts ascending again (default).

Select the lowest available channel. Always start at the lowest channel number in the Trunk Group and if that channel is not available, select the next higher channel.

64 3Com VCX V7122 SIP VoIP Gateway User Manual

Mode Description

Cyclic Descending Select the next available channel in descending cycle order.

Always select the next lower channel number in the Trunk Group.

When the gateway reaches the lowest channel number in the Trunk

Group, it selects the highest channel number in the Trunk Group and then start descending again.

Descending

Number + Cyclic Ascending

Select the highest available channel. Always start at the highest channel number in the Trunk Group and if that channel is not available, select the next lower channel.

First select the gateway port according to the called number (see the note below). If the called number isn’t found, then select the next available channel in ascending cyclic order. Note that if the called number is found, but the port associated with this number is busy, the call is released.

The internal numbers of the gateway’s B-channels are defined in the ‘Trunk Group

Table’ under the ‘Phone Number’ column. For detailed information on the ‘Trunk

Group Table’, refer see Configuring the Trunk Group Table on page 62 ).

Advanced Configuration

Use this subdivided menu to set the gateway’s advanced configuration parameters (for advanced users only).

Configuring the Network Settings

From the Network Settings page you can define:

Ethernet Ports Information (read-only).

To configure the Network Settings parameters, follow these steps:

1 Open the ‘Network Settings’ screen (Advanced Configuration menu > Network

Settings); the ‘Network Settings’ screen is displayed.

2 Configure the Network Settings parameters.

3 Click the Submit button to save your changes.

4 To save the changes so they are available after a power fail, see Save Configuration on page 88 .

3Com VCX V7122 SIP VoIP Gateway User Manual 65

Figure 29

Network Settings Screen

Note that the default RTP Base UDP Port is 6000.

Configuring the SNMP Managers Table

The SNMP Managers table allows you to configure the attributes of up to five SNMP managers.

To configure the SNMP Managers Table, follow these steps:

1 Access the ‘Network Settings’ screen (Advanced Configuration menu > Network

Settings); the ‘Network Settings’ screen is displayed (see Figure 29 ).

2 Open the SNMP Managers Table screen by clicking the arrow sign (-->) to the right of the SNMP Managers Table label; the SNMP Managers Table screen is displayed (see

Figure 30 on page 67 ).

3 Configure the SNMP managers parameters.

4 Click the Submit button to save your changes.

5 Click the Close Window button.

6 To save the changes so they are available after a power fail see

Save Configuration on page 88 .

66 3Com VCX V7122 SIP VoIP Gateway User Manual

Figure 30

SNMP Managers Table Screen

If you clear a checkbox and click Submit, all settings in the same row revert to their defaults.

Multiple Routers Support

Multiple routers support is designed to assist the media gateway when it operates in a multiple routers network. The gateway learns the network topology by responding to ICMP redirections and caches them as routing rules (with expiration time).

When a set of routers operating within the same subnet serve as gateways to that network and intercommunicate using a dynamic routing protocol (such as OSPF, etc.), the routers can determine the shortest path to a certain destination and signal the remote host the existence of the better route. Using multiple router support the media gateway can utilize these router messages to change its next hop and establish the best path.

Note: Multiple Routers support is an integral feature that doesn’t require configuration.

Simple Network Time Protocol Support

The Simple Network Time Protocol (SNTP) client functionality generates requests and reacts to the resulting responses using the NTP version 3 protocol definitions (according to RFC

1305). Through these requests and responses, the NTP client is able to synchronize the system time to a time source within the network, thereby eliminating any potential issues should the local system clock 'drift' during operation. By synchronizing time to a network time source, traffic handling, maintenance, and debugging actions become simplified for the network administrator.

The NTP client follows a simple process in managing system time; the NTP client requests an NTP update, receives an NTP response, and updates the local system clock based on a configured NTP server within the network.

The client requests a time update from a specified NTP server at a specified update interval.

In most situations this update interval should be every 24 hours based on when the system was restarted. The NTP server identity (as an IP address) and the update interval are configurable parameters that can be specified either in the ini file (NTPServerIP,

NTPUpdateInterval respectively) or via an SNMP MIB object.

When the client receives a response to its request from the identified NTP server it must be interpreted based on time zone, or location, offset that the system is to a standard point of reference called the Universal Time Coordinate (UTC). The time offset that the NTP client

3Com VCX V7122 SIP VoIP Gateway User Manual 67

should use is a configurable parameter that can be specified either in the ini file

(NTPServerUTCOffset) or via an SNMP MIB object.

If required, the clock update is performed by the client as the final step of the update process. The update is done in such a way as to be transparent to the end users. For instance, the response of the server may indicate that the clock is running too fast on the client. The client slowly robs bits from the clock counter in order to update the clock to the correct time. If the clock is running too slow, then in an effort to catch the clock up, bits are added to the counter, causing the clock to update quicker and catch up to the correct time.

The advantage of this method is that it does not introduce any disparity in the system time, that is noticeable to an end user, or that could corrupt call timeouts and timestamps.

Configuring the Channel Settings

The Channels Settings screen enables you to set the VoIP gateway channel parameters, such as Input and Output voice gain, Jitter buffer characteristics, Modem, Fax and DTMF transport modes. These parameters are applied to all VCX V7122 channels.

Note that several Channels Settings parameters can be configured per call using profiles

(see Configuring the Profile Definitions on page 58 ).

Channel parameters are changeable on-the-fly. Changes take effect from next call.

To configure the Channel Settings parameters, follow these steps:

1 Open the ‘Channel Settings’ screen (Advanced Configuration menu > Channel

Settings); the ‘Channel Settings’ screen is displayed.

2 Configure the Channel Settings parameters.

3 Click the Submit button to save your changes.

4 To save the changes so they are available after a power fail, see

Save Configuration on page 88 .

68 3Com VCX V7122 SIP VoIP Gateway User Manual

Figure 31

Channel Settings Screen

The parameters ‘MF Transport Type’ and the 5 Answer Detector parameters are not applicable to the VCX V7122.

The parameters ‘DTMF Transport Type’ and ‘Fax Transport Mode’ are overridden by the parameters ‘IsDTMFUsed’ and ‘IsFaxUsed’ respectively.

Configuring the Trunk Settings

To configure the Trunk Settings, follow these steps:

1 Open the ‘Trunk Settings’ screen (Advanced Configuration menu > Trunk Settings); the ‘Trunk Settings’ screen is displayed.

Initially, the screen appears with the parameters fields grayed (indicating read-only). The

Stop Trunk button appears at the bottom of the screen.

The Trunk Status indicators appear colored. Table 21 on page 70 shows the possible indicators and their descriptions.

3Com VCX V7122 SIP VoIP Gateway User Manual 69

Figure 32

E1/T1 Trunk Settings Screen

2 To configure the parameters of a specific trunk, from the trunks displayed on the top, select the trunk you want to configure by clicking the Trunk’s Status indicator. The first parameter named ‘Trunk ID’ changes according to the trunk you click. The parameters displayed are for the selected trunk only.

Table 21

Trunks Status Color Indicator Keys

Indicator Color Description

Gray Disabled

Green Active-OK

70 3Com VCX V7122 SIP VoIP Gateway User Manual

Orange D-channel Alarm

(ISDN only)

3 To modify the selected trunk’s parameters, click the Stop Trunk button; the trunk is stopped, the status of the parameter ‘Trunk Configuration State’ changes to ‘Non Active’, the parameters are no longer grayed and can be modified and the Apply Trunk

Settings button appears at the bottom of the screen.

When all trunks are stopped, the Apply to all Trunks button also appears at the bottom of the screen.

If the trunk can’t be stopped because it provides the gateway’s clock (assuming the VCX V7122 is synchronized with the E1/T1 clock), assign a different E1/T1 trunk to provide the gateway’s clock or enable ‘TDM Bus PSTN Auto Clock’ on the

TDM Bus Settings screen.

To assign a different E1/T1 trunk that provides the gateway’s clock, access the

‘TDM Bus Setting’ screen and change the ‘TDM Bus Local Reference’ number to any other trunk number (this operation can be performed on-the-fly).

4 Select the ‘Protocol Type’ you use. Note that different trunks can be defined with different protocols (CAS or ISDN variants) on the same gateway (subject to the constraints in the

VCX V7122 Release Notes).

When modifying the ‘Protocol Type’ field, the menu is automatically updated according to the selected protocol (ISDN, CAS or Transparent). Additional parameters are appropriate to the selected protocol type.

5 Modify the relevant trunk configuration parameters according to your requirements.

6 To configure the different behavior bits: either enter the exact hexadecimal value of the bits in the field to the right of the relevant behavior parameter, or directly configure each bit field by completing the following steps:

Click the arrow button (-->) to the right of the relevant behavior parameter; a new window appears.

Modify each bit field according to your requirements.

7 After modifying the parameters:

To apply the changes to the selected trunk only, click the Apply Trunk Settings button.

To apply the changes to all the trunks, click the Apply to all Trunks button.

The screen is refreshed, parameters become read-only (indicated by being grayed). The

Stop Trunk button appears at the bottom of the screen.

8 To save the changes so they are available after a power fail, see

Save Configuration on page 88 .

3Com VCX V7122 SIP VoIP Gateway User Manual 71

Some parameter configuration options require a device reset; when this is the case, the Web Interface prompts the user.

9 To reset the VCX V7122, see

Save Configuration on page 88 .

Configuring the TDM Bus Settings

To configure the TDM Bus Settings parameters, follow these steps:

1 Open the ‘TDM Bus Settings’ screen (Advanced Configuration menu > TDM Bus

Settings); the ‘TDM Bus Settings’ screen is displayed.

2 Configure the TDM Bus Settings parameters.

3 Click the Submit button to save your changes.

4 To save the changes so they are available after a power fail, see Save Configuration on page 88 .

5 A device reset is required to activate the TDM Bus Settings parameters. To reset the

VCX V7122, see Save Configuration on page 88 .

Figure 33

TDM Bus Settings Screen

Usually the ‘PCM Law Select’ parameter is set to A-law for E1 trunks and to

µ

-law for T1 trunks.

See Appendix E: VCX V7122 Clock Settings on page 211 for information on configuring the

‘TDM Bus Clock Source’, ‘TDM Bus Enable Fallback’ and ‘TDM Bus PSTN Auto Clock’ parameters.

Restoring and Backing Up the Gateway Configuration

The Configuration File screen enables you to restore (load a new ini file to the gateway) or to back up (make a copy of the VoIP gateway ini file and store it in a directory on your computer) the current configuration the gateway is using.

Back up your configuration if you want to protect your VoIP gateway programming. The backup ini file includes only those parameters that were modified and contain other than default values.

72 3Com VCX V7122 SIP VoIP Gateway User Manual

Restore your configuration if the VoIP gateway has been replaced or has lost its programming information, you can restore the VoIP gateway configuration from a previous backup or from a newly created ini file. To restore the VoIP gateway configuration from a previous backup you must have a backup of the VoIP gateway information stored on your computer.

To restore or back up the ini file:

Open the ‘Configuration File’ screen (Advanced Configuration menu > Configuration

File); the ‘Configuration File’ screen is displayed.

Figure 34

Configuration File Screen

To back up the ini file, follow these steps:

1 Click the Get ini File button; the ‘File Download’ window opens.

2 Click the Save button; the ‘Save As’ window opens.

3 Navigate to the folder where you want to save the ini file.

4 Click the Save button; the VoIP gateway copies the ini file into the folder you selected.

To restore the ini file, follow these steps:

1 Click the Browse button.

2 Navigate to the folder that contains the ini file you want to load.

3 Click the file and click the Open button; the name and path of the file appear in the field beside the Browse button.

4 Click the Send ini File button, and click OK in the prompt; the gateway is automatically reset (from the cmp version stored on the flash memory).

Regional Settings

The ‘Regional Settings’ screen enables you to set and view the gateway’s internal date and time and to load to the gateway the following configuration files: Call Progress Tones, CAS and Voice Prompts. For detailed information on the configuration files, see Chapter 7:

Configuration Files on page 139 .

3Com VCX V7122 SIP VoIP Gateway User Manual 73

To configure the date and time of the VCX V7122, follow these steps:

1 Open the ‘Regional Settings’ screen (Advanced Configuration menu > Regional

Settings); the ‘Regional Settings' screen is displayed.

Figure 35

Regional Settings Screen

2 Enter the time and date where the gateway is installed.

3 Click the Set Date & Time button; the date and time are automatically updated.

Note that after performing a hardware reset, the date and time are returned to their defaults and should be updated.

To load a configuration file to the VoIP gateway, follow these steps:

1 Open the ‘Regional Settings’ screen (Advanced Configuration menu > Regional

Settings); the ‘Regional Settings’ screen is displayed (see Figure 35 ).

2 Click the Browse button adjacent to the file you want to load.

3 Navigate to the folder that contains the file you want to load.

4 Click the file and click the Open button; the name and path of the file appear in the field beside the Browse button.

5 Click the Send File button that is next to the field that contains the name of the file you want to load. An exclamation mark in the screen section indicates that the file’s loading doesn’t take effect on-the-fly (e.g., CPT file).

6 Repeat steps 2 to 5 for each file you want to load.

Saving a configuration file to flash memory may disrupt traffic on the VCX V7122.

To avoid this, disable all traffic on the device before saving to flash memory.

A device reset is required to activate a loaded CPT file.

7 To save the loaded auxiliary files so they are available after a power fail, see Save

Configuration on page 88 .

74 3Com VCX V7122 SIP VoIP Gateway User Manual

8 To reset the VCX V7122, see Save Configuration on page 88 .

Changing the VCX V7122 Username and Password

To prevent unauthorized access to the VCX V7122, it is recommended that you change the username and password (both are case-sensitive) that are used to access the Web

Interface.

To change the username and password, follow these steps:

1 Open the ‘Change Password’ screen (Advanced Configuration menu > Change

Password); the ‘Change Password’ screen is displayed.

Figure 36

Change Password Screen

2 In the ‘User Name’ and ‘Password’ fields, enter the new username and the new password respectively. Note that the username and password can be a maximum of 7 case-sensitive characters.

3 In the ‘Confirm Password’ field, reenter the new password.

4 Click the Change Password button; the new username and password are applied and the ‘Enter Network Password’ screen appears, shown in Figure 16 on page 43 .

5 Enter the updated username and password in the ‘Enter Network Password’ screen.

Status and Diagnostic

Use this subdivided menu to view and monitor the gateway’s channels, Syslog messages, hardware / software product information, and to assess the gateway’s statistics and IP connectivity information.

Gateway Statistics

Use the screens under Gateway Statistics to monitor real-time activity such as IP

Connectivity information, call details and call statistics, including the number of call attempts, failed calls, fax calls, etc.

Note: The Gateway Statistics screens doesn’t refresh automatically. To view updated information re-access the screen you require.

IP Connectivity

The IP Connectivity screen provides you with an online read-only network diagnostic connectivity information on all destination IP addresses configured in the Tel to IP Routing table.

Note: This information is available only if the parameter ‘AltRoutingTel2IPEnable’ (described in Table 28 on page 119 ) is set to 1 (Enable) or 2 (Status Only).

3Com VCX V7122 SIP VoIP Gateway User Manual 75

The information in columns ‘Quality Status’ and ‘Quality Info.’ (per IP address) is reset if two minutes elapse without a call to that destination.

To view the IP connectivity information, follow these steps:

1 Set ‘AltRoutingTel2IPEnable’ to 1 or 2.

2 Open the ‘IP Connectivity’ screen (Status & Diagnostics menu > Gateway Statistics submenu > IP Connectivity); the ‘IP Connectivity’ screen is displayed ( see Figure 37 ).

Figure 37

IP Connectivity Screen

Table 22

IP Connectivity Parameters

Column Name

IP Address

Host Name

Connectivity Method

Connectivity Status

Description

IP address defined in the destination IP address field in the Tel to IP Routing table. or

IP address that is resolved from the host name defined in the destination IP address field in the

Tel to IP Routing table.

Host name (or IP address) defined in the destination IP address field in the Tel to IP Routing table.

The method according to which the destination IP address is queried periodically (currently only by ping).

Displays the status of the IP address’ connectivity according to the method in the ‘Connectivity

Method’ field.

Can be one of the following:

OK

Lost

= Remote side responds to periodic connectivity queries.

= Remote side didn’t respond for a short period.

Fail

Init

= Remote side doesn’t respond.

= Connectivity queries not started (e.g., IP address not resolved).

Disable = The connectivity option is disabled (‘AltRoutingTel2IPMode’ equals 0 or 2).

76 3Com VCX V7122 SIP VoIP Gateway User Manual

Table 22

IP Connectivity Parameters

Column Name

Quality Status

Quality Info.

DNS Status

Description

Determines the QoS (according to packet loss and delay) of the IP address.

Can be one of the following:

Unknown = Recent quality information isn’t available.

OK

Poor

Note: This field is applicable only if the parameter ‘AltRoutingTel2IPMode’ is set to 2 or 3.

Displays QoS information: delay and packet loss, calculated according to previous calls.

Note: This field is applicable only if the parameter ‘AltRoutingTel2IPMode’ is set to 2 or 3.

Can be one of the following:

DNS Disable

DNS Resolved

DNS Unresolved

IP Tel and Tel IP Call Counters

The IP Tel and Tel IP Call Counters screens provide you with statistic information on incoming and outgoing calls. You can reset this information by clicking the Reset Counters button.

To view the IP Tel and Tel IP Call Counters information:

Open the Call Counters screen you want to view (Status & Diagnostics menu >

Gateway Statistics submenu); the relevant Call Counters screen is displayed. Figure 38 shows the ‘Tel IP Call Counters’ screen.

Figure 38

Tel IP Call Counters Screen

Monitoring the VCX V7122 Trunks and Channels

The Trunk & Channel Status screen provides real time monitoring on the current status of the VCX V7122 trunks & channels.

3Com VCX V7122 SIP VoIP Gateway User Manual 77

To monitor the status of the trunks and B-channels follow this steps:

Open the ‘Trunk & Channel Status’ screen (Status & Diagnostics menu > Channel

Status); the ‘Trunk & Channel Status’ screen is displayed.

Figure 39

VCX V7122 Trunk & Channel Status Screen

The number of trunks and channels that appear on the screen depends on the system configuration. The example above depicts a system with 8 T1 spans.

The trunk and channel status indicators appear colored. Figure 40 shows the possible indicators and their descriptions.

Figure 40

Trunk and Channel Status Color Indicator Keys

To monitor the details of a specific channel, follow these steps:

1 Click the numbered icon of the specific channel whose detailed status you need to check/monitor; the channel-specific Channel Status screen appears, shown in Figure 41 .

Click the submenu links to check/view a specific channel’s parameter settings.

78 3Com VCX V7122 SIP VoIP Gateway User Manual

Figure 41

Channel Status Details Screen

Activating the Internal Syslog Viewer

The Message Log screen displays Syslog debug messages sent by the gateway.

Note that it is not recommended to keep a ‘Message Log’ session open for a prolonged period (see the Note below). For prolonged debugging use an external Syslog server, see

Syslog Support on page 163 .

See the Debug Level parameter ‘GwDebugLevel’ (described in Table 24 on page 94 ) to determine the Syslog logging level.

To activate the Message Log, follow these steps:

1 In the General Parameters screen under Advanced Parameters submenu (accessed from the Protocol Management menu), set the parameter ‘Debug Level’ to 5. This parameter determines the Syslog logging level, in the range 0 to 5, where 5 is the highest level.

2 Open the ‘Message Log’ screen (Status & Diagnostics menu > Message Log); the

‘Message Log’ screen is displayed.

3Com VCX V7122 SIP VoIP Gateway User Manual 79

Figure 42

Message Log Screen

3 Select the messages, copy them and paste them into a text editor such as Notepad.

Send this txt file to 3Com Technical Support for diagnosis and troubleshooting.

4 To clear the screen of messages, click on the submenu Message Log; the screen is cleared and new messages begin appearing.

Do not keep the ‘Message Log’ screen minimized for a prolonged period as a prolonged session may cause the VCX V7122 to overload. As long as the screen is open (even if minimized), a session is in progress and messages are sent.

Closing the screen (and accessing another) stops the messages and terminates the session.

System Information

The System Information screen displays specific hardware and software product information.

This information can help you to expedite any troubleshooting process. Capture the screen and email it to 3Com Technical Support personnel to ensure quick diagnosis and effective corrective action. From this screen you can also view and remove any loaded auxiliary files used by the VCX V7122 (stored in the RAM).

To access the System Information screen:

Open the ‘System Information’ screen (Status & Diagnostics menu > System

Information); the ‘System Information’ screen is displayed.

80 3Com VCX V7122 SIP VoIP Gateway User Manual

Figure 43

System Information Screen

To delete any of the loaded auxiliary files, follow these steps:

1 Press the Delete button to the right of the files you want to delete. Deleting a file takes effect only after the VCX V7122 is reset.

2 Click the Reset button on the main menu bar; the Reset screen is displayed.

3 Select the Burn option and click the Reset button. The VCX V7122 is reset and the auxiliary files you chose to delete are discarded.

Software Update Menu

The ‘Software Update’ menu enables users to upgrade the VCX V7122 software by loading a new cmp file along with the ini and a suite of auxiliary files, or to update the existing auxiliary files.

The ‘Software Update’ menu comprises two submenus:

Software Update Wizard (see Software Upgrade Wizard below).

Auxiliary Files (see Auxiliary Files on page 86 ).

When upgrading the VCX V7122 software, you must load the new cmp file with all

other related configuration files.

Software Upgrade Wizard

The Software Upgrade Wizard guides users through the process of software upgrade: selecting files and loading them to the gateway. The wizard also enables users to upgrade software while maintaining the existing configuration. Using the wizard obligates users to load a cmp file. Users can choose to also use the Wizard to load the ini and auxiliary files

(e.g., Call Progress Tones) but this option cannot be pursued without loading the cmp file.

For the ini and each auxiliary file type, users can choose to reload an existing file, load a new file or not load a file at all.

3Com VCX V7122 SIP VoIP Gateway User Manual 81

The Software Upgrade Wizard requires the VCX V7122 to be reset at the end of the process, disrupting any of its traffic. To avoid disruption, disable all traffic on the VCX V7122 before initiating the Wizard.

To use the Software Upgrade Wizard, follow these steps:

1 Stop all traffic on the VCX V7122 (see the note above).

2 Open the ‘Software Upgrade Wizard’ (Software Update menu > Software Upgrade

Wizard); the ‘Start Software Upgrade’ screen appears.

Figure 44

Start Software Upgrade Screen

At this point, the process can be canceled with no consequence to the VCX V7122

(click the Cancel button). If you continue the process (by clicking the Start

Software Upgrade button, the process must be followed through and completed with a VCX V7122 reset at the end. If you click the Cancel button in any of the subsequent screens, the VCX V7122 is automatically reset with the configuration that was previously burned in flash memory.

3 Click the Start Software Upgrade button; the ‘Load a cmp file’ screen appears (see

Figure 45 ).

When in the Wizard process, the rest of the Web application is unavailable and the background Web screen is disabled. After the process is completed, access to the full Web application is restored.

82 3Com VCX V7122 SIP VoIP Gateway User Manual

Figure 45

Load a cmp File Screen

4 Click the Browse button, navigate to the cmp file and click the button Send File; the cmp file is loaded to the VCX V7122 and you’re notified as to a successful loading

(see Figure 46 ).

Figure 46

cmp File Successfully Loaded into the VCX V7122 Notification

5 Note that the four action buttons (Cancel, Reset, Back, and Next) are now activated

(following cmp file loading).

You can now choose to either: current configuration files. that were previously stored in flash memory. Note that these are NOT the files you loaded in the previous Wizard steps.

Click Figure 45 .

3Com VCX V7122 SIP VoIP Gateway User Manual 83

Click Figure 47 . Loading a new ini file or any other auxiliary file listed in the Wizard is optional.

Note that as you progress, the file type list on the left indicates which file type loading is in process by illuminating green (until ‘FINISH’).

Figure 47

Load an ini File Screen

6 In the ‘Load an ini File’ screen, you can now choose to either: default checked, becomes unchecked. Click Send File; the ini file is loaded to the VCX

V7122 and you’re notified as to a successful loading. configuration’ remains checked by default). file is loaded, the VCX V7122 uses its factory-preconfigured values.

You can now choose to either: that were previously stored in flash memory. Note that these are NOT the files you loaded in the previous Wizard steps. now as well as utilizing the other configuration files.

Click Figure 45 on page 83 .

Click Figure 48 ; Loading a new CPT file or any other auxiliary file listed in the Wizard is optional.

84 3Com VCX V7122 SIP VoIP Gateway User Manual

Figure 48

Load a CPT File Screen

7 Follow the same procedure you followed when loading the ini file (see Step

6 ). The same procedure applies to the ‘Load a VP file’ (not applicable to the VCX V7122 gateway) screen and ‘Load a coefficient file’ screen.

8 In the ‘FINISH’ screen (see

Figure 49 on page 86 ), the Next button is disabled. Complete the upgrade process by clicking Reset or Cancel.

Button Result

Reset

Cancel

The VCX V7122 ‘burns’ the newly loaded files to flash memory. The ‘Burning files to flash memory’ screen appears. Wait for the ‘burn’ to finish. When it finishes, the ‘End Process’ screen appears displaying the burned configuration files (see Figure 50 on page 86 ).

The VCX V7122 resets, utilizing the files previously stored in flash memory. (Note that these are

NOT the files you loaded in the previous Wizard steps).

3Com VCX V7122 SIP VoIP Gateway User Manual 85

Figure 49

FINISH Screen

Figure 50

‘End Process’ Screen

9 Click the End Process button; the ‘Quick Setup’ screen appears and the full Web application is reactivated.

Auxiliary Files

The ‘Auxiliary Files’ screen enables you to load to the gateway the following files: CAS, Call

Progress Tones, Voice Prompts and Prerecorded Tones (PRT). For detailed information on these files see Chapter 7: Configuration Files on page 139 . For information on deleting these files from the VCX V7122 see System Information on page 80 .

Table 23 presents a brief description of each auxiliary file.

Table 23

Auxiliary Files Descriptions

File Type

CAS

Description

Up to 8 different CAS files containing specific CAS protocol definitions.

These files are provided to support various types of CAS signaling.

Voice Prompts The voice announcement file contains a set of Voice Prompts to be played by the VCX

V7122 during operation.

Applicable only to the VXML application.

86 3Com VCX V7122 SIP VoIP Gateway User Manual

File Type

Call Progress Tones

Prerecorded Tones

Description

This is a region-specific, telephone exchange-dependent file that contains the Call

Progress Tones levels and frequencies that the VoIP gateway uses. The default CPT file is:

U.S.A.

The dat PRT file enhances the gateway’s capabilities of playing a wide range of telephone exchange tones that cannot be defined in the Call Progress Tones file.

To load an auxiliary file to the gateway, follow these steps:

1 Open the ‘Auxiliary Files’ screen (Software Upgrade menu > Load Auxiliary Files); the

‘Auxiliary Files’ screen is displayed.

Figure 51

Auxiliary Files Screen

2 Click the Browse button that is in the field for the type of file you want to load.

3 Navigate to the folder that contains the file you want to load.

4 Click the file and click the Open button; the name and path of the file appear in the field beside the Browse button.

5 Click the Send File button that is next to the field that contains the name of the file you want to load. An exclamation mark in the screen section indicates that the file’s loading doesn’t take effect on-the-fly (e.g., CPT file).

6 Repeat steps 2 to 5 for each file you want to load.

Saving an auxiliary file to flash memory may disrupt traffic on the VCX V7122. To avoid this, disable all traffic on the device before saving to flash memory.

A device reset is required to activate a loaded CPT file, and may be required for

the activation of certain ini file parameters.

7 To save the loaded auxiliary files so they are available after a power fail see

Save Configuration on page 88 .

8 To reset the VCX V7122 see

Save Configuration on page 88 .

3Com VCX V7122 SIP VoIP Gateway User Manual 87

Updating the Software Upgrade Key

The VCX V7122 devices are supplied to customers with Software Upgrade Keys already preconfigured in the devices.

Customers can later upgrade their device’s features and capabilities by specifying what upgrades they require, and purchasing a new Software Upgrade Key from 3Com to match their specification.

The Software Upgrade Key is sent to customers as a string in an ini file, to be loaded into the device. Stored in the device’s non-volatile flash memory, the string defines the features and capabilities allowed by the specific key purchased by the customer. The device allows customers to utilize only these features and capabilities. A new key overwrites a previously installed key.

For detailed information on the Software Upgrade Key, see Appendix H: Software Upgrade

Key on page 229 .

Save Configuration

The Save Configuration screen enables users to save the current parameter configuration and the loaded auxiliary files to the non-volatile memory so they are available after a power fail. Parameters that are only saved to the volatile memory revert to their previous settings after hardware reset.

Note that when performing a software reset (i.e., via Web or SNMP) you can choose to save the changes to the non-volatile memory. Therefore, there is no need to use the Save

Configuration screen.

Saving changes to the non-volatile memory may disrupt traffic on the gateway. To

avoid this, disable all traffic before saving.

To save the changes to the non-volatile, follow these steps:

1 Click the Save Configuration button on the main menu bar; the ‘Save Configuration’ screen is displayed.

Figure 52

Save Configuration Screen

2 Click the Save Configuration button in the middle of the screen; a confirmation message appears when the save is complete.

88 3Com VCX V7122 SIP VoIP Gateway User Manual

Resetting the VCX V7122

The Reset screen enables you to remotely reset the gateway. Before reset you can choose to save the gateway configuration to flash memory.

To reset the VCX V7122, follow these steps:

1 Click the Reset button on the main menu bar; the Reset screen is displayed.

Figure 53

Reset Screen

2 Select one of the following options:

Burn - (default) the current configuration is burned to flash prior to reset.

Don’t Burn - resets the device without burning the current configuration to flash (discards all modifications to the configuration).

3 Click the Reset button. If the Burn option is selected, all configuration changes are saved to flash memory. If the Don’t Burn option is selected, all configuration changes are discarded. The device is shut down and re-activated. A message about the waiting period is displayed. The screen is refreshed.

When Gatekeeper is used, the gateway issues an Unregister request before it is reset (either from the Embedded Web Server, SNMP or BootP).

3Com VCX V7122 SIP VoIP Gateway User Manual 89

90 3Com VCX V7122 SIP VoIP Gateway User Manual

C

HAPTER

6:

INI

F

ILE

C

ONFIGURATION OF

THE

VCX V7122

As an alternative to configuring the VoIP gateway using the Web Interface (see Chapter 5:

Web Management on page 41 ), it can be configured by loading the ini file containing

Customer-configured parameters.

The ini file is loaded via the BootP/TFTP utility (see Appendix B: The BootP/TFTP

Configuration Utility on page 193 ) or via any standard TFTP server. It can also be loaded through the Web Interface (see Restoring and Backing Up the Gateway Configuration on page 72 ).

The ini file configuration parameters are stored in the VCX V7122 non-volatile memory after the file is loaded. When a parameter is missing from the ini file, a default value is assigned to that parameter (according to the cmp file loaded on the VCX V7122) and stored in the nonvolatile memory (thereby overriding the value previously defined for that parameter).

Therefore, to restore the default configuration parameters, use the ini file without any valid parameters or with a semicolon (;) preceding all lines in the file.

Some of the VCX V7122 parameters are configurable through the ini file only (and not via the

Web). These parameters usually determine a low-level functionality and are seldom changed for a specific application.

Secured ini File

The ini file contains sensitive information that is required for the functioning of the VCX

V7122. It is loaded to, or retrieved from, the device via TFTP or HTTP. These protocols are unsecured and vulnerable to potential hackers. Therefore an encoded ini file significantly reduces these threats.

You can choose to load an encoded ini file to the VCX V7122. When you load an encoded ini file, the retrieved ini file is also encoded. Use the ‘TrunkPack Downloadable Conversion

Utility’ to encode or decode the ini file before you load it to, or retrieve it from, the device.

Note that the encoded ini file’s loading procedure is identical to the regular ini file’s loading procedure. For information on encoding / decoding an ini file see Encoding / Decoding an ini

File on page 223 .

Modifying an ini File

To modify the ini file, follow these steps:

1 Get the ini file from the gateway using the Embedded Web Server (see

Restoring and

Backing Up the Gateway Configuration on page 72 ).

2 Open the file (the file opens in Notepad or a Customer-defined text file editor) and modify the ini file parameters according to your requirements. Save and close the file.

3 Load the modified ini file to the gateway (using either the BootP/TFTP utility or the

Embedded Web Server).

3Com VCX V7122 SIP VoIP Gateway User Manual 91

This method preserves the programming that already exists in the device, including special default values that were preconfigured when the unit was manufactured.

Before loading the ini file to the gateway, verify that the extension of the ini file

saved on your PC is correct: Verify that the check box ‘Hide file extension for known file types’ (My computer>Tools>Folder Options>View) is unchecked. Then,

confirm that the ini file name extension is xxx.ini and NOT erroneously xxx.ini.ini or

xxx~.ini.

The ini File Content

The ini file contains the following SIP gateway information:

Basic, Logging, Web and RADIUS parameters shown in Table 24 on page 94 .

SNMP parameters shown in Table 25 on page 101 .

SIP Configuration parameters shown in Table 26 on page 103 .

Number Manipulation and Routing parameters shown in Table 28 on page 119 .

E1/T1 Configuration Parameters shown in Table 29 on page 127 .

Channel Parameters shown in Table 30 on page 133 .

Configuration Files parameters shown in Dynamic Jitter Buffer Operation on page 136 .

The ini File Structure

The ini file can contain any number of parameters. The parameters are divided into groups by their functionality. The general form of the ini file is shown in Figure 54 below.

Figure 54

ini File Structure

[Sub Section Name]

Parameter_Name = Parameter_Value

Parameter_Name = Parameter_Value

; REMARK

[Sub Section Name]

The ini File Structure Rules

Lines beginning with a semi-colon ‘;’ (as the first character) are ignored.

A Carriage Return must be the final character of each line.

The number of spaces before and after "=" is not relevant.

If there is a syntax error in the parameter name, the value is ignored.

Syntax errors in the parameter value field can cause unexpected errors (because parameters may be set to the wrong values).

Sub-section names are optional.

String parameters, representing file names, for example CallProgressTonesFileName, must be placed between two inverted commas (‘…’).

92 3Com VCX V7122 SIP VoIP Gateway User Manual

The parameter name is NOT case-sensitive; the parameter value is NOT case-sensitive

except for coder names.

The ini file should be ended with one or more carriage returns.

The ini File Example

Figure 55 shows an example of an ini file for the VoIP gateway.

Figure 55

SIP ini File Example

PCMLawSelect = 1

ProtocolType = 1

TerminationSide = 0

FramingMethod = 0

LineCode = 2

TDMBusClockSource = 4

ClockMaster = 0

;Channel Params

DJBufferMinDelay = 75

RTPRedundancyDepth = 1

IsProxyUsed = 1

ProxyIP = 192.168.122.179

CoderName = g7231,90

;List of serial B-channel numbers

TrunkGroup_1 = 0/1-24,1000

TrunkGroup_2 = 1/1-24,2000

TrunkGroup_3 = 2/1-24,3000

TrunkGroup_4 = 3/1-24,4000

EnableSyslog = 1

SyslogServerIP = 10.2.2.1

CallProgressTonesFilename = 'CPUSA.dat'

;CASFileName = ‘E_M_WinkTable.dat’

SaveConfiguration = 1

Basic, Logging, Web, and RADIUS Parameters

In Table 24 , parameters in brackets are the format in the Embedded Web

Server

*

.

3Com VCX V7122 SIP VoIP Gateway User Manual 93

Table 24

Basic, Logging, Web and RADIUS Parameters

ini File Field Name

Web Parameter Name

*

Valid Range and Description

EthernetPhyConfiguration 0 = 10 Base-T half-duplex

1 = 10 Base-T full-duplex

2 = 100 Base-TX half-duplex

3 = 100 Base-TX full-duplex

4 = auto-negotiate (Default)

Auto-negotiate falls back to half-duplex mode (HD) when the opposite port is not in auto-negotiate, but the speed (10 Base-T, 100 Base -TX) in this mode is always configured correctly.

DHCPEnable

[Enable DHCP]

0 = Disable DHCP support on the gateway (default).

1 = Enable DHCP support on the gateway.

After the gateway is powered up, it attempts to communicate with a BootP server. If a BootP server is not responding and if DHCP is enabled, then the gateway attempts to get its IP address and other network parameters from the DHCP server.

Web Note: After you enable the DHCP Server (from the Web browser) follow this procedure:

Click the Submit button.

Save the configuration using the ‘Save Configuration’ button (before you reset the gateway). For information on how to save the configuration see

Save Configuration on page 88 .

Reset the gateway directly. (Web reset doesn’t trigger the BootP/DHCP procedure and the parameter DHCPEnable reverts to ‘0’).

Note that throughout the DHCP procedure the BootP/TFTP application must be deactivated. Otherwise, the gateway receives a response from the BootP server instead of the DHCP server.

Note: The DHCPEnable is a special "Hidden" parameter. Once defined and saved in flash memory, its assigned value doesn’t revert to its default even if the parameter doesn't appear in the ini file.

EnableDiagnostics

EnableLanWatchDog

[Enable LAN Watchdog]

0 = No diagnostics (default)

1 = Perform diagnostics

0 = Disable LAN Watch-Dog (default).

1 = Enable LAN Watch-Dog.

If LAN Watch-Dog is enabled, the gateway restarts when a network failure is detected.

DNSPriServerIP

[DNS Primary Server IP]

DNSSecServerIP

[DNS Secondary

Server IP]

IP address of the primary DNS server in dotted format notation.

IP address of the secondary DNS server in dotted format notation.

94 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

Web Parameter Name

*

DNS2IP

[Internal DNS Table]

GWAppDelayTime

[Delay After Reset [sec]]

DisableNAT

StaticNATIP

[NAT IP Address]

SyslogServerIP

[Syslog Server IP

Address]

EnableSyslog

[Enable Syslog]

BaseUDPport

[RTP Base UDP Port]

IPDiffServ

[RTP IP Diff. Serv]

Valid Range and Description

Internal DNS table, used to resolve host names to IP addresses. Two different IP addresses (in dotted format notation) can be assigned to a hostname.

DNS2IP = <Hostname>, <first IP address>, <second IP address>

Note 1: If the internal DNS table is configured, the gateway first tries to resolve a domain name using this table. If the domain name isn’t found, the gateway performs a DNS resolution using an external DNS server.

Note 2: This parameter can appear up to 10 times.

Defines the amount of time (in seconds) the gateway’s operation is delayed after a reset cycle.

The default value is 0 seconds.

Note: This feature helps to overcome connection problems caused by some

LAN routers.

0 = NAT is enabled

1 = NAT is disabled (default)

If NAT is enabled, then the source IP address, of the first received RTP packet on a new session, is compared to the remote IP address, stated when session was opened. If they are not identical, then destination IP address of the outgoing RTP packets becomes the source IP address of the first incoming packet.

Static NAT IP address.

Global gateway IP address. Define if static Network Address Translation

(NAT) device is located between the gateway and the Internet.

IP address (in dotted format notation) of the computer you are using to run the Syslog Server.

The Syslog Server is an application designed to collect the logs and error messages generated by the VoIP gateway.

Note: The default UDP Syslog port is 514.

For information on the Syslog server, see Syslog Support on page 163 .

1 = Send the logs and error message generated by the gateway to the

Syslog Server. If you select 1, you must enter an IP address in the parameter SyslogServerIP.

0 = Logs and errors are not sent to the Syslog Server (default).

Note 1: Syslog messages may increase the network traffic.

Note 2: To configure the Syslog logging levels, use the parameter

‘GwDebugLevel’.

Starting UDP port for RTP channels. Should be above 6000 for VCX V7122

SIP gateways.

The default port is 6000.

0 to 63 value for setting the Diff Services Code Point (DSCP).

The default value is 0.

If defined, it overrides the IP TOS and IP Precedence settings.

Applies only to RTP packets.

IPPrecedence

[RTP IP Precedence]

0 to 7 (default 0)

Sets the value of the IP precedence field in the IP header for all RTP packets.

3Com VCX V7122 SIP VoIP Gateway User Manual 95

ini File Field Name

Web Parameter Name

*

Valid Range and Description

IPTOS

[RTP IP TOS]

0 to 15 (default 0).

Sets the value of the IP Type Of Service field in the IP header for all RTP packets.

MaxEchoCancellerLength and

EchoCancellerLength

Note: Both parameters must be set to the same value.

Maximum Echo Canceler Length in msec:

0 = Internal decision to keep max channel capacity (currently 32 msec)

4 = 32 msec

11 = 64 msec

22 = 128 msec

The default value is 0.

Note 1: When set to 64 msec or more, the number of available gateway channels is reduced (by a factor of 5/6).

For example:

Gateway with 8 E1 spans capacity is reduced to 6 spans (180 channels), while gateway with 8 T1 spans capacity remains the same (192 channels).

Note 2: The gateway must be reset after the value of

‘MaxEchoCancellerLength’ is changed.

GwDebugLevel

[Debug Level]

Defines the Syslog logging level (usually set to 5 if debug traces are needed).

0 = Debug is disabled (default)

1 = Flow debugging is enabled

2 = Flow and board interface debugging are enabled

3 = Flow, board interface and stack interface debugging are enabled

4 = Flow, board interface, stack interface and session manager debugging are enabled

5 = Flow, board interface, stack interface, session manager and board interface expanded debugging are enabled

CDRReportLevel

[CDR Report Level]

CDRSyslogServerIP

[CDR Server IP Address]

0 = CDR is not used

1 = Call Detail Record is sent to the Syslog server at the end of each call.

2 = CDR report is send to Syslog at start and at the end of each call

The CDR Syslog message complies with RFC 3161 and is identified by:

Facility = 17 (local1) and Severity = 6 (Informational)

Note: This parameter replaces the previous “EnableCDR” parameter

Defines the destination IP address for CDR logs.

The default value is a null string that causes the CDR messages to be sent with all Syslog messages.

Note: The CDR messages are sent to UDP port 514 (default Syslog port).

HeartBeatDestIP

HeartBeatDestPort

Destination IP address (in dotted format notation) to which the gateway sends proprietary UDP ‘ping’ packets.

The default IP address is 0.0.0.0.

Destination UDP port to which the heartbeat packets are sent.

The range is 0 to 64000.

The default is 0.

HeartBeatIntervalmsec Delay (in msec) between consecutive heartbeat packets.

10 = 100000.

-1 = disabled (default).

96 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

Web Parameter Name

*

Valid Range and Description

NTPServerIP

[NTP Server IP Address]

NTPServerUTCOffset

[NTP UTC Offset]

NTPUpdateInterval

[NTP Update Interval]

IP address (in dotted format notation) of the NTP server.

The default IP address is 0.0.0.0 (the internal NTP client is disabled).

For information on NTP support, see Simple Network Time Protocol Support on page 67 .

Defines the UTC (Universal Time Coordinate) offset (in seconds) from the

NTP server.

The default offset is 0. The offset range is –43200 to 43200 seconds.

Defines the time interval (in seconds) the NTP client requests for a time update.

The default interval is 86400 seconds (24 hours). The range is 0 to

214783647 seconds.

Note: It isn’t recommended to be set beyond one month (2592000 seconds).

Disconnect Supervision Parameters

DisconnectOnBroken

Connection

[Disconnect on Broken

Connection]

BrokenConnectionEvent

Timeout

[Broken Connection

Timeout]

0 = Don’t release the call.

1 = Call is released If RTP packets are not received for a predefined timeout

(default).

Note 1: If enabled, the timeout is set by the parameter

‘BrokenConnectionEventTimeout’, in 100 msec resolution. The default timeout is 10 seconds: (BrokenConnectionEventTimeout =100).

Note 2: This feature is applicable only if RTP session is used without Silence

Compression. If Silence Compression is enabled, the Gateway doesn’t detect that the RTP connection is broken.

Note 3: During a call, if the source IP address (from where the RTP packets were sent) is changed without notifying the Gateway, the Gateway will filter these RTP packets. To overcome this issue, set

‘DisconnectOnBrokenConnection=0’; the Gateway doesn’t detect RTP packets arriving from the original source IP address, and will switch (after

300 msec) to the RTP packets arriving from the new source IP address.

The amount of time (in 100 msec units) an RTP packets isn’t received, after which a call is disconnected.

The valid range is 1 to 1000. The default value is 100 (10 seconds).

Note 1: Applicable only if ‘DisconnectOnBrokenConnection = 1’.

Note 2: Currently this feature works only if Silence Suppression is disabled.

EnableSilenceDisconnect

[Disconnect Call on

Silence Detection]

1 = The gateway disconnect calls in which silence occurs in both (call) directions for more than 120 seconds.

0 = Call is not disconnected when silence is detected (default).

The silence duration can be set by the ‘FarEndDisconnectSilencePeriod’ parameter (default 120).

Note: To activate this feature set DSP Template to 2 or 3.

FarEndDisconnectSilence

Period

[Silence Detection Period]

Duration of silence period (in seconds) prior to call disconnection.

The range is 10 to 28800 (8 hours). The default is 120 seconds.

Applicable to gateways, that use DSP templates 2 or 3.

FarEndDisconnectSilence

Method

[Silence Detection

Method]

Silence detection method.

0 (None) = Silence detection option is disabled.

1 (Packets Count) = According to packet count.

3Com VCX V7122 SIP VoIP Gateway User Manual 97

ini File Field Name

Web Parameter Name

*

Valid Range and Description

FarEndDisconnectSilence

Threshold

Threshold of the packet count (in percents), below which is considered silence by the media gateway.

The valid range is 1 to 100. The default is 8%.

Note: Applicable only if silence is detected according to packet count

(FarEndDisconnectSilenceMethod = 1).

Web-Related Parameters

DisableWebTask 0 = Enable Web management (default)

1 = Disable Web management

ResetWebPassword Allows resetting to default of Web password to:

Username: “Admin”

Password: “Admin”

DisableWebConfig

LogoWidth

0 = Enable changing parameters from Web (default)

1 = Operate Web Server in “read only” mode

HTTP port used for Web management (default = 80) HTTPport

Customizing the Web Appearance Parameters

For detailed information on customizing the Web Interface, see Appendix F: Customizing the VCX V7122

Web Interface on page 213 .

UseProductName 0 = Disabled (default).

1 = Enabled.

If enabled, the ‘UserProductNane’ text string is displayed instead of the default product name.

UserProductName

UseWebLogo

Text string that replaces the product name.

The default is “VCX V7122”.

The string can be up to 29 characters.

0 = Logo image is used (default).

1 = Text string is used instead of a logo image.

If enabled, the 3Com default logo (or any other logo defined by the

‘LogoFileName’ parameter) is replaced with a text string defined by the

‘WebLogoText’ parameter.

WebLogoText Text string that replaces the logo image.

The string can be up to 15 characters.

Width (in pixels) of the logo image.

Note: The optimal setting depends on the resolution settings.

The default value is 441, which is the width of 3Com’s displayed logo.

LogoFileName

BkgImageFileName

Name of the image file containing the user’s logo.

File name can be up to 47 characters.

The logo file name can be used to replace the 3Com default Web logo with a

User defined logo.

Use a gif, jpeg or jpg image file.

Name of the file containing the user’s background image.

File name can be up to 47 characters.

The background file can be used to replace the 3Com default background image with a User defined background.

Use a gif, jpeg or jpg image file.

98 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

Web Parameter Name

*

Valid Range and Description

RADIUS-Related Parameters

EnableRADIUS

[Enable RADIUS]

AAAIndications

[AAA Indications]

0 = RADIUS application is disabled (default).

1 = RADIUS application is enabled.

0 = No indications (default).

3 = Accounting only.

MaxRADIUSSessions

[Max. RADIUS Sessions]

Number of concurrent calls that can communicate with the RADIUS server

(optional).

The valid range is 0 to 240.

The default value is 240.

SharedSecret

[Shared Secret]

RADIUSRetransmission

[RADIUS Max.

Retransmissions]

RadiusTO

“Secret” used to authenticate the gateway to the RADIUS server. It should be a cryptographically strong password.

Number of retransmission retries.

The valid range is 1 to 10.

The default value is 3.

The interval between each retry (measured in seconds).

The valid range is 1 to 30.

The default value is 10.

IP address of Authentication and Authorization server. RADIUSAuthServerIP

[RADIUS Authentication

Server IP Address]

RADIUSAuthPort

[RADIUS Authentication

Port]

Port number of Authentication and Authorization server.

The default value is 1645.

RADIUSAccServerIP

[RADIUS Accounting

Server IP Address]

IP address of accounting server.

RADIUSAccPort

[RADIUS Accounting Port]

Port number of Radius accounting server.

The default value is 1646.

RadiusAccountingType

[RADIUS Accounting

Type]

Determines when a RADIUS accounting report is issued.

0 = At the Release of the call only (default).

1 = At the Connect and Release of the call.

2 = At the Setup and Release of the call.

BootP and TFTP Parameters

IniFileURL Specifies the name of the ini file and the location of the TFTP server from which the gateway loads the ini and configuration files.

For example: tftp://192.168.0.1/filename tftp://192.10.77.13/config<MAC>

Note: The optional string “<MAC>” is replaced with the gateway’s MAC address.

Therefore, the gateway requests an ini file name that contains its MAC address. This option enables loading different configurations for specific gateways.

3Com VCX V7122 SIP VoIP Gateway User Manual 99

ini File Field Name

Web Parameter Name

*

Valid Range and Description

CmpFileURL

The BootP parameters are special "Hidden" parameters. Once defined and saved in the flash memory, they are used even if they don't appear in the ini file.

BootPRetries BootP retries. Sets the number of BootP requests the device sends during start-up. The device stops sending BootP requests when either BootP reply is received or Number of Retries is reached.

1 = 1 BootP retry, 1 second.

2 = 2 BootP retries, 3 second.

3 = 3 BootP retries, 6 second (default).

4 = 10 BootP retries, 30 second.

5 = 20 BootP retries, 60 second.

6 = 40 BootP retries, 120 second.

7 = 100 BootP retries, 300 second.

15 = BootP retries indefinitely.

Note: This parameter only takes effect from the next reset of the device.

BootPSelectiveEnable

Specifies the name of the cmp file and the location of the TFTP server from which the gateway loads a new cmp file and updates itself.

For example: tftp://192.168.0.1/filename

Note 1: When this parameter is set in the ini file, the gateway always loads the cmp file after it is reset.

Note 2: The version of the loaded cmp file isn’t checked.

Enables the Selective BootP mechanism.

1 = Enabled.

0 = Disabled (default).

The Selective BootP mechanism enables the gateway’s integral BootP client to filter unsolicited BootP/DHCP replies (accepts only BootP replies that contain the text “AUDC" in the vendor specific information field). This option is useful in environments where enterprise BootP/DHCP servers provide undesired responses to the gateway’s BootP requests.

Note: When working with DHCP (EnableDHCP=1), the selective BootP feature must be disabled.

BootPDelay

ExtBootPReqEnable

The interval between the device’s startup and the first BootP/DHCP request that is issued by the device.

1 = 1 second (default).

2 = 3 second.

3 = 6 second.

4 = 30 second.

5 = 60 second.

Note: This parameter only takes effect from the next reset of the device.

0 = Disable (default).

1 = Enable extended information to be sent in BootP request.

If enabled, the device uses the vendor specific information field in the BootP request to provide device-related initial startup information such as board type, current IP address, software version, etc. For a full list of the vendor specific Information fields, see BootP Support on page 169 .

The BootP/TFTP configuration utility displays this information in the ‘Client

Info’ column (see Figure 65 on page 195 ).

Note: This option is not available on DHCP servers.

100 3Com VCX V7122 SIP VoIP Gateway User Manual

SNMP Parameters

In Table 25 , parameters in brackets are the format in the Embedded Web

Server

*

.

Table 25

SNMP Parameter

ini File Field Name

Web Parameter Name

*

DisableSNMP

[Enable SNMP]

SNMPPort

SNMPTrustedMGR_x

Valid Range and Description

0 = SNMP is enabled (default).

1 = SNMP is disabled and no traps are sent.

The device’s local UDP port used for SNMP Get/Set commands.

The range is 100 to 3999.

The default port is 161.

Up to five IP addresses of remote trusted SNMP managers from which the

SNMP agent accepts and processes get and set requests.

Note 1: If no values are assigned to these parameters any manager can access the device.

Note 2: Trusted managers can work with all community strings.

SNMP Trap Parameters

SNMPManagerTableIP_x

[SNMP Managers Table]

SNMPManagerTrapPort_x

[SNMP Managers Table]

Up to five IP addresses of remote hosts that are used as SNMP Managers. The device sends SNMP traps to these IP addresses.

Enter the IP address in dotted format notation, for example 108.10.1.255.

Note: The first entry (out of the five) replaces the obsolete parameter

SNMPManagerIP.

Up to five parameters used to define the Port numbers of the remote SNMP

Managers. The device sends SNMP traps to these ports.

Note: The first entry (out of the five) replaces the obsolete parameter

SNMPTrapPort.

The default SNMP trap port is 162.

The SNMP trap port must be between 100 to 4000.

SNMPManagerIsUsed_x

[SNMP Managers Table]

Up to five parameters, each determines the validity of the parameters (IP address and port number) of the corresponding SNMP Manager used to receive

SNMP traps.

0 = Disabled (default)

1 = Enabled

SNMPManagerTrapSendingEnable_x

[SNMP Managers Table]

Up to five parameters, each determines the activation/deactivation of sending traps to the corresponding SNMP Manager.

0 = Sending is disabled

1 = Sending is enabled (default)

SNMPManagerIP

Note: Obsolete parameter, use

SNMPManagerTableIP_x instead.

IP address (in dotted format notation) for the computer that is used as the first

SNMP Manager. The SNMP Manager is a device that is used for receiving

SNMP Traps.

Note 1: To enable the device to send SNMP Traps, set the ini file parameter

SNMPManagerIsUsed to 1.

Note 2: If you want to use more than one SNMP manger, ignore this parameter and use the parameters ‘SNMPManagerTableIP_x’ instead.

3Com VCX V7122 SIP VoIP Gateway User Manual 101

ini File Field Name

Web Parameter Name

*

Valid Range and Description

SNMP Community String Parameters

SNMPReadOnlyCommunityString_x Read-only community string (up to 19 chars).

The default string is “public”.

SNMPReadWriteCommunityString_x Read-write community string (up to 19 chars).

The default string is “private”.

SNMPTrapCommunityString_x Community string used in traps (up to 19 chars).

The default string is “trapuser”.

SetCommunityString

Note: Obsolete parameter, use

SNMPReadWriteCommunityString_x instead.

SNMP community string (up to 19 chars).

Default community string for read “public”, for set & get “private”.

SNMPManagerIP

Note: Obsolete parameter, use

SNMPManagerTableIP_x instead.

IP address (in dotted format notation) for the computer that is used as the first

SNMP Manager. The SNMP Manager is a device that is used for receiving

SNMP Traps.

Note 1: To enable the device to send SNMP Traps, set the ini file parameter

SNMPManagerIsUsed to 1.

Note 2: If you want to use more than one SNMP manger, ignore this parameter and use the parameters ‘SNMPManagerTableIP_x’ instead.

SIP Configuration Parameters

In Table 26 , parameters in brackets are the format in the Embedded Web Server

*

.

102 3Com VCX V7122 SIP VoIP Gateway User Manual

Table 26

SIP Configuration Parameters

ini File Field Name

Web Parameter Name

*

Valid Range and Description

ControlIPDiffServ

[Signaling DiffServ]

SIPGatewayName

[Gateway Name]

IsProxyUsed

[Enable Proxy]

ProxyIP

[Proxy IP Address]

ProxyIP

ProxyIP

ProxyIP

[Redundant Proxy IP

Address]

ProxyName

[Proxy Name]

Defines the value of the 'DiffServ' field in the IP header for SIP messages.

The valid range is 0 to 63. The default value is 0.

Use this parameter to assign a name to the device (For example:

‘gateway1.com’). Ensure that the name you choose is the one that the Proxy is configured with to identify your media gateway.

Note: If specified, the gateway Name is used as the host part of the SIP URL, in the ‘From’ header. If not specified, the gateway IP address is used instead

(default).

0 = no Proxy used [internal phones table used] (default)

1 = Proxy is used

IP address of the primary Proxy server you are using.

Enter the IP address as FQDN or in dotted format notation (for example

201.10.8.1).

You can also specify the selected port in the format: <IP Address>:<port>.

This parameter is applicable only if you select ‘Yes’ in the ‘Is Proxy Used’ field.

If you enable Proxy Redundancy (by setting EnableProxyKeepAlive=1), the gateway can work with up to three Proxy servers. If there is no response from the primary Proxy, the gateway tries to communicate with the redundant

Proxies. When a redundant Proxy is found, the gateway either continues working with it until the next failure occurs or reverts to the primary Proxy (see the ‘Redundancy Mode’ parameter). If none of the Proxy servers respond, the gateway goes over the list again.

The gateway also provides real time switching (hotswap mode), between the primary and redundant proxies (‘IsProxyHotSwap=1’). If the first Proxy doesn’t respond to Invite message, the same Invite message is immediately sent to the second Proxy.

Note 1: If ‘EnableProxyKeepAlive=1’, the gateway monitors the connection with the Proxies by using keep-alive messages ("OPTIONS").

Note 2: To use Proxy Redundancy, you must specify one or more redundant

Proxies using multiple ’ProxyIP= <IP address>’ definitions.

Note 3: When port number is specified, DNS SRV queries aren’t performed, even if ‘EnableProxySRVQuery’ is set to 1.

IP addresses of the redundant Proxies you are using.

Enter the IP address as FQDN or in dotted format notation (for example

192.10.1.255). You can also specify the selected port in the format: <IP

Address>:<port>.

Note 1: This parameter is available only if you select “Yes” in the ‘Is Proxy

Used’ field.

Note 2: When port number is specified, DNS SRV queries aren’t performed, even if ‘EnableProxySRVQuery’ is set to 1.

ini file note: The IP addresses of the redundant Proxies are defined by the second, third and forth repetition of the ini file parameter ‘ProxyIP’.

Home Proxy Domain Name. If specified, the name is used as Request-URI in

REGISTER, INVITE and other SIP messages.

If the proxy name isn’t specified, the Proxy IP address is used instead.

3Com VCX V7122 SIP VoIP Gateway User Manual 103

ini File Field Name

Web Parameter Name

*

Valid Range and Description

EnableProxySRVQuery

[Enable Proxy SRV

Queries]

AlwaysSendToProxy

[Always Use Proxy]

Enables the use of DNS Service Record (SRV) queries to discover Proxy servers.

0 = Disabled (default).

1 = Enabled.

If enabled and the Proxy IP address parameter contains a domain name without port definition (e.g., ProxyIP = domain.com), an SRV query is performed. The SRV query returns up to four Proxy host names and their weights. The gateway then performs DNS A-record queries for each Proxy host name (according to the received weights) to locate up to four Proxy IP addresses. Therefore, if the first SRV query returns two domain names, and the A-record queries return 2 IP addresses each, no more searches are performed.

If the Proxy IP address parameter contains a domain name with port definition

(e.g., ProxyIP = domain.com:5080), the gateway performs a regular DNS Arecord query.

Note: This mechanism is applicable only if ‘EnableProxyKeepAlive = 1’.

0 = Use standard SIP routing rules (default).

1 = All SIP messages and Responses are sent to Proxy server.

Note: Applicable only if Proxy server is used.

SendINVITEToProxy

[Send All Invite to Proxy]

0 = INVITE messages, generated as a result of Transfer or Redirect, are sent directly to the URL (according to the refer-to header in the REFER message or contact header in 30x response) (default).

1 = All INVITE messages, including those generated as a result of Transfer or

Redirect are sent to Proxy.

Note: Applicable only if Proxy server is used and “AlwaysSendtoProxy=0”.

EnableProxyKeepAlive

[Enable Proxy Keep

Alive]

0 = Disable (default).

1 = Keep alive with Proxy, by sending "OPTIONS" SIP message every

“ProxyKeepAliveTime”.

Note: This parameter must be enabled when Proxy redundancy is used.

ProxyKeepAliveTime

[Proxy Keep Alive Time]

Defines the Proxy keep-alive time interval (in seconds) between OPTIONS messages.

The default value is 60 seconds.

UseGatewayNameFor

Options

[Use Gateway Name for

OPTIONS]

0 = Use the gateway’s IP address in keep-alive OPTIONS messages (default).

1 = Use ‘GatewayName’ in keep-alive OPTIONS messages.

The OPTIONS Request-URI host part contains either the gateway’s IP address or a string defined by the parameter ‘Gatewayname’.

The gateway uses the OPTIONS request as a keep-alive message to its primary and redundant Proxies.

IsProxyHotSwap

[Enable Proxy Hotswap]

Enable Proxy Hot Swap redundancy mode.

0 = Disabled (default)

1 = Enabled

If Hot Swap is enabled, SIP INVITE message is first sent to the primary Proxy server. If there is no response from the primary Proxy server for

“ProxyHotSwapRtx” retransmissions, the INVITE message is resent to the redundant Proxy server.

104 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

Web Parameter Name

*

Valid Range and Description

ProxyHotSwapRtx

[Number of RTX before

Hotswap]

Number of retransmitted INVITE messages before call is routed (hot swap) to another Proxy

Range: 1-30

The default is 3.

Note: This parameter is also used for alternative routing using the Tel to IP

Routing table. If a domain name in the routing table is resolved into 2 IP addresses, and if there is no response for ‘ProxyHotSwapRtx’ retransmissions to the Invite message that is sent to the first IP address, the gateway immediately initiates a call to the second IP address.

ProxyRedundancyMode

[Redundancy Mode]

IsTrustedProxy

[Is Proxy Trusted]

0 = Parking mode: gateway continues working with the last active Proxy until the next failure. (default)

1 = Homing mode: gateway always tries to work with the primary Proxy server

(switches back to the primary Proxy whenever it is available).

Note: To use ProxyRedundancyMode, enable Keep-alive with Proxy option

(EnableProxyKeepAlive=1).

This parameter isn’t applicable and must always be set to 1.

The parameter ‘AssertedIdMode’ should be used instead.

IsFallbackUsed

[

Enable Fallback to

Routing Table

]

UserName

[User Name]

0 = Gateway fallback is not used (default).

1 = Internal Telephone to IP Routing table is used when Proxy servers are not available.

When the gateway falls back to the internal Telephone to IP Routing table, the gateway continues scanning for a Proxy. When the gateway finds an active

Proxy, it switches from internal routing back to Proxy routing.

Note: To enable the redundant Proxies mechanism set

‘EnableProxyKeepAlive’ to 1.

Username used for Registration and for BASIC/DIGEST authentication process with Proxy.

Parameter doesn’t have a default value (empty string).

Password

[Password]

Cnonce

[Cnonce]

IsRegisterNeeded

[Enable Registration]

RegistrarIP

[Registrar IP Address]

RegistrarName

[Registrar Name]

GWRegistrationName

[Gateway Registration

Name]

Password used for BASIC/DIGEST authentication process with Proxy.

The default is “

Default_Passwd”.

String used by the SIP Server and client to provide mutual authentication (free format, i.e., “Cnonce = 0a4f113b”).

The default is “

Default_Cnonce

”.

0 = Gateway does not register to Proxy/Registrar (default).

1 = Gateway registers to Proxy/Registrar at power up.

IP address of Registrar server (optional). If not specified, the gateway registers to Proxy server.

Registrar Domain Name.

If specified, the name is used as Request-URI in Register messages.

If isn’t specified (default), the Registrar IP address or Proxy name or Proxy IP address is used instead.

Defines the user name that is used in From and To headers of Register messages.

If ‘GWRegistrationName’ isn’t specified (default), the ’Username’ parameter is used instead.

3Com VCX V7122 SIP VoIP Gateway User Manual 105

ini File Field Name

Web Parameter Name

*

Valid Range and Description

RegistrationTime

[Registration Time]

RegistrationTimeDivider

[Re-registration Timing

(%)]

Defines the re-registration timing (in percentage). The timing is a percentage of the re-register timing set by the Registration server.

The valid range is 50 to 100. The default value is 50.

For example: If ‘RegistrationTimeDivider = 70’ (%) and Registration Expires time = 3600, the gateway resends its registration request after 3600 x 70% =

2520 sec.

RegistrationRetryTime

[Registration Retry

Time]

Registration expired timeout (seconds). The value is used in "Expires = " header. Typically a value of 3600 is assigned, for one hour registration.

The gateway resumes registration when half the defined timeout period expires.

Defines the time period (in seconds) after which a Registration request is resent if registration fails with 4xx, or there is no response from the

Proxy/Registrar.The default is 30 seconds. The range is 10 to 3600.

PrackMode

[PRACK Mode]

PRACK mechanism mode for 1XX reliable responses:

0 = Disabled

1 = Supported (default)

2 = Required

Note 1: The Supported and Required headers contain the “100rel” parameter.

Note 2: The VCX V7122 sends PRACK message if 180/183 response is received with “100rel” in the Supported or the Required headers.

AssertedIdMode

[Asserted Identity Mode]

0 = None (default).

1 = P-asserted.

2 = P-preferred.

The Asserted ID mode defines the header that is used in the generated

INVITE request. The header also depends on the calling Privacy: allowed or restricted.

The P-asserted (or P-preferred) headers are used to present the originating party’s Caller ID. The Caller ID is composed of a Calling Number and

(optionally) a Calling Name.

P-asserted (or P-preferred) headers are used together with the Privacy header. If Caller ID is restricted the “Privacy: id” is included. Otherwise, for allowed Caller ID the “Privacy: none” is used. If Caller ID (received from PSTN) is restricted, the From header is set to <[email protected]>.

EnableRPIheader

[Enable Remote Party

ID]

SIPDestinationPort

[SIP Destination Port]

LocalSIPPort

[SIP Local Port]

IsUserPhone

[Use “user=phone” in

SIP URL]

0 = Disable (default).

1 = RPI (Remote-Party-ID) headers are generated in SIP INVITE message for both called and calling numbers.

SIP UDP destination port for sending SIP messages.

The default port is 5060.

Local UDP port used to receive SIP messages.

The default port is 5060.

0 = Doesn’t use "user=phone" string in SIP URL.

1 = "user=phone" string is part of the SIP URL (default).

IsUserPhoneInFrom

[Use “user=phone” in

From header]

0 = Doesn’t use ";user=phone" string in From header (default).

1 = ";user=phone" string is part of the From header.

106 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

Web Parameter Name

*

Valid Range and Description

CoderName

[Coders]

CoderName = Coder,ptime (can appear up to 5 times)

The following coder names can be selected: g711Alaw64k – G.711 A-law. g711Ulaw64k – G.711

µ-law. g7231 – G.723.1 6.3 kbps (default). g7231r53 g726

– G.723 5.3 kbps.

– G.726 ADPCM 32 kbps (Payload Type = 2). g729 – G.729A.

NetCoder6_4 – NetCoder 6.4 kbps.

NetCoder7_2 – NetCoder 7.2 kbps.

NetCoder8 – NetCoder 8.0 kbps.

NetCoder8_8 – NetCoder 8.8 kbps.

Transparent – Transparent coder.

Note: The coder name is case-sensitive.

The RTP packetization period (ptime, in msec) depends on the selected Coder name, and can have the following values: g711 family g729 g723 family

– 10,20,30,40,50,60,80,100,120 (default=20).

– 10,20,30,40,50,60,80,100,120 (default=20).

– 30,60,90,120,150 (default = 30).

G.726 – 10, 20, 40, 60, 80, 100, 120 (default=20).

NetCoder family – 20, 40, 60, 80, 100, 120 (default=20).

Note 1: If not specified, the ptime gets a default value.

Note 2: Each coder should appear only once.

Note 3: The ptime specifies the maximum packetization time the gateway can receive.

Note 4: G.729B is supported if the coder G.729 is selected and

‘EnableSilenceCompression’ is enabled.

For example:

CoderName = g711Alaw64k,20

CoderName = g711Ulaw64k,40

CoderName = g7231,90

TransparentPayloadType

Specifies the payload type that is used when the selected coder is set to

‘Transparent’.

The valid range is 96-120. The default value is 56.

IsFaxUsed

[Enable T.38 Fax Relay]

0 = T.38 Fax relay disabled (default)

1 = Enable T.38 Fax Relay

Note: FaxTransportMode can be set to 0 (transparent). The gateway automatically changes the Fax transport mode to T.38 if “IsFaxUsed=1” and fax is detected.

If “IsFaxUsed=0” fax can be sent (transparently) if G.711 coder is used.

T38UseRTPPort Defines that the T.38 packets are sent / received using the same port as RTP packets.

0 = Use the RTP port +2 to send / receive T.38 packets (default).

1 = Use the same port as the RTP port to send / receive T.38 packets.

3Com VCX V7122 SIP VoIP Gateway User Manual 107

ini File Field Name

Web Parameter Name

*

Valid Range and Description

CngDetectorMode

[CNG Detector Mode]

0 = Don’t detect CNG (default).

2 = Detect CNG on caller side and start fax session (if IsFaxUsed=1).

Usually T.38 fax session starts when the “preamble” signal is detected by the answering side. Some SIP gateways do not support the detection of this fax signal on the answering side. For these cases it is possible to configure the gateways to start the T.38 fax session when the CNG tone is detected by the originating side. However this mode is not recommended.

DefaultReleaseCause

[Default Release Cause]

Default Release Cause (for IP to Tel calls), used when the gateway initiates a call release, and if an explicit matching cause for this release isn’t found, a default release cause can be configured. The default release cause is described in the Q.931 notation, and translated to corresponding SIP equivalent response value.

The default release cause is: NO_ROUTE_TO_DESTINATION (3).

Other common values are: NO_CIRCUIT_AVAILABLE (34) or

DESTINATION_OUT_OF_ORDER (27), etc.

Note: The default release cause is described in the Q.931 notation, and is translated to corresponding SIP 40x or 50x value. For example: 404 for 3, 503 for 34 and 502 for 27.

For mapping of SIP to Q.931 and Q.931 to SIP release causes, see

Appendix I: Release Reason Mapping on page 233 .

IPAlertTimeout

[Tel to IP No Answer

Timeout]

Defines the time (in seconds) the gateway waits for a 200 OK response from the called party (IP side) after sending an Invite message. If the timer expires, the call is released.

The valid range is 0 to 3600. The default value is 180.

SipSessionExpires

[Session-Expires Time]

0 = Not activate (default).

Timeout [seconds] for Keeping a "re-INVITE" message alive within a SIP session.

MINSE

[Minimum Session-

Expires]

SIPMaxRtx

[SIP Maximum Rtx]

Defines the time (in seconds) that is used in the Min-SE header field. This field defines the minimum time that the user agent supports for session refresh.

The valid range is 10 to 100000. The default value is 90.

Number of UDP retransmissions of SIP messages.

The range is 1 to 7.

The default value is 7.

SipT1Rtx

[SIP T1 Retransmission

Timer (msec)]

The time interval (in msec) between the first transmission of a SIP message and the first retransmission of the same message.

The default is 500.

Note: The time interval between subsequent retransmissions of the same SIP message starts with SipT1Rtx and is multiplied by two until SipT2Rtx.

For example (assuming that SipT1Rtx = 500 and SipT2Rtx = 4000):

The first retransmission is sent after 500 msec.

The second retransmission is sent after 1000 (2*500) msec.

The third retransmission is sent after 2000 (2*1000) msec.

The fourth retransmission and subsequent retransmissions until SIPMaxRtx are sent after 4000 (2*2000) msec.

SipT2Rtx

[SIP T2 Retransmission

Timer (msec)]

The maximum interval (in msec) between retransmission of SIP messages.

The default is 4000.

Note: The time interval between subsequent retransmissions of the same SIP message starts with SipT1Rtx and is multiplied by two until SipT2Rtx.

108 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

Web Parameter Name

*

Valid Range and Description

EnableEarlyMedia

[Enable Early Media]

EnableTransfer

[Enable Transfer]

XferPrefix

[Transfer Prefix]

EnableHold

[Enable Hold]

EnableForward

[Enable Call Forward]

EnableCallWaiting

[Enable Call Waiting]

RxDTMFOption

[Declare RFC 2833 in

SDP]

0 = Early Media is disabled (default).

1 = Enable Early Media.

If enabled, the VCX V7122 gateway sends 183 Session Progress response with SDP (instead of 180 ringing), enabling the setup of the media stream prior to the answering of the call. Sending 183 response depends on the Progress

Indicator. It is sent only if PI=1 or PI=8 was received in Proceeding or Alert PRI messages. For CAS gateways see the ‘ProgressIndicator2IP’ parameter.

Note: Generally, this parameter is set to 1.

0 = Call transfer is not allowed.

1 = The gateway responds to a Refer message with "Referred To" header to initiates a Call Transfer.

Defined string that is added, as a prefix, to the transferred called number, when Refer/3xx message is received.

Note 1: The number manipulation rules apply to the user part of the Refer-

TO/Contact URL before it is sent in the Invite message.

Note 2: The xferprefix parameter can be used to apply different manipulation rules to differentiate the transferred number from the original dialed number.

0 = Hold service disabled (default).

1 = Hold service is enabled, held tone is played to holding party.

0 = Disable call forward (default).

1 = Enable call forward service.

The VCX V7122 doesn't initiate call forward, it can only respond to call forward requests.

0 = Disabled (default).

1 = Enabled.

If enabled, when the gateway initiates a Tel to IP call to a destination that is busy, it plays a Call Waiting Ringback tone to the originator of the call.

Note 1: The gateway’s Call Progress Tones file must include a Call Waiting

Ringback tone.

Note 2: The EnableHold parameter must be enabled on the called side.

Defines the supported Receive DTMF negotiation method.

0 = Don’t declare RFC 2833 Telephony-event parameter in SDP.

1 = n/a.

2 = n/a.

3 = Declare RFC 2833 “Telephony-event” parameter in SDP (default).

The gateway is designed to always be receptive to RFC 2833 DTMF relay packets. Therefore, it is always correct to include the “Telephony-event” parameter as a default in the SDP. However some gateways use the absence of the “telephony-event” from the SDP to decide to send DTMF digits inband using G.711 coder. If this is the case you can set “RxDTMFOption=0”.

3Com VCX V7122 SIP VoIP Gateway User Manual 109

ini File Field Name

Web Parameter Name

*

Valid Range and Description

TxDTMFOption

[DTMF RFC2833

Negotiation]

0 = No negotiation, DTMF digit is sent according to the “DTMFTransportType” parameter (default).

4 = Enable RFC 2833 payload type (PT) negotiation.

Note 1: This parameter is applicable only if “IsDTMFUsed=0” (out-of-band

DTMF is not used).

Note 2: If enabled, the gateway:

Negotiates RFC 2833 payload type using local and remote SDPs.

Sends DTMF packets using RFC 2833 PT according to the received SDP.

Expects to receive RFC 2833 packets with the same PT as configured by the

“RFC2833PayloadType” parameter.

Note 3: If the remote party doesn’t include the RFC 2833 DTMF relay payload type in the SDP, the gateway uses the same PT for send and for receive.

Note 4: If TxDTMFOption is set to 0, the RFC 2833 payload type is set according to the parameter ‘RFC2833PayloadType’ for both transmit and receive.

IsDTMFUsed

[Use Out-of-Band DTMF]

Use out-of-band signaling to relay DTMF digits.

0 = Disable, DTMF digits are sent according to DTMFTransportType parameter (default).

1 = Enable sending DTMF digits within INFO or NOTIFY messages.

When out-of-band DTMF transfer is used DTMFTransportType is automatically set to 0.

OutOfBandDTMFFormat

[Out-of-Band DTMF

Format]

The exact method to send out-of-band DTMF digits.

1 = INFO format (Nortel)

2 = INFO format (Cisco) - (default)

3 = NOTIFY format <draft-mahy-sipping-signaled-digits-01.txt>

Note 1: To use out-of-band DTMF, set “IsDTMFUsed=1” or “Enable DTMF = yes”.

Note 2: When using out-of-band DTMF, the “DTMFTransportType” parameter is automatically set to 0, to erase the DTMF digits from RTP path.

DisableAutoDTMFMute Enables / disables the automatic mute of DTMF digits when out-of-band DTMF transmission is used.

0 = Auto mute is used (default).

1 = No automatic mute of in-band DTMF.

When ‘DisableAutoDTMFMute=1’, the DTMF transport type is set according to the parameter ‘DTMFTransportType’ and the DTMF digits aren’t muted if outof-band DTMF mode is selected (’IsDTMFUsed =1’). This enables the sending of DTMF digits in-band (transparent of RFC 2833) in addition to out-of-band

DTMF messages.

Note: Usually this mode is not recommended.

MaxActiveCalls

[Max Number Of Active

Calls]

Defines the maximum number of calls that the gateway can have active at the same time. If the maximum number of calls is reached, new calls are not established.

The default value is max available channels (no restriction on the maximum number of calls). The valid range is 1 to 240.

MaxCallDuration

[Max Call Duration (sec)]

Defines the maximum call duration in seconds. If this time expires, both sides of the call are released (IP and Tel).

The default time is 0 seconds (no limitation).

110 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

Web Parameter Name

*

Valid Range and Description

EnableBusyOut

[Enable Busy Out]

EnableDigitDelivery2IP

[Enable Digit Delivery

to IP]

EnableDigitDelivery

[Enable Digit Delivery

to Tel]

0 = Not used (default).

1 = Enable busy out.

If Proxy is not responding (according to the Proxy keep alive mechanism) or if there is a failure in the network, and if fallback isn’t enabled

(IsFallbackUsed=0), all E1/T1 trunks are automatically put out of service by sending a remote alarm (AIS) or Service Out message for T1 PRI trunks that support these messages (NI-2, 4/5-ESS, DMS-100 and Meridian). Note that behavior varies between different protocol types.

0 = Disabled (default).

1 = Enable digit delivery to IP.

The digit delivery feature enables sending of DTMF digits to the destination IP address after the Tel IP call was answered.

To enable this feature, modify the called number to include at least one ’p’ character.

The gateway uses the digits before the ‘p’ character in the initial Invite message. After the call was answered the gateway waits for the required time

(# of ‘p’ * 1.5 seconds) and then sends the rest of the DTMF digits using the method chosen (in-band, out-of-band).

Note: The called number can include several ‘p’ characters (1.5 seconds pause).

For example, the called number can be as follows: pp699, p9p300.

The digit delivery feature enables sending of DTMF digits to the Gateway’s Bchannel after the call is answered.

0 = Disabled (default).

1 = Enable Digit Delivery feature for VCX V7122 (two stage dialing).

Note: For incoming IP Tel calls, if the called number includes the characters

‘w’ or ‘p’, the VCX V7122 Gateway places a call with the first part of the called number, and plays DTMF digits after the call is answered.

If the character ‘p’ (pause) is used, the VCX V7122 waits for 1.5 seconds before playing the next DTMF digit.

If the character ‘w’ is used, the VCX V7122 waits for detection of dial tone before it starts playing DTMF digits. The character ‘w’ can appear once in the called number, and must precede any ‘p’ character. The ‘p’ character can appear several times.

For example: if the number “1007766p100” is defined as the called number, the VCX V7122 places a call with 1007766 as the destination number, then, after the call is answered, it waits for 1.5 seconds and plays the rest of the number (100) as DTMF digits.

Other examples: 1664wpp102, 66644ppp503, 7774w100pp200.

3Com VCX V7122 SIP VoIP Gateway User Manual 111

Profile Parameters

CoderName_ID

[Coder Group Settings]

Coder list for Profiles (up to five coders in each group).

The CoderName_ID parameter (ID from 1 to 4) provides groups of coders that can be associated with IP or Tel profiles.

You can select the following coders: g711Alaw64k – G.711 A-law. g711Ulaw64k – G.711

µ-law. g7231 g7231r53

– G.723.1 6.3 kbps (default).

– G.723 5.3 kbps. g726 g729

– G.726 ADPCM 32 kbps (Payload Type = 2).

– G.729A.

NetCoder6_4 – NetCoder 6.4 kbps.

NetCoder7_2 – NetCoder 7.2 kbps.

NetCoder8 – NetCoder 8.0 kbps.

NetCoder8_8 – NetCoder 8.8 kbps.

Transparent – Transparent coder.

The RTP packetization period (ptime, in msec) depends on the selected Coder name, and can have the following values: g711 family g729 g723 family

G.726 family

– 10, 20, 30, 40, 50, 60, 80, 100, 120 (default=20).

– 10,20,30,40,50,60,80,100,120 (default=20).

– 30, 60, 90, 120, 150 (default = 30).

– 10, 20, 30, 40, 50, 60, 80, 100,120 (default=20)

NetCoder family – 20, 40, 60, 80, 100,120 (default=20).

Note 1: If not specified, the ptime gets a default value.

Note 2: Each coder should appear only once.

Note 3: The ptime specifies the maximum packetization time the Gateway can receive.

Note 4: G.729B is supported if the coder G.729 is selected and

‘EnableSilenceCompression’ is enabled.

ini file note 1: This parameter (CoderName_ID) can appear up to 20 times

(five coders in four coder groups).

ini file note 2: The coder name is case-sensitive.

ini file note 3: Enter in the format: Coder,ptime.

For example, the following three coders belong to coder group with ID=1:

CoderName_1 = g711Alaw64k,20

CoderName_1 = g711Ulaw64k,40

CoderName_1 = g7231,90

112 3Com VCX V7122 SIP VoIP Gateway User Manual

IPProfile_ID

[IP Profile Settings]

TelProfile_ID

[Tel Profile Settings]

IPProfile_<Profile ID> =

<Profile Name>,<Preference>,<Coder Group ID>,

<IsFaxUsed *>,<DJBufMinDelay *>, <DJBufOptFactor *>,

<IPDiffServ *>,<ControlIPDiffServ *>,<EnableSilenceCompression>,

<RTPRedundancyDepth>

Preference = (1-10) The preference option is used to determine the priority of the Profile. If both IP and Tel profiles apply to the same call, the coders and other common parameters of the preferred Profile are applied to that call. If the

Preference of the Tel and IP Profiles is identical, the Tel Profile parameters are applied.

For example:

IPProfile_1 = name1,2,1,0,10,13,15,44,1,1

IPProfile_2 = name2,$$,$$,$$,$,$$,$$,$$,$$,1

$$ = Not configured, the default value of the parameter is used.

(*) = Common parameter used in both IP and Tel profiles.

Note 1: The IP ProfileID can be used in the Tel2IP and IP2Tel routing tables

(Prefix and PSTNPrefix parameters).

Note 2: ‘Profile Name’ assigned to a ProfileID, enabling User’s to

identify it intuitively and easily.

Note 3: This parameter can appear up to 4 times.

TelProfile_<Profile ID> =

<Profile Name>,<Preference>,<Coder Group ID>,

<IsFaxUsed *>,<DJBufMinDelay *>, <DJBufOptFactor *>,

<IPDiffServ *>,<ControlIPDiffServ*>,<DtmfVolume>,<InputGain>,

<VoiceVolume>, <EnableDigitDelivery>, <ECE>

Preference = (1-10) The preference option is used to determine the priority of the Profile. If both IP and Tel profiles apply to the same call, the coders and other common parameters of the preferred Profile are applied to that call. If the

Preference of the Tel and IP Profiles is identical, the Tel Profile parameters are applied.

For examples:

TelProfile_1 = FaxProfile,1,2,0,10,5,22,33,2,22,34,1,1

TelProfile_2 = ModemProfile,0,10,13,$$,$$,$$,$$,$$,0,$$,0,1

$$ = Not configured, the default value of the parameter is used.

(*) = Common parameter used in both IP and Tel profiles.

Note 1: The Tel ProfileID can be used in the Trunk Group table (TrunkGroup_x parameter).

Note 2: ‘Profile Name’ assigned to a ProfileID, enabling User’s to

identify it intuitively and easily.

Note 3: This parameter can appear up to 4 times.

3Com VCX V7122 SIP VoIP Gateway User Manual 113

ISDN and CAS Interworking-Related Parameters

In Table 27 , parameters in brackets are the format in the Embedded Web Server

*

.

Table 27

ISDN and CAS Interworking-Related Parameters

ini File Field Name

Web Parameter Name

*

Valid Range and Description

PlayRBTone2Tel

[Play Ringback Tone

to Tel]

0 (Don’t play) = The ISDN / CAS gateway doesn’t play a Ringback Tone

(RBT). No PI is sent to the ISDN, unless the parameter ‘Progress Indicator to

ISDN’ is configured differently.

1 (Play) = The CAS gateway plays a local RBT to PSTN after receipt of a 180 ringing response (with or without SDP).

Note: Reception of a 183 response doesn’t cause the CAS gateway to play an

RBT (unless ‘SIP183Behavior = 1’).

The ISDN gateway functions according to the parameter

‘LocalISDNRBToneSource’:

If the ISDN gateway receives a 180 ringing response (with or without SDP) and ‘LocalISDNRBToneSource = 1’, it plays a RBT and sends an Alert with PI

= 8 (unless the parameter ‘Progress Indicator to ISDN’ is configured differently).

If ‘LocalISDNRBToneSource = 0’, the ISDN gateway doesn’t play an RBT and an Alert message (without PI) is sent to the ISDN. In this case, the PBX /

PSTN should play the RBT to the originating terminal by itself.

Note: Reception of a 183 response doesn’t cause the ISDN gateway to play an RBT; the gateway issues a Progress message (unless

‘SIP183Behavior = 1’).

If ‘SIP183Behavior = 1’, the 183 response is treated the same way as a 180 ringing response.

2 = Play according to “early media” (default).

If a 180 response is received and the voice channel is already open (due to a previous 183 early media response or due to an SDP in the current 180 response), the ISDN / CAS gateway doesn’t play the RBT; PI = 8 is sent in an

ISDN Alert message (unless the parameter ‘Progress Indicator to ISDN’ is configured differently).

If a 180 response is received but the “early media” voice channel is not opened, the CAS gateway plays an RBT to the PSTN; the ISDN gateway functions according to the parameter ‘LocalISDNRBToneSource’:

If ‘LocalISDNRBToneSource = 1’, the ISDN gateway plays an RBT and sends an ISDN Alert with PI = 8 to the ISDN (unless the parameter ‘Progress

Indicator to ISDN’ is configured differently).

If ‘LocalISDNRBToneSource = 0’, the ISDN gateway doesn’t play an RBT.

No PI is sent in the ISDN Alert message (unless the parameter ‘Progress

Indicator to ISDN’ is configured differently). In this case, the PBX / PSTN should play an RBT tone to the originating terminal by itself.

Note: Reception of a 183 response results in an ISDN Progress message

(unless ‘SIP183Behavior = 1’).

If ‘SIP183Behavior = 1’ (183 is handled in the same way as a 180+SDP), the gateway sends an Alert message with PI = 8, without playing an RBT.

114 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

Web Parameter Name

*

Valid Range and Description

PlayRBTone2IP

[Play Ringback Tone to

IP]

0 = Ringback tone isn’t played (default).

1 = Ringback tone is played (to IP) after SIP 183 session progress response is sent.

If configured to ‘1’ (Play),

For IP Tel calls, if a Progress or an Alert message with PI is sent from the

ISDN and ‘EnableEarlyMedia = 1’, the VCX V7122 opens a voice channel and sends 183 response. It doesn’t play a Ringback tone to IP (assuming that the

Ringback tone is played by the ISDN).

Otherwise, if a voice channel isn’t opened, the VCX V7122 plays Ringback tone to IP, after receiving an Alert message from the ISDN. It sends 183 response, signaling the originating party to open a voice channel in order to hear the played Ringback tone.

Note 1: To enable the gateway to send a 183 response, set

‘EnableEarlyMedia’ to 1.

Note 2: If ‘EnableDigitDelivery = 1’, the gateway doesn’t play a Ringback tone to IP and doesn’t send a 183 response.

ProgressIndicator2ISDN 0, 1 or 8

-1 = Not configured (default).

If set to "0" PI is not sent to ISDN.

If set to "1" or "8" the PI value is sent to PSTN in Q.931/Proceeding and

Alerting messages.

If not configured, the PI in ISDN messages is set according to the "Play

Ringback to Tel" parameter.

Usually if PI = 1 or 8, the PSTN/PBX cuts through the audio channel without playing local Ringback tone, enabling the originating party to hear remote Call

Progress Tones or network announcements.

ProgressIndicator2IP -1 = (Not configured) for ISDN spans, the PI that is received in ISDN

Proceeding, Progress and Alert messages is used as described in the following options (default).

0 = (No PI) For IP Tel call, the gateway sends “180 Ringing” SIP response to

IP after receiving ISDN Alert, or (for CAS) after placing a call to PBX/PSTN.

8, 1 = For IP Tel call, if ‘EnableEarlyMedia=1’, the gateway sends “183 session in progress” message + SDP, after a call is placed to PBX/PSTN over the trunk. This is used to cut through voice path, before remote party answers the call, enabling the originating party to listen to network Call Progress Tones

(such as Ringback tone or other network announcements).

PIForDisconnectMsg

[Set PI in Rx Disconnect

Message]

Defines the gateway’s behavior when a Disconnect message is received from the ISDN before a Connect message was received.

“Not configured” = Sends a 183 message according to the received PI in the

ISDN Disconnect message. If PI = 1 or 8, the gateway sends a 183 response, enabling the PSTN to play a voice announcement to the IP side. If there isn’t a

PI in the Disconnect message, the call is released (default).

0 = Do not send a 183 message to IP. The call is released.

1, 8 = Sends 183 message to IP.

ConnectOnProgressInd 0 = Connect message isn’t sent after 183 Session Progress is received

(default).

1 = Connect message is sent after 183 Session Progress is received.

This feature enables the play of announcements from IP to PSTN without the need to answer the Tel IP call. It can be used with PSTN networks that don’t support the opening of a TDM channel before an ISDN Connect message is received.

3Com VCX V7122 SIP VoIP Gateway User Manual 115

ini File Field Name

Web Parameter Name

*

Valid Range and Description

SIP183Behavior

[183 Message Behavior]

Defines the ISDN message that is sent when 183 Session Progress message is received for IP Tel calls.

0 = Progress message (default).

1 = Alert message.

When set to 1, the gateway sends an Alert message (after the receipt of a 183 response) instead of an ISDN Progress message.

LocalISDNRBToneSource

[Local ISDN Ringback

Tone Source]

Determines whether Ringback tone is played to the ISDN by the PBX / PSTN or by the gateway.

0 = PBX / PSTN (default).

1 = Gateway.

This parameter is applicable to ISDN protocols. It is used simultaneously with the parameter ’PlayRBTone2Tel’.

PSTNAlertTimeout

[PSTN Alert Timeout]

Alert Timeout in seconds (ISDN T2 timer) for outgoing calls to PSTN.

The default is 180 seconds. The range is 0 to 240.

Note: The PSTN stack T2 timer can be overridden by a lower value, but it can’t be increased.

ISDNTransferCapability

[ISDN Transfer

Capabilities]

ScreeningInd2IP

[Send Screening

Indicator to IP]

Defines the IP ISDN Transfer Capability of the Bearer Capability IE in ISDN

Setup messages.

0 = Audio 3.1 (default).

1 = Speech.

2 = Data.

Note: If this parameter isn’t configured or equals to ‘–1’, Audio 3.1 capability is used.

The parameter can overwrite the calling number screening indication for ISDN

Tel IP calls.

-1 = not configured (interworking from ISDN to IP) or set to 0 for CAS.

0 = user provided, not screened.

1 = user provided, verified and passed.

2 = user provided, verified and failed.

3 = network provided.

Note: Applicable only if Remote Party ID (RPID) header is enabled.

SupportRedirectInFacility 0 = Not Supported (default).

1 = Supports partial retrieval of Redirect Number (number only) from a Facility

IE in ISDN Setup messages. Applicable to Redirect number according to

ECMA-173 Call Diversion Supplementary Services.

Note: To enable this feature, ‘ISDNDuplicateQ931BuffMode’ must be set to 1.

EnableCIC 0 = Do not relay the Carrier Identification Code (CIC) to ISDN (default).

1 = CIC is relayed to ISDN in Transit Network Selection IE.

If enabled, the CIC code (received in an Invite Request-URI) is included in a

TNS IE in ISDN Setup message. For example: INVITE sip:555666;[email protected] sip/2.0.

Note: Currently this feature is supported only in SIP ISDN direction.

116 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

Web Parameter Name

*

Valid Range and Description

EnableAOC

TimeForReorderTone

DisconnectOnBusyTone

[Disconnect Call on

Detection of Busy Tone]

0 = Do not disconnect call on detection of busy tone.

1 = Disconnect call on detection of busy tone (default).

This parameter is applicable to CAS protocols.

PlayBusyTone2ISDN

Busy or Reorder Tone duration the CAS gateway plays before releasing the line.

The valid range is 0 to 15. The default value is 10 seconds.

Applicable also to ISDN if ‘PlayBusyTone2ISDN = 2’. Selection of Busy or

Reorder tone is done according to release cause received from IP.

This parameter enables the VCX V7122 ISDN gateway to play a Busy or a

Reorder tone to the PSTN after a call is released.

0 = Immediately sends an ISDN Disconnect message (default).

1 = Sends an ISDN Disconnect message with PI=8 and plays a Busy or a

Reorder tone to the PSTN (depending on the release cause).

2 = Delays the sending of an ISDN Disconnect message for

‘TimeForReorderTone’ seconds and plays a Busy or a Reorder tone to the

PSTN. Applicable only if the call is released from the IP before it reaches the

Connect state. Otherwise, the Disconnect message is sent immediately and no tones are played.

TrunkTransferMode_X

0 = Not used (default).

1 = ISDN Advice of Charge (AOC) messages are interworked to SIP.

The gateway supports reception of ISDN (Euro ISDN) AOC messages. AOC messages can be received during a call (Facility messages) or at the end of a call (Disconnect or Release messages). The gateway converts the AOC messages into SIP Info (during a call) and Bye (end of a call) messages using a proprietary AOC SIP header. The gateway supports both Currency and

Pulse AOC messages.

0 = Not supported (default).

1 = Supports CAS NFA DMS-100 transfer.

When a SIP Refer message is received, the gateway performs a Blind

Transfer by executing a CAS Wink and dialing the Refer-to number to the

Switch and then releasing the call.

Note: A specific NFA CAS table is required.

EnableTDMoverIP

[Enable TDM Tunneling]

0

= Disabled (default).

1 = TDM Tunneling is enabled.

When TDM Tunneling is enabled, the originating VCX V7122 automatically initiates SIP calls from all enabled B-channels belonging to the E1/T1/J1 spans that are configured with the ‘Transparent’ protocol. The called number of each call is the internal phone number of the B-channel that the call originates from.

The IP to Trunk Group routing table is used to define the destination IP address of the terminating VCX V7122. The terminating VCX V7122 gateway automatically answers these calls if its E1/T1 protocol is set to ‘Transparent’

(ProtocolType = 5).

CASTransportType

[CAS Transport Type]

0 = Disable CAS relay (default).

1 = Enable CAS relay mode using RFC 2833.

The CAS relay mode can be used with the TDM tunneling feature to enable tunneling over IP for both voice and CAS signaling bearers.

3Com VCX V7122 SIP VoIP Gateway User Manual 117

ini File Field Name

Web Parameter Name

*

XChannelHeader

Valid Range and Description

AddIEinSetup

[Add IE in SETUP]

SendIEonTG

[Trunk Groups to

Send IE]

ISDNDMSTimerT310

ISDNJapanNTTTimer

T3JA

0 = x-channel header is not used (default).

1 = x-channel header is generated, with trunk/B-channel information.

The header provides information on the E1/T1 physical trunk/B-channel on which the call is received or placed. For example “x-channel: DS/DS1-5/22”.

This header is generated by the VCX V7122 and is sent in the following messages: INVITE and 183/180/200OK responses.

This parameter enables to add an optional Information Element data (in hex format) to ISDN SETUP message.

For example: to add the following IE: “0x20,0x02,0x00,0xe1”, define:

“AddIEinSetup = 200200e1”.

Note: This IE is sent from the Trunk Group IDs defined by the parameter

‘SendIEonTG’.

A list of Trunk Group IDs (up to 50 characters) from where the optional ISDN

IE, defined by the parameter ‘AddIEinSetup’, is sent.

For example: "SendIEonTG = 1,2,4,10,12,6”.

Overrides the T310 timer for the DMS-100 ISDN variant.

This parameter enables users to increase the 10 seconds timeout from call

Setup until Alert is received up to 30 seconds.

The valid range is 10 to 30. The default value is 10 (seconds).

Note: Applicable only to Nortel DMS and Nortel MERIDIAN PRI variants

(ProtocolType = 14 and 35).

T3_JA timer (in seconds).

This parameter overrides the internal PSTN T3 timeout on the Users Side (TE side).

If an outgoing call from the VCX V7122 to an ISDN subscriber is not answered during this timeout, the call is released.

The valid range is 10 to 180. The default value is 50.

Applicable only to Japan NTT PRI variant (ProtocolType = 16).

118 3Com VCX V7122 SIP VoIP Gateway User Manual

Number Manipulation and Routing Parameters

In Table 28 , parameters in brackets are the format in the Embedded Web Server

*

.

Table 28

Number Manipulation and Routing Parameters

ini File Field Name

Web Parameter Name

*

TrunkGroup_x

[Trunk Group Table]

ChannelList

Note: It is recommended to use TrunkGroup_x parameter instead.

Valid Range and Description

TrunkGroup_x = T/a-b,c,d x = Trunk group ID (1 to 99).

T = Physical trunk number (0 to 7). a = Starting B-channel (from 1). b = Ending B-channel (up to 31). c = Phone number associated with the first channel (optional). d = Optional Tel Profile ID (1 to 5).

For example:

TrunkGroup_1 = 0/1-31,1000 (for E1 span).

TrunkGroup_1 = 1/1-31,$$,1.

TrunkGroup_2 = 2/1-24,3000 (for T1 span).

Trunk group is the recommended method to configure the gateway's B-channels. The parameter ’ChannelList’ (although still supported) mustn’t be used simultaneously with Trunk Groups.

Note 1: An optional Tel Profile ID (1 to 5) can be applied to each group of Bchannels.

Note 2: Parameters can be skipped by using the sign "$$".

List of phone numbers, used to define the enabled B-channels for gateway operation,

‘a, b, c,d’ a = first channel. b = number of channels starting from ‘a’. c = the phone number of the first channel. d = Tel Profile ID assigned to the group of channels. example: ChannelList = ‘0,30,1001’

Defines phone numbers 1001 to 1030 for 30 gateway channels.

The ini file can include up to ten ‘ChannelList = ‘ entries.

Usually single ChannelList parameter is enough to define the complete 8 trunk gateway:

ChannelList = ‘0,240,1000’; For eight E1 spans.

ChannelList = ‘0,192,1000’; For eight T1 CAS spans.

Phone numbers can be defined individually per E1 or T1 span:

For E1 spans (CAS or ISDN): 0 to 29 for first span, 30 to 59 for second span, 60 to 89 for 3 rd

span, 90 to 119 for 4 th

span.

For T1 ISDN spans: 0 to 22 for first span, 23 to 45 for second span, 46 to 68 for 3 rd span, and 69 to 91 for 4 th

span.

For T1 CAS signaling: 0 to 23 for first span, 24 to 47 for second span, 48 to 71 for 3 rd span, and 72 to 95 for 4 th

span.

It is suggested to use Trunk Groups in VCX V7122 gateway to define enabled Bchannels, instead of ChannelList parameter.

3Com VCX V7122 SIP VoIP Gateway User Manual 119

ini File Field Name

Web Parameter Name

*

Valid Range and Description

DefaultNumber

[Default Destination Number]

Defines the telephone number that the gateway uses if the parameters

‘TrunkGroup_x’ or ’ChannelList‘ don’t include a phone number. The parameter is used as a starting number for the list of B-channels comprising all trunk groups in the gateway.

ChannelSelectMode

[Channel Select Mode]

Defines common rule of port allocation for IP to TEL calls.

0 = By phone number - Select the gateway port according to the called number (see the note below).

1 = Cyclic Ascending - Select the next available channel in an ascending cycle order.

Always select the next higher channel number in the Trunk Group. When the gateway reaches the highest channel number in the Trunk Group, it selects the lowest channel number in the Trunk Group and then starts ascending again (default).

2 = Ascending - Select the lowest available channel. Always start at the lowest channel number in the Trunk Group and if that channel is not available, select the next higher channel.

3 = Cyclic Descending - Select the next available channel in descending cycle order.

Always select the next lower channel number in the Trunk Group. When the gateway reaches the lowest channel number in the Trunk Group, it selects the highest channel number in the Trunk Group and then start descending again.

4 = Descending - Select the highest available channel. Always start at the highest channel number in the Trunk Group and if that channel is not available, select the next lower channel.

5 = Number + Cyclic Ascending – First select the gateway port according to the called number (see the note below). If the called number isn’t found, then select the next available channel in ascending cyclic order. Note that if the called number is found, but the port associated with this number is busy, the call is released.

Note: The internal numbers of the gateway’s B-channels are defined by the

‘TrunkGroup_x’ parameter (under ‘Phone Number’).

TrunkGroupSettings

[Trunk Group Settings]

AddTrunkGroupAsPrefix

[Add Trunk Group ID as

Prefix]

Defines rules for port allocation for specific Trunk Groups, if such rule doesn’t exist, the global rule defined by ChannelSelectMode applies. a, b a = Trunk Group ID number. b = Channel select mode for that Trunk Group.

Available values are identical to those defined by the ChannelSelectMode parameter.

0 = not used.

1 = For Tel IP incoming call, Trunk Group ID is added as prefix to destination phone number. Applicable only if trunk group ID are configured.

Can be used to define various routing rules.

AddPortAsPrefix

[Add Trunk ID as Prefix]

ReplaceEmptyDstWithPort

Number

[Replace Empty Destination

with Port Number]

0 = Don’t add (default).

1 = Add trunk ID number (single digit in the range 1 to 8) as a prefix to the called phone number for Tel IP incoming calls.

This option can be used to define various routing rules.

0 = Disabled (default).

1 = Enabled, Internal channel number is used as a destination number if called number is missing.

Note: Applicable only to Tel IP calls, if called number is missing.

120 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

Web Parameter Name

*

Valid Range and Description

UseSourceNumberAsDisplay

Name

[Use Source Number as

Display Name]

AlwaysUseRouteTable

[Use Routing Table for Host

Names and Profiles]

0 = Interworks the Tel calling name to SIP Display Name (default).

1 = Set Display Name to Calling Number if not configured.

Applicable to Tel IP calls. If enabled and if the incoming Tel to IP call doesn’t contain the calling party name, the calling number is used instead.

All CAS protocols don’t provide the calling party name. Therefore, in CAS, if this parameter is enabled, the Display Name is identical to the calling number.

Use the internal Tel to IP routing table to obtain the URL Host name and (optionally) an IP profile (per call), even if Proxy server is used.

0 = Don’t use (default).

1 = Use.

Note: This Domain name is used, instead of Proxy name or Proxy IP address, in the

INVITE SIP URL.

Prefix

[Tel to IP Routing Table]

PSTNPrefix

[IP to Trunk Group Routing

Table]

Mapping phone number to IP address, using phone number prefix.

Selection of IP address (for Tel To IP calls) is according to destination and source prefixes.

Prefix = <Destination Phone Prefix>, <IP Address>,<Src Phone Prefix>,

<IP Profile ID>

For example:

Prefix = 20,10.2.10.2,202,1

Prefix = 10[340-451]xxx#,10.2.10.6,*,1

Prefix = *,gateway.domain.com,*

Note 1: An optional IP ProfileID (1 to 5) can be applied to each routing rule.

Note 2: <destination / source phone prefix> can be single number or a range of numbers.

Note 3: This parameter can appear up to 50 times.

Note 4: Parameters can be skipped by using the sign "$$", for example:

Prefix = $$,10.2.10.2,202,1

For available notations, see Dialing Plan Notation on page 50 .

For detailed information on this feature, see Tel to IP Routing Table on page 52 .

PSTNPrefix = a,b,c,d,e a = Destination Number Prefix b = Trunk group ID (1 to 99) c = Source Number Prefix d = Source IP address e = IP Profile ID (1 to 5)

Selection of trunk groups (for IP to Tel calls) is according to destination number, source number and source IP address.

Note 1: To support the ‘in call alternative routing’ feature, users can use two entries that support the same call, but assign them with a different trunk groups. The second entry functions as an alternative selection if the first rule fails as a result of one of the release reasons listed in the AltRouteCauseIP2Tel table.

Note 2: An optional IP ProfileID (1 to 5) can be applied to each routing rule.

Note 3: The Source IP Address can include the “x” wildcard to represent single digits.

For example: 10.8.8.xx represents all IP addresses between 10.8.8.10 to 10.8.8.99.

3Com VCX V7122 SIP VoIP Gateway User Manual 121

ini File Field Name

Web Parameter Name

*

Valid Range and Description

RemovePrefix

[IP to Tel Remove Routing

Table Prefix]

RouteModeIP2Tel

[IP to Tel routing Mode]

0 = Don't remove prefix (default).

1 = Remove PSTN prefix (defined in the routing table) from a telephone number of an incoming IP call, before forwarding it to PSTN.

Applicable only if number manipulation is performed after call routing for IP Tel calls

(RouteModeIP2Tel = 0).

0 = Route calls before number manipulation (default).

1 = Route calls after number manipulation.

Defines order between routing calls to Trunk group and manipulation of destination number.

RouteModeTel2IP

[Tel to IP routing Mode]

SwapRedirectNumber

[Swap Redirect and Called

Numbers]

0 = Don't change numbers (default).

1 = Incoming ISDN call that includes redirect number (sometimes referred as "original called number") uses this number instead of the called number.

AddTON2RPI

[Add Number Plan and Type

to Remote Party ID Header]

0 = TON/PLAN parameters aren’t included in the RPID header.

1 = TON/PLAN parameters are included in the RPID header (default).

If RPID header is enabled (EnableRPIHeader = 1) and ‘AddTON2RPI=1’, it is possible to configure the calling and called number type and number plan using the

Number Manipulation tables for Tel IP calls.

NumberMapTel2IP

[Destination Phone Number

Manipulation Table for

Tel

IP calls]

0 = Route calls before number manipulation (default).

1 = Route calls after number manipulation.

Defines order between routing incoming calls to IP, using routing table, and manipulation of destination number.

Not applicable if Outbound Proxy is used.

Manipulates the destination number for Tel to IP calls.

NumberMapTel2IP = a,b,c,d,e,f,g a b

= Destination number prefix

= Number of stripped digits from the left, or (if brackets are used) from the right. A combination of both options is allowed. c = String to add as prefix, or (if brackets are used) as suffix. A combination of both options is allowed. d = Number of remaining digits from the right e f

= Number Plan used in RPID header

= Number Type used in RPID header g = Source number prefix

The ‘b’ to ‘f’ manipulations rules are applied if the called and calling numbers match the ‘a’ and ‘g’ conditions.

The manipulation rules are executed in the following order: ‘b’, ‘d’ and ‘c’.

Parameters can be skipped by using the sign "$$", for example:

NumberMapTel2IP=01,2,972,$$,0,0,$$

NumberMapTel2IP=03,(2),667,$$,0,0,22

122 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

Web Parameter Name

*

NumberMapIP2Tel

[Destination Phone Number

Manipulation Table for

IP

Tel calls]

SourceNumberMapTel2IP

[Source Phone Number

Manipulation Table for

Tel

IP calls]

Valid Range and Description

Manipulate the destination number for IP to Tel calls.

NumberMapIP2Tel = a,b,c,d,e,f,g,h,i a b

= Destination number prefix

= Number of stripped digits from the left, or (if brackets are used) from the right. A combination of both options is allowed. c = String to add as prefix, or (if brackets are used) as suffix. A combination of i g h both options is allowed. d = Number of remaining digits from the right e f

= Q.931 Number Plan

= Q.931 Number Type

= Source number prefix

= Not applicable, set to $$

= Source IP address

The ‘b’ to ‘f’ manipulation rules are applied if the called and calling numbers match the ‘a’, ‘g’ and ‘i’ conditions.

The manipulation rules are executed in the following order: ‘b’, ‘d’ and ‘c’.

Parameters can be skipped by using the sign "$$", for example:

NumberMapIP2Tel =01,2,972,$$,0,$$,034

NumberMapIP2Tel =03,(2),667,$$,$$,0,22,$$,10.13.77.8

Note: The Source IP address can include the “x” wildcard to represent single digits.

For example: 10.8.8.xx represents all the addresses between 10.8.8.10 to 10.8.8.99.

SourceNumberMapTel2IP = a,b,c,d,e,f,g,h a b

= Source number prefix

= Number of stripped digits from the left, or (if in brackets are used) from right. A Combination of both options is allowed. c = String to add as prefix, or (if in brackets are used) as suffix. A Combination of both options is allowed. d = Number of remaining digits from the right e f g h

= Number Plan used in RPID header

= Number Type used in RPID header

=Destination number prefix

=Calling number presentation (0 to allow presentation, 1 to restrict presentation)

The ‘b’ to ‘f’ and ‘h’ manipulation rules are applied if the called and calling numbers match the ‘a’ and ‘g’ conditions.

The manipulation rules are executed in the following order: ‘b’, ‘d’ and ‘c’.

Parameters can be skipped by using the sign "$$", for example:

SourceNumberMapTel2IP=01,2,972,$$,0,0,$$,1

SourceNumberMapTel2IP=03,(2),667,$$,0,0,22,0

3Com VCX V7122 SIP VoIP Gateway User Manual 123

ini File Field Name

Web Parameter Name

*

Valid Range and Description

SourceNumberMapIP2Tel

[Source Phone Number

Manipulation Table for

IP

Tel calls]

Manipulate the source number for IP to Tel calls.

SourceNumberMapIP2Tel = a,b,c,d,e,f,g,h a b

= Source number prefix

= Number of stripped digits from the left, or (if brackets are used) from the right. A combination of both options is allowed. c = String to add as prefix, or (if brackets are used) as suffix. A combination of both options is allowed. d = Number of remaining digits from the right e f

= Q.931 Number Plan

= Q.931 Number Type g h

= Destination number prefix

=Calling number presentation (0 to allow presentation, 1 to restrict presentation)

The ‘b’ to ‘f’ and ‘h’ manipulation rules are applied if the called and calling numbers match the ‘a’ and ‘g’ conditions.

The manipulation rules are executed in the following order: ‘b’, ‘d’ and ‘c’.

Parameters can be skipped by using the sign "$$", for example:

SourceNumberMapIP2Tel =01,2,972,$$,0,$$,034,1

SourceNumberMapIP2Tel =03,(2),667,$$,$$,0,22

The following Number Plan and Type values are supported in the Destination and Source Manipulation tables:

0,0 = Unknown, Unknown

9,0 = Private, Unknown

9,1 = Private, Level 2 Regional

9,2 = Private, Level 1 Regional

9,3 = Private, PISN Specific

9,4 = Private, Level 0 Regional (local)

1,0 = Public(ISDN/E.164), Unknown

1,1 = Public(ISDN/E.164), International

1,2 = Public(ISDN/E.164), National

1,3 = Public(ISDN/E.164), Network Specific

1,4 = Public(ISDN/E.164), Subscriber

1,6 = Public(ISDN/E.164), Abbreviated

124 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

Web Parameter Name

*

DestNumberType

DestNumberPlan

SourceNumberType

SourceNumberPlan

SecureCallsFromIP

[IP Security]

Valid Range and Description

0 = Unknown

1 = International Number

2 = National Number (default)

3 = Network Specific Number

4 = Subscribe Number (or local)

6 = Abbreviated number

7 = Reserved for extension

Used for IP PSTN calls.

The Number Type (TON) parameter is used in ISDN/Q.931 Setup messages.

Not all combinations of TON/NPI (Number Plan Indication) are allowed. Usually, you need to select one of the following TON/NPI sets:

0/0 (Unknown/Unknown)

1/1 (ISDN/International)

1/2 (ISDN/National)

1/4 (ISDN/Subscriber)

Note: Numbering Plan/Type in the Number Manipulation table, if present, overrides the value defined by the global parameter. Using the Number Manipulation tables enables configuration of the Numbering Plan and Type on a per call basis, according to destination (or source) number.

0 = Unknown

1 = ISDN/Telephony Numbering Plan (default)

3 = Data Numbering Plan

4 = Telex Numbering Plan

8 = National Standard Numbering Plan

9 = Private Numbering Plan

15 = Reserved for extension

Used for IP PSTN calls.

The Number Type (NPI) parameter is used in ISDN/Q.931 Setup messages.

Not all the Combinations of TON/NPI are allowed. Usually, you need to select one of the following TON/NPI sets:

0/0 (Unknown/Unknown)

1/1 (ISDN/International)

1/2 (ISDN/National)

1/4 (ISDN/Subscriber)

Note: Numbering Plan/Type in the Number Manipulation table, if present, overrides the value defined by the global parameter. Using the Number Manipulation tables enables configuration of the Numbering Plan and Type on a per call basis, according to destination (or source) number.

Same as the description for parameter ‘DestNumberType’.

Same as the description for parameter ‘DestNumberPlan’.

0 = Gateway accepts all SIP calls (default).

1 = Gateway accepts SIP calls only from IP addresses defined in the Tel to IP routing table. The gateway rejects all calls from unknown IP addresses.

For detailed information on the Tel to IP Routing table see Tel to IP Routing Table on page 52 .

Note: Specifying the IP address of a Proxy server in the Tel to IP Routing table enables the gateway to only accept calls originating in the Proxy server and rejects all other calls.

3Com VCX V7122 SIP VoIP Gateway User Manual 125

ini File Field Name

Web Parameter Name

*

Valid Range and Description

AltRouteCauseTel2IP

[Reasons for Alternative

Routing Table]

AltRouteCauseIP2Tel

[Reasons for Alternative

Routing Table]

Table of call failure reason values received from the IP side. If a call is released as a result of one of these reasons, the gateway tries to find an alternative route to that call in the ‘Tel to IP Routing’ table.

For example:

AltRouteCauseTel2IP = 486 (Busy here).

AltRouteCauseTel2IP = 480 (Temporarily unavailable).

AltRouteCauseTel2IP = 408 (No response).

Note 1: The 408 reason can be used to specify that there was no response from the remote party to the INVITE request.

Note 2: This parameter can appear up to 5 times.

Table of call failure reason values received from the pstn side (in Q.931 presentation). If a call is released as a result of one of these reasons, the gateway tries to find an alternative trunk group to that call in the ‘IP to Trunk Group Routing’ table.

For example:

AltRouteCauseIP2Tel = 3

AltRouteCauseIP2Tel = 1

(No route to destination).

(Unallocated number).

AltRouteCauseIP2Tel = 17 (Busy here).

Note 1: This parameter can appear up to 5 times.

Note 2: If the VCX V7122 fails to establish a cal to the PSTN because it has no available channels in a specific trunk group (e.g., all of the trunk group’s channels are occupied, or the trunk group’s spans are disconnected or out of sync), it uses the internal release cause ‘3’ (no route to destination). This cause can be used in the

‘AltRouteCauseIP2Tel’ table to define routing to an alternative trunk group.

FilterCalls2IP

[Filter Calls To IP]

Alternative Routing Parameters

0 = Disabled (default).

1 = Enabled.

If the filter calls to IP feature is enabled, then when a Proxy is used, the gateway first checks the Tel IP routing table before making a call through the Proxy. If the number is not allowed (number isn’t listed or a Call Restriction routing rule,

IP=0.0.0.0, is applied), the call is released.

AltRoutingTel2IPEnable

[Enable Alt Routing Tel to IP]

Operation modes of the Alternative Routing mechanism:

0 = Disabled (default).

1 = Enabled.

2 = Enabled for status only, not for routing decisions.

AltRoutingTel2IPMode

[Alt Routing Tel to IP Mode]

0 (None) = Alternative routing is not used.

1 (Conn) = Alternative routing is performed if ping to initial destination failed.

2 (QoS) = Alternative routing is performed if poor quality of service was detected.

3 (All) = Alternative routing is performed if, either ping to initial destination failed, or poor quality of service was detected, or DNS host name is not resolved (default).

Note: QoS is quantified according to delay and packet loss, calculated according to previous calls.

For information on the Alternative Routing feature,see Configuring the Gateway’s

Alternative Routing (based on Connectivity and QoS) on page 150 .

126 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

Web Parameter Name

*

Valid Range and Description

IPConnQoSMaxAllowedPL

[Max Allowed Packet Loss

for Alt Routing]

Packet loss percentage at which the IP connection is considered a failure.

The range is 1% to 20%. The default value is 20%.

IPConnQoSMaxAllowedDelay

[Max Allowed Delay for Alt

Routing]

Transmission delay (in msec) at which the IP connection is considered a failure.

The range is 100 to 1000. The default value is 250 msec.

E1/T1 Configuration Parameters

In Table 29 , parameters in brackets are the format in the Embedded Web Server

*

.

Table 29

E1/T1/J1 Configuration Parameters

ini File Field Name

Web Parameter Name

*

Valid Range and Description

PCMLawSelect

[PCM Law Select]

FramingMethod

[Framing Method]

1 = A-law

3 =

µ-Law

Usually A-Law is used for E1 spans and

µ-Law for T1 and J1 spans.

Selects the framing method to be used for E1/T1 spans.

For E1

0 = Multiframe with CRC4 (default, automatic mode, if CRC is identified in the Rx,

CRC is sent in Tx, otherwise no CRC). a = Double frame c = Multiframe with CRC4

For T1

0 or D = Extended super frame with CRC6 (default)

1 or B = Super frame D4, F12 (12-Frame multiframe)

A = F4 (4-Frame multiframe)

C = Extended super frame without CRC6

F = J1 - Japan (ESF with CRC6 and JT)

FramingMethod_x

[Framing Method]

ProtocolType

[Protocol Type]

Same as "FramingMethod" for a specific Trunk ID (x = 0 to 7)

Sets the PSTN protocol to be used for this trunk.

E1_EURO_ISDN

T1_CAS

T1_RAW_CAS

= 1

= 2

E1_MFCR2 =

E1_CAS_R2 =

E1_RAW_CAS

T1_NI2_ISDN

T1_4ESS_ISDN = 11

T1_NTT_ISDN

3Com VCX V7122 SIP VoIP Gateway User Manual

= 16 /* Japan - Nippon Telegraph

127

ini File Field Name

Web Parameter Name

*

Valid Range and Description

E1_AUSTEL_ISDN

T1_HKT_ISDN

E1_KOR_ISDN

T1_HKT_ISDN

E1_QSIG

T1_QSIG

= 17

= 18

= 19

= 20

= 21

= 23

/* Australian Telecom

/* Hong Kong - HKT

/* Korean operator

/* Hong Kong - HKT over T1

/*Basic call only

/*Basic call only

Note: The VCX V7122 simultaneously supports different variants of CAS and PRI protocols on different E1/T1 spans (no more than four simultaneous PRI variants).

ProtocolType_x

[Protocol Type]

TerminationSide

[ISDN Termination Side]

Same as "ProtocolType" for specific Trunk ID (x = 0 to 7)

Selects the ISDN termination side. Applicable only to ISDN protocols.

0 = ISDN User Termination Side (TE) (default).

1 = ISDN Network Termination Side (NT).

Note: Select ‘User Side’ when the PSTN or PBX side is configured as ‘Network side’, and vice-versa. If you don’t know the VCX V7122 ISDN termination side, choose

‘User Side’ and see the ‘Status & Diagnostics>Channel Status’ screen. If the Dchannel alarm is indicated, choose ‘Network Side’.

TerminationSide_x

[ISDN Termination Side]

ClockMaster

[Clock Master]

Same as "TerminationSide" for specific Trunk ID (x = 0 to 7).

ClockMaster_x

[Clock Master]

TDMBusClockSource

[TDM Bus Clock Source]

Same as "ClockMaster" for specific Trunk ID (x = 0 to 7)

TDMBusPSTNAutoClock

Enable

[TDM Bus PSTN Auto Clock]

0 = Recover the clock from first E1/T1 line (default).

1 = Recover the clock from any connected slave E1/T1 line.

This parameter is relevant only if "TDMBusClockSource = 4".

TDMBusLocalReference

[TDM Bus Local Reference]

1 = Generate clock from local source (default).

4 = Recover clock from PSTN line.

See Appendix E: VCX V7122 Clock Settings on page 211 for detailed information on configuring the gateway’s clock settings.

0 to 7 (default = 0).

Physical Trunk ID from which the gateway recovers its clock. Applicable only if

"TDMBusClockSource = 4" and "PSTNAutoClockEnable = 0".

LineCode

[Line Code]

0 = Recover clock from the E1/T1 line (default).

1 = The clock is generated by the gateway.

See Appendix E: VCX V7122 Clock Settings on page 211 for extended details of how to configure the gateway’s clock settings.

LineCode_x

[Line Code]

0 = use B8ZS line code (for T1 trunks only) default.

1 = use AMI line code.

2 = use HDB3 line code (for E1 trunks only).

Use to select B8ZS or AMI for T1 spans, and HDB3 or AMI for E1 spans.

Same as "LineCode" for a specific Trunk ID (x = 0 to 7)

128 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

Web Parameter Name

*

BchannelNegotiation

[B-channel Negotiation]

NFASGroupNumber_x

[NFAS Group Number]

DchConfig_x

[D-channel Configuration]

ISDNNFASInterfaceID_x

[NFAS Interface ID]

CASTableIndex_x

[CAS Table]

CASFileName_0

CASFileName_1

CASFileName_7

CASTablesNum

Valid Range and Description

Determines the ISDN B-Channel negotiation mode.

0 = Preferred.

1 = Exclusive (default).

Applicable to ISDN protocols.

0 = Non NFAS trunk (default).

1 to 4 = NFAS group number.

Indicates the NFAS group number (NFAS member) for the selected trunk.

"x" identifies the Trunk ID (0-7).

Trunks that belong to the same NFAS group have the same number.

With ISDN Non-Facility Associated Signaling you can use single D-channel to control multiple PRI interfaces.

Applicable only to T1 ISDN protocols.

0 = Primary Trunk (default).

1 = Backup Trunk.

2 = NFAS Trunk.

D-channel configuration parameter defines primary, backup (optional) and Bchannels only trunks.

"x" identifies the Trunk ID (0-7).

Primary trunk contains D-channel that is used for signaling.

Backup trunk contains backup D-channel that is used if the primary D-channel fails.

The other NFAS trunks contain only 24 B-channels, without a signaling D-channel.

Applicable only to T1 ISDN protocols.

Backup trunk is not supported for DMS PRI variants.

Defines a different Interface ID for each T1 trunk.

The valid range is 0 to 100.

The default interface ID equals to the trunk’s ID (0 to 7).

’x’ identifies the trunk ID (0-7)

Note: To set the NFAS interface ID, configure: ISDNIBehavior_x to include ‘512’ feature, per each T1 trunk.

Defines CAS protocol for each Trunk ID (x = 0 to 7) from a list of protocols defined by the "CASFileName_Y" parameter.

For example:

CASFileName_0 = 'E_M_WinkTable.dat'

CASFileName_1 = 'E_M_ImmediateTable.dat'

CASTableIndex_0 = 0

CASTableIndex_1 = 0

CASTableIndex_2 = 1

CASTableIndex_3 = 1

Trunks 0 and 1 use the E&M Winkstart CAS protocol, while trunks 2 and 3 use the

E&M Immediate Start CAS protocol.

CAS file name (such as "E_M_WinkTable.dat") defines the CAS protocol. It is possible to define up to 8 different CAS files by repeating the “CASFileName” parameter. Each CAS file can be associated with one or more of the gateway trunks using "CASTableIndex_x" parameter.

1 to 8. Indicates how many CAS protocol configurations files are loaded.

3Com VCX V7122 SIP VoIP Gateway User Manual 129

ini File Field Name

Web Parameter Name

*

Valid Range and Description

IdleABCDPattern

[Idle ABCD Pattern]

IdlePCMPattern

[Idle PCM Pattern]

LineBuildOut.Loss

[Line Build Out Loss]

Range 0x0 to 0xF.

Default = -1 (default pattern = 0000).

ABCD (CAS) Pattern to be applied to CAS signaling bus when the channel is idle.

This is only relevant when using PSTN interface with CAS protocols. Set to -1 for default.

Range 0x00 to 0xFF.

Default = -1 (default pattern = 0xFF for

µ-Law, 0x55 for A-law.

PCM Pattern to be applied to E1/T1 timeslot (B-channel) when the channel is idle.

0 = 0 dB (default)

1 = -7.5 dB

2 = -15 dB

3 = -22.5 dB

Selects the line build out loss to be used for T1 trunks.

N/A for E1 trunks.

ISDNRxOverlap

ISDNRxOverlap_x

[Enable Receiving of

Overlap Dialing]

TimeBetweenDigits

[Inter Digit Timeout for

Overlap Dial]

0 = Disabled (default).

1 = Enabled.

Any number bigger than one = Number of digits to receive.

Note 1: If enabled the VCX V7122 receives ISDN called number that is sent in the

"Overlap" mode.

Note 2: The INVITE to IP is sent only after the number (including “Sending Complete”

Info Element) was fully received (in SETUP and/or subsequent INFO Q.931 messages).

For detailed information on ISDN overlap dialing, see ISDN Overlap Dialing on page

144 .

Enable / disable Rx ISDN overlap per trunk ID (x = 0 to 7).

0 = Disabled (default).

1 = Enabled.

Note 1: If enabled, the VCX V7122 receives ISDN called number that is sent in the

"Overlap" mode.

Note 2: The SETUP message to IP is sent only after the number (including the

‘Sending Complete’ Info Element) was fully received (via SETUP and/or subsequent

INFO Q.931 messages).

Note3: The ‘MaxDigits’ parameter can be used to limit the length of the collected number for VCX V7122 ISDN overlap dialing (if sending complete was not received).

Defines the time (in seconds) that the gateway waits between digits that are received from the ISDN when Tel IP overlap dialing is performed. When this inter-digit timeout expires, the gateway uses the collected digits for the called destination number.

The range is 1 to 10 seconds. The default value is 4 seconds.

MaxDigits

[Max Digits In Phone Num for

Overlap Dialing]

Defines the maximum number of collected destination number digits received from the ISDN when Tel IP overlap dialing is performed. When the number of collected digits reaches the maximum, the gateway uses these digits for the called destination number.

The range is 1 to 49. The default value is 30.

R2Category

[MFC R2 Category]

MFC R2 Calling Party Category (CPC). The parameter provides information on calling party such as National or International call, Operator or Subscriber and

Subscriber priority. The parameter range is 1 to 15, defining one of the MFC R2 tones.

130 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

Web Parameter Name

*

Valid Range and Description

RegretTime

HeldTimeout

Determines the time period (in seconds) the gateway waits for an MFC R2 Resume

(Reanswer) signal once a Suspend (Clear back) signal was received from the PBX. If this timer expires, the call is released.

The valid range is 0 to 255. The default value is 0.

Applicable only for MFC R2 CAS Brazil variant.

Determines the time period the gateway can stay on-hold. If a Resume (un-hold Re-

Invite) message is received before the timer expires, the call is renewed. If this timer expires, the call is released.

-1 = Indefinitely (default).

0 - 2400 =Time to wait in seconds.

Currently applicable only to MFC R2 CAS variants.

ISDN Flexible Behavior Parameters

ISDN protocol is implemented in different Switches / PBXs by different vendors. Several implementations vary a little from the specification. Therefore, to provide a flexible interface that supports these ISDN variants, the ISDN behavior parameters are used.

ISDNInCallsBehavior

[Incoming Calls Behavior]

2048 = Sends Channel ID in the first response to an incoming Q.931 Call Setup message. Otherwise, the Channel ID is sent only if the gateway requires to change the proposed Channel ID (default).

8192 = Sends Channel ID in a Q.931 Call Proceeding message.

65536 = Includes Progress Indicator (PI=8) in Setup ACK message, if an empty called number is received in an incoming Setup message. Applicable to overlap dialing mode. The parameter also directs the gateway to play a dial tone, until the next called number digits are received.

262144 = NI-2 second redirect number – Users can select and use (in Invite messages) the NI-2 second redirect number, if two redirect numbers are received in

Q.931 Setup for incoming Tel IP calls.

Note: To configure the gateway to support several ‘ISDNInCallsBehavior’ features, summarize the individual feature values. For example to support both ‘2048’ and

‘65536’ features, set ‘ISDNInCallsBehavior = 67584.

ISDNIBehavior

[Q.931 Layer Response

Behavior]

1 = Q.931 Status message isn’t sent if Q.931 received message contains an unknown/unrecognized IE(s). By default the Status message is sent. This parameter applies only to PRI variants in which sending of Status message is optional.

2 = Q.931 Status message isn’t sent if an optional IE with invalid content is received.

By default the Status message is sent. This parameter applies only to PRI variants in which sending of Status message is optional.

4 = Accepts unknown/unrecognized Facility IE. Otherwise, (default) the Q.931 message that contains the unknown Facility IE is rejected. This parameter applies to

PRI variants where a complete ASN1 decoding is performed on Facility IE.

128 = Connect ACK message is sent in response to received Q.931 Connect.

Applicable only to Euro ISDN User side outgoing calls. Otherwise, the Connect ACK is not sent (default).

512 = Enables to configure T1 NFAS Interface ID (see the parameter

‘ISDNNFASInterfaceID_x’). Applicable to 4/5ESS, DMS, NI-2 and HKT variants.

2048 = Always set the Channel Identification IE to explicit Interface ID, even if the Bchannel is on the same trunk as the D-channel. Applicable to 4/5ESS, DMS and NI-2 variants.

3Com VCX V7122 SIP VoIP Gateway User Manual 131

ini File Field Name

Web Parameter Name

*

Valid Range and Description

ISDNGeneralCCBehavior

[General Call Control

Behavior]

ISDNOutCallsBehavior

[Outgoing Calls Behavior]

ISDNIBehavior_x

[Q.931 Layer Response

Behavior]

ISDNInCallsBehavior_x

[Incoming Calls Behavior]

ISDNOutCallsBehavior_x

[Outgoing Calls Behavior]

131072 = Clears the call on reception of Q.931 Status with incompatible state.

Otherwise, (default) no action is taken.

Note: To configure the gateway to support several ‘ISDNIBehavior’ features, summarize the individual feature values. For example to support both ‘512’ and ‘2048’ features, set ‘ISDNIBehavior = 2560’.

16 = The gateway clears down the call if it receives a Notify message specifying

‘User-Suspended’. A Notify (User-Suspended) message is used by some networks

(e.g., in Italy or Denmark) to indicate that the remote user has cleared the call, especially in the case of a long distance voice call.

32 = Applies only to ETSI E1 lines (30B+D). Enables handling the differences between the newer QSIG standard (ETS 300-172) and other ETSI-based standards

(ETS 300-102 and ETS 300-403) in the conversion of B-channel ID values into timeslot values:

In ‘regular ETSI’ standards, the timeslot is identical to the B-channel ID value, and the range for both is 1–15 and 17–31. The D-channel is identified as channel-id #16 and carried into the timeslot #16.

In newer QSIG standards, the channel-id range is 1–30, but the timeslot range is still

1–15 and 17–31. The D-channel is not identified as channel-id #16, but is still carried into the timeslot #16.

When this bit is set, the channel ID #16 is considered as a valid B-channel ID, but timeslot values are converted to reflect the range 1–15 and 17–31. This is the new

QSIG mode of operation. When this bit is not set (default), the channel_id #16 is not allowed, as for all ETSI-like standards.

1024 = Numbering plan / type for T1 IP Tel calling number are defined according to the manipulation tables or according to RPID header (default). Otherwise, the Plan / type for T1 calls are set according to the length of the calling number

Same as the description for parameter ‘ISDNBehavior’ for a specific trunk ID

(x = 0–7)

Same as the description for parameter ‘ISDNInCallsBehavior’ for a specific trunk ID

(x = 0–7)

Same as the description for parameter ‘ISDNOutCallsBehavior’ for a specific trunk ID

(x = 0–7)

Channel Parameters

The Channel Parameters define the DTMF, Fax and Modem transfer modes. See

Appendix D: Fax and Modem Transport Modes on page 209 for a detailed description of Fax and Modem transfer modes; see Redirect Number and Calling Name (Display) on page 143 for detailed description on DTMF transport modes.

Note that the Default Channel Parameters are applied to all VCX V7122 channels.

In Table 30 , parameters in brackets are the format in the Embedded Web Server

*

.

132 3Com VCX V7122 SIP VoIP Gateway User Manual

Table 30

Channel Parameters

ini File Field Name

Web Parameter Name

*

DJBufMinDelay

[Dynamic Jitter Buffer

Minimum Delay]

DJBufOptFactor

[Dynamic Jitter Buffer

Optimization Factor]

Valid Range and Description

0 to 150 msec (default = 70).

Dynamic Jitter Buffer Minimum Delay.

Note: For more information on the Jitter Buffer, see Dynamic Jitter Buffer Operation on page 136 .

Dynamic Jitter Buffer frame error / delay optimization factor.

You can enter a value from 0–13.

The default factor is 7.

Note 1: Set to 13 for data (fax & modem) calls.

Note 2: For more information on the Jitter Buffer, see Dynamic Jitter Buffer

Operation on page 136 .

FaxTransportMode

[Fax Transport Mode]

Fax Transport Mode that the gateway uses.

You can select:

0 = Disable.

1 = T.38 Relay (default).

2 = Bypass.

Note: If parameter IsFaxUsed = 1, then FaxTransportMode is always set to 1 (T.38 relay).

FaxRelayEnhancedRedundancy

Depth

[Fax Relay Enhanced

Redundancy Depth]

0–4 (default = 0).

Number of repetitions applied to control packets when using T.38 standard.

FaxRelayRedundancyDepth

[Fax Relay Redundancy Depth]

Number of times that each fax relay payload is retransmitted to the network.

You can enter a value from 0 to 2.

The default value is 0.

FaxRelayMaxRate

[Fax Relay Max Rate (bps)]

Limits the maximum rate at which fax messages are transmitted.

0 = 2.4 kbps.

1 = 4.8 kbps.

2 = 7.2 kbps.

3 = 9.6 kbps.

4 = 12.0 kbps.

5 = 14.4 kbps (default).

FaxRelayECMEnable

[Fax Relay ECM Enable]

FaxModemBypassCoderType

[Fax/Modem Bypass Coder

Type]

0 = Disable using ECM (Error Correction Mode) mode during Fax Relay.

1 = Enable using ECM mode during Fax Relay (default).

Coder the gateway uses when performing fax/modem bypass. Usually, high-bit-rate coders such as G.711 should be used.

You can select:

0 = G711 A-law 64 (default).

1 = G711

µ-law.

4 = G726 32.

11 = G726_40.

CNGDetectorMode

[CNG Detector Mode]

0 = Disable (default).

1 = Event Only (N/A).

2 = Relay. T.38 fax relay session is initiated by the originating fax if ‘IsFaxUsed = 1’.

Note that using this mode isn’t recommended.

3Com VCX V7122 SIP VoIP Gateway User Manual 133

ini File Field Name

Web Parameter Name

*

Valid Range and Description

FaxModemBypassM

[Fax/Modem Bypass Packing

Factor]

FaxBypassPayloadType

[Fax Bypass Payload Type]

ModemBypassPayloadType

DetFaxOnAnswerTone

[Detect Fax on Answer Tone]

FaxModemBypassBasicRTP

PacketInterval

FaxModemBypassDJBufMin

Delay

NSEMode

Number of (20 msec) coder payloads that are used to generate a Fax/Modem

Bypass packet.

You can enter a value of 1, 2 or 3 coder payloads.

The default value is 1 coder payload.

Determines the Fax Bypass RTP dynamic payload type.

The valid range is 96 to 120. The default value is 102.

Modem Bypass dynamic payload type (range 0-127).

The default value is 103.

0 = Starts T.38 procedure on detection of V.21 preamble (default).

1 = Starts T.38 Procedure on detection of CED fax answering tone.

0 = set internally (default).

1 = 5 msec (not recommended).

2 = 10 msec.

3 = 20 msec.

0 to 150 msec (default=40).

Determines the Jitter Buffer delay during Fax and Modem bypass session.

Cisco compatible fax and modem bypass mode.

0 = NSE disabled (default).

1 = NSE enabled.

Note 1: This feature can be used only if VxxModemTransportType=2 (Bypass).

Note 2: If NSE mode is enabled the SDP contains the following line:

“a=rtpmap:100 X-NSE/8000”

Note 3: To use this feature:

The Cisco gateway must include the following definition: "modem passthrough nse payload-type 100 codec g711alaw".

Set the Modem transport type to Bypass mode (‘VxxModemTransportType = 2’) for all modems.

Configure the gateway parameter NSEPayloadType= 100

In NSE bypass mode the gateway starts using G.711 A-Law (default) or G.711

µ-

Law, according to the parameter ‘FaxModemBypassCoderType’. The payload type used with these G.711 coders is a standard one (8 for G.711 A-Law and 0 for G.711

µ-Law). The parameters defining payload type for the “old” 3Com Bypass mode.

‘FaxBypassPayloadType’ and ‘ModemBypassPayloadType’ are not used with NSE

Bypass. The bypass packet interval is selected according to the parameter

‘FaxModemBypassBasicRtpPacketInterval’.

NSEPayloadType

V22ModemTransportType

[V.22 Modem Transport Type]

NSE payload type for Cisco Bypass compatible mode.

The valid range is 96-127. The default value is 105.

Note: Cisco gateways usually use NSE payload type of 100.

V.22 Modem Transport Type that the gateway uses.

You can select:

0 = Transparent.

2 = Modem Bypass (default).

V23ModemTransportType

[V.23 Modem Transport Type]

V.23 Modem Transport Type that the gateway uses.

You can select:

0 = Transparent .

2 = Modem Bypass (default).

134 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

Web Parameter Name

*

Valid Range and Description

V32ModemTransportType

[V.32 Modem Transport Type]

V.32 Modem Transport Type that the gateway uses.

You can select:

0 = Transparent.

2 = Modem Bypass (default).

Note: This option applies to V.32 and V.32bis modems.

V34ModemTransportType

[V.34 Modem Transport Type]

InputGain

[Input Gain]

V.34 Modem Transport Type that the gateway uses.

You can select:

0 = Transparent.

2 = Modem Bypass (default).

Note: This option applies to V.34 and V.90 modems.

PCM input gain control in dB. This parameter sets the level for the received

(PSTN IP) signal.

You can enter a value from -32 to 31 dB.

The default value is 0 dB.

Note: This parameter is intended for advanced users. Changing it affects other gateway functionalities.

VoiceVolume

[Voice Volume]

Voice gain control in dB. This parameter sets the level for the transmitted

(IP PSTN) signal.

You can enter a value from -32 to 31 dB.

The default value is 0 dB.

0 = Disable redundancy packets generation (default).

1 = Enable generation of RFC 2198 redundancy packets.

RTPRedundancyDepth

[RTP Redundancy Depth]

RFC2198PayloadType RTP redundancy packet payload type, according to RFC 2198.

The range is 96-127. The default is 104.

Applicable if “RTPRedundancyDepth=1”.

EnableSilenceCompression

[Silence Suppression]

The parameter SCE is used to maintain backward compatibility.

0 = Silence Suppression disabled (default).

1 = Silence Suppression enabled.

2 [Enable without adaptation] = A single silence packet is sent during silence period

(applicable only to G.729).

Silence Suppression is a method conserving bandwidth on VoIP calls by not sending packets when silence is detected.

EnableEchoCanceller

[Echo Canceler]

The parameter ECE is used to maintain backward compatibility.

0 = Echo Canceler disabled.

1 = Echo Canceler Enabled (default).

Note: Refer also to the parameters ‘MaxEchoCancellerLength’ and

‘EchoCancellerLength’ (described in Table 24 on page 94 ).

EnableStandardSIDPayloadType

[Enable RFC 3389 CN Payload

Type]

0 = Disable (default).

1 = Enable.

If enabled, the SID (comfort noise) packets are sent with the RTP SID payload type according to RFC 3389. Applicable to G.711 and G.726 coders.

If disabled, the G.711 SID packets are sent in a proprietary method.

DTMFVolume

[DTMF Volume]

-31 to 0 corresponding to -31 dBm to 0 dBm in 1 dB steps (default = -11 dBm) DTMF gain control.

3Com VCX V7122 SIP VoIP Gateway User Manual 135

ini File Field Name

Web Parameter Name

*

DTMFTransportType

[DTMF Transport Type]

RFC2833PayloadType

[RFC 2833 Payload Type]

MGCPDTMFDetectionPoint

DTMFInterDigitInterval

DTMFDigitLength

Valid Range and Description

0 = Erase digits from voice stream, do not relay to remote.

2 = Digits remain in voice stream.

3 = Erase digits from voice stream, relay to remote according to RFC 2833.

Note: This parameter is automatically updated if one of the following parameters is configured: IsDTMFUsed, TxDTMFOption or RxDTMFOption.

The RFC 2833 DTMF relay dynamic payload type.

Range: 96 to 99, 106 to 127; Default = 96.

The 100, 102 to 105 range is allocated for proprietary usage.

Cisco is using payload type 101 for RFC 2833.

Note: When RFC 2833 payload type (PT) negotiation is used (TxDTMFOption=4), this payload type is used for the received DTMF packets. If negotiation isn’t used, this payload type is used for receive and for transmit.

0 = DTMF event is reported on the start of a detected DTMF digit.

1 = DTMF event is reported on the end of a detected DTMF digit (default).

Note: The parameter is used for out-of-band dialing.

Time in msec between generated DTMFs to PSTN side.

Default = 100 (msec).

Time in msec for generating of DTMF tone to PSTN side.

Default = 100 (msec).

Dynamic Jitter Buffer Operation

Voice frames are transmitted at a fixed rate. If the frames arrive at the other end at the same rate, voice quality is perceived as good. In many cases, however, some frames can arrive slightly faster or slower than the other frames. This is called jitter (delay variation), and degrades the perceived voice quality. To minimize this problem, the gateway uses a jitter buffer. The jitter buffer collects voice packets, stores them and sends them to the voice processor in evenly spaced intervals.

The VCX V7122 uses a dynamic jitter buffer that can be configured using two parameters:

Minimum delay, ‘DJBufMinDelay’ (0 msec to 150 msec). Defines the starting jitter capacity of the buffer. For example, at 0 msec, there is no buffering at the start. At the default level of 70 msec, the gateway always buffers incoming packets by at least 70 msec worth of voice frames.

Optimization Factor, ‘DJBufOptFactor’ (0 to 12, 13). Defines how the jitter buffer tracks to changing network conditions. When set at its maximum value of 12, the dynamic buffer aggressively tracks changes in delay (based on packet loss statistics) to increase the size of the buffer and doesn’t decays back down. This results in the best packet error performance, but at the cost of extra delay. At the minimum value of 0, the buffer tracks delays only to compensate for clock drift and quickly decays back to the minimum level.

This optimizes the delay performance but at the expense of a higher error rate.

The default settings of 70 msec Minimum delay and 7 Optimization Factor should provide a good compromise between delay and error rate. The jitter buffer "holds" incoming packets for

70 msec before making them available for decoding into voice. The coder polls frames from the buffer at regular intervals to produce continuous speech. As long as delays in the network do not change (jitter) by more than 70 msec from one packet to the next, there is

136 3Com VCX V7122 SIP VoIP Gateway User Manual

always a sample in the buffer for the coder to use. If there is more than 70 msec of delay at any time during the call, the packet arrives too late. The coder tries to access a frame and is not able to find one. The coder must produce a voice sample even if a frame is not available.

It therefore compensates for the missing packet by adding a Bad-Frame-Interpolation (BFI) packet. This loss is then flagged as the buffer being too small. The dynamic algorithm then causes the size of the buffer to increase for the next voice session. The size of the buffer may decrease again if the gateway notices that the buffer is not filling up as much as expected. At no time does the buffer decrease to less than the minimum size configured by the Minimum delay parameter.

Special Optimization Factor Value: 13

One of the purposes of the Jitter Buffer mechanism is to compensate for clock drift. If the two sides of the VoIP call are not synchronized to the same clock source, one RTP source generates packets at a lower rate, causing under-runs at the remote Jitter Buffer. In normal operation (optimization factor 0 to 12), the Jitter Buffer mechanism detects and compensates for the clock drift by occasionally dropping a voice packet or by adding a BFI packet.

Fax and modem devices are sensitive to small packet losses or to added BFI packets.

Therefore to achieve better performance during modem and fax calls, the Optimization

Factor should be set to 13. In this special mode the clock drift correction is performed less frequently - only when the Jitter Buffer is completely empty or completely full. When such condition occurs, the correction is performed by dropping several voice packets simultaneously or by adding several BFI packets simultaneously, so that the Jitter Buffer returns to its normal condition.

Configuration Files Parameters

The configuration files (Call Progress Tones, PRT, Voice Prompts and CAS) can be loaded to the VCX V7122 via the Embedded Web Server (see Auxiliary Files on page 86 ), or via

TFTP session.

To load the configuration files via TFTP, follow these steps:

1 In the ini file, define the files to be loaded to the device. You can also define in the ini file whether the loaded files should be stored in the non-volatile memory so that the TFTP process is not required every time the device boots up.

2 Locate the configuration files you want to load and the ini file in the same directory.

3 Invoke a BootP/TFTP session; the ini and configuration files are loaded onto the device.

Table 31 below describes the ini file parameters that are associated with the configuration files.

Table 31

Configuration File Parameters

ini File Field Name

CallProgressTonesFilename

VoicePromptsFileName

Valid Range and Description

The name of the file containing the Call Progress Tones definitions. See Chapter 7:

Configuration Files on page 139 for additional information on how to create and load this file.

The name (and path) of the file containing the Voice Prompts definitions. See

Prerecorded Tones (PRT) File on page 141 for additional information on how to create and load this file.

3Com VCX V7122 SIP VoIP Gateway User Manual 137

ini File Field Name

CASfilename

CASfilename_x

CASTablesNum

PrerecordedTonesFileName

SaveConfiguration

Valid Range and Description

This is the name of the file containing specific CAS protocol definition (such as

‘E_M_WinkTable.dat’). These files are provided to support various types of CAS signaling.

It is possible to load up to 8 different CAS files (x=0 to 7), by repeating the

CASFileName parameter. Each CAS file can be associated with one or more of the gateway trunks, using "CASTableIndex_x" parameter.

Number, 1 to 8. Specifies how many CAS configuration files are loaded.

The name (and path) of the file containing the Prerecorded Tones.

Set to 1 to store the CPT, PRT, CAS and Voice Prompts files in the non-volatile memory.

138 3Com VCX V7122 SIP VoIP Gateway User Manual

C

HAPTER

7: C

ONFIGURATION

F

ILES

This section describes the configuration (dat) files that are loaded (in addition to the ini file) to the gateway. The configuration files are:

Call Progress Tones file (see Configuring the Call Progress Tones below).

Prerecorded Tones file (see Prerecorded Tones (PRT) File on page 141 ).

Voice Prompts file (see Voice Prompts File on page 142 ).

CAS protocol configuration files (see CAS Protocol Configuration Files on page 142 ).

To load any of the configuration files to the VCX V7122 use the Embedded Web Server (see

Auxiliary Files on page 86 ) or alternatively specify the name of the relevant configuration file in the gateway’s ini file and load it (the ini file) to the gateway (see Loading the cmp File,

Booting the Device on page 194 ).

Configuring the Call Progress Tones

The Call Progress Tones, configuration file used by the VCX V7122 is a binary file (with the extension dat) which contains the definitions of the Call Progress Tones (levels and frequencies) that are detected / generated by the VCX V7122.

Users can either use, one of the supplied VCX V7122 configuration (dat) files, or construct their own file. To construct their own configuration file, users are recommended, to modify the supplied usa_tone.ini file (in any standard text editor) to suit their specific requirements, and to convert it (the modified ini file) into binary format using the “TrunkPack Downloadable

Conversion Utility” supplied with the software package. For the description of the procedure on how to convert CPT ini file to a binary dat file, see Converting a CPT ini File to a Binary dat File on page 220 .

Note that only the dat file can be loaded to the VCX V7122 gateway.

To load the Call Progress Tones (dat) file to the VCX V7122, use the Embedded Web Server

(see Auxiliary Files on page 86 ) or the ini file (see Dynamic Jitter Buffer Operation on page 136 ).

Format of the Call Progress Tones Section in the ini File

Using the CPT section of this configuration file, the User can create up to 16 different Call

Progress Tones using up to 15 different frequencies (in the range of 300 Hz to 1980 Hz).

Each of these Call Progress Tones is specified by its tone frequency (either single or dual frequencies are supported) and its tone cadence. The tone cadence is specified by 2 sets of on/off periods (you can discard the use of the first on/off cycle by setting the relevant parameters to zero). When a tone is composed of a single frequency, the second frequency field must be set to zero.

For a continuous tone (such as dial tone), only the “First Signal On time” should be specified.

In this case, the parameter specifies the detection period. For example, if it equals 300, the tone is detected after 3 seconds (300 x 10 msec). The minimum detection time is 100 msec.

3Com VCX V7122 SIP VoIP Gateway User Manual 139

Users can specify several tones of the same type. These additional tones are used only for tone detection. Generation of a specific tone conforms to the first definition of the specific tone. For example, Users can define an additional dial tone by appending the second dial tone’s definition lines to the first tone definition in the ini file. The VCX V7122 reports dial tone detection if either of the two tones is detected.

The Call Progress Tones section of the ini file format starts from the following string:

[NUMBER OF CALL PROGRESS TONES] – Contains the following key:

“Number of Call Progress Tones” defining the number of Call Progress Tones that are defined in the file.

[CALL PROGRESS TONE #X] – containing the Xth tone definition (starting from 1 and not exceeding the number of Call Progress Tones defined in the first section) using the following keys:

Tone Type – Call Progress Tone type

Figure 56

Call Progress Tone Types

7. Reorder Tone

17.Call Waiting Ringback Tone

23. Hold Tone

Low Freq [Hz] – Frequency in hertz of the lower tone component in case of dual frequency tone, or the frequency of the tone in case of single tone.

High Freq [Hz] – Frequency in hertz of the higher tone component in case of dual frequency tone, or zero (0) in case of single tone.

Low Freq Level [-dBm] – Generation level 0 dBm to –31 dBm in [dBm].

High Freq Level – Generation level. 0 to –31 dBm. The value should be set to ‘32’ in the case of a single tone.

First Signal On Time [10 msec] – “Signal On” period (in 10 msec units) for the first cadence on-off cycle.

First Signal Off Time [10 msec] – “Signal Off” period (in 10 msec units) for the first cadence on-off cycle.

Second Signal On Time [10 msec] – “Signal On” period (in 10 msec units) for the second cadence on-off cycle.

Second Signal Off Time [10 msec] – “Signal Off” period (in 10 msec units) for the second cadence on-off cycle.

Default Duration [msec] – The default duration (in 1 msec units) of the generated tone.

When the same frequency is used for a continuous tone and a cadence tone, the

‘Signal On Time’ parameter of the continues tone must have a value that is greater than the ‘Signal On Time’ parameter of the cadence tone. Otherwise the continues tone is detected instead of the cadence tone.

The tones frequency should differ by at least 40 Hz from one tone to other defined tones.

140 3Com VCX V7122 SIP VoIP Gateway User Manual

For example: To configure the dial tone to 440 Hz only, define the following text:

Figure 57

Defining a Dial Tone Example

#Dial tone

[CALL PROGRESS TONE #1]

Tone Type=1

Low Freq [Hz]=440

High Freq [Hz]=0

Low Freq Level [-dBm]=10 (-10 dBm)

High Freq Level [-dBm]=32 (use 32 only if a single tone is required)

First Signal On Time [10msec]=300; the dial tone is detected after 3 sec

First Signal Off Time [10msec]=0

Second Signal On Time [10msec]=0

Second Signal Off Time [10msec]=0

Prerecorded Tones (PRT) File

The Call Progress Tones mechanism has several limitations, such as a limited number of predefined tones and a limited number of frequency integrations in one tone. To work around these limitations and provide tone generation capability that is more flexible, the PRT file can be used. If a specific prerecorded tone exists in the PRT file, it takes precedence over the same tone that exists in the CPT file and is played instead of it.

Note that the prerecorded tones are used only for generation of tones. Detection of tones is performed according to the CPT file.

PRT File Format

The PRT dat file contains a set of prerecorded tones to be played by the VCX V7122 during operation. Up to 40 tones can be stored in a single file. The prerecorded tones (raw data

PCM or L8 files) are prepared offline using standard recording utilities (such as CoolEdit

TM

) and combined into a single file using the TrunkPack Downloadable Conversion utility (see

Creating a Loadable Prerecorded Tones File on page 224 ).

The raw data files must be recorded with the following characteristics:

Coders: G.711 A-law, G.711 µ-law or Linear PCM

Rate: 8 kHz

The generated PRT file can then be loaded to the VCX V7122 using the BootP/TFTP utility

(see Configuration Files Parameters on page 137 ) or via the Embedded Web Server (see

Auxiliary Files on page 86 ).

The maximum size of the combined PRT file that can be stored in RAM is 1 MB. The PRT file is permanently stored in flash memory only if there is available space. Otherwise, it should be loaded (using the BootP/TFTP utility) each time the VCX V7122 is reset.

The prerecorded tones are played repeatedly. This enables you to record only part of the tone and play it for the full duration. For example, if a tone has a cadence of 2 seconds on and 4 seconds off, the recorded file should contain only these 6 seconds. The PRT module repeatedly plays this cadence for the configured duration. Similarly, a continuous tone can be played by repeating only part of it.

3Com VCX V7122 SIP VoIP Gateway User Manual 141

Voice Prompts File

The Voice Prompts file is applicable only to the VXML application.

The voice announcement file contains a set of Voice Prompts to be played by the VCX

V7122 during operation. The voice announcements are prepared offline using standard recording utilities and combined into a single file using the TrunkPack Downloadable

Conversion Utility.

The generated announcement file can then be loaded to the VCX V7122 using the

BootP/TFTP utility (see Dynamic Jitter Buffer Operation on page 136 ) or via the Embedded

Web Server (see Auxiliary Files on page 86 ).

If the size of the combined Voice Prompts file is less than 1 MB, it can permanently be stored in flash memory. Larger files, up to 10 MB, are stored in RAM, and should be loaded again

(using BootP/TFTP utility) after the VCX V7122 is reset.

The Voice Prompts integrated file is a collection of raw voice recordings and / or wav files.

These recordings can be prepared using standard utilities, such as CoolEdit, Goldwave and others. The raw voice recordings must be sampled at 8000 kHz / mono / 8 bit. The wav files must be recorded with G.711

µ-Law/A-Law/Linear.

When the list of recorded files is converted to a single voiceprompts.dat file, every Voice

Prompt is tagged with an ID number, starting with “1”. This ID is used later by the VCX

V7122 to start playing the correct announcement. Up to 1000 Voice Prompts can be used.

The Voice Prompt ID is used in the VoiceXML file to specify the message that is to be played.

3Com provides a professionally recorded English (U.S.) Voice Prompts file.

To generate and load the Voice Prompts file, follow these steps:

1 Prepare one or more voice files using standard utilities.

2 Use the TrunkPack Downloadable Conversion Utility to generate the voiceprompts.dat file from the pre-recorded voice messages (see Creating a Loadable Voice Prompts File on page 221 ).

3 Load the voiceprompts.dat file to the VCX V7122 either by using a TFTP procedure (see

Dynamic Jitter Buffer Operation on page 136 ), or via the Embedded Web Server (see

Auxiliary Files on page 86 ).

CAS Protocol Configuration Files

The CAS Protocol Configuration Files contain the CAS Protocol definitions to be used for

CAS-terminated trunks. Users can either use the files supplied or construct their own files.

It is possible to load up to eight files and to use different files for different trunks.

Note that all CAS files loaded together must belong to the same Trunk Type (either E1 or T1).

142 3Com VCX V7122 SIP VoIP Gateway User Manual

C

HAPTER

8: G

ATEWAY

C

APABILITIES

D

ESCRIPTION

Proxy or Registrar Registration Example

REGISTER sip:servername SIP/2.0

VIA: SIP/2.0/UDP 212.179.22.229;branch=z9hG4bRaC7AU234

From: <sip:[email protected]>;tag=1c29347

To: <sip:[email protected]>

Call-ID: [email protected]

Seq: 1 REGISTER

Expires: 3600

Contact: sip:[email protected]

Content-Length: 0

The "servername" string is defined according to the following rules:

The "servername" is equal to "RegistrarName" if configured. The "RegistrarName" can be any string.

Otherwise, the "servername" is equal to "RegistrarIP" (either FQDN or numerical IP address), if configured.

Otherwise the "servername" is equal to "ProxyName" if configured. The "ProxyName" can be any string.

Otherwise the "servername" is equal to "ProxyIP" (either FQDN or numerical IP address).

The parameter ‘GWRegistrationName’ can be any string. If the parameter is not defined, the parameter ‘UserName’ is used instead.

The "sipgatewayname" parameter (defined in the ini file or set from the Web browser), can be any string. Some Proxy servers require that the "sipgatewayname" (in Register messages) is set equal to the Registrar / Proxy IP address or to the Registrar / Proxy domain name.

The Register message is sent to the Registrar’s IP address (if configured) or to the Proxy’s

IP address. The message is sent once per gateway. The registration request is resent according to the parameter ‘RegistrationTimeDivider’. For example, if

‘RegistrationTimeDivider = 70’ (%) and Registration Expires time = 3600, the gateway resends its registration request after 3600 x 70% = 2520 sec. The default value of

‘RegistrationTimeDivider’ is 50%.

Redirect Number and Calling Name (Display)

The following tables define the VCX V7122 redirect number and calling name (Display) support for various PRI variants:

3Com VCX V7122 SIP VoIP Gateway User Manual 143

Table 32

Calling Name (Display)

NT TE Yes Yes No

TE NT Yes Yes No

Table 33

Redirect Number

Yes

No

NT TE Yes Yes Yes

TE NT Yes Yes Yes

Yes

No

ISDN Overlap Dialing

Overlap dialing is a dialing scheme used by several ISDN variants to send and / or receive called number digits one right after the other (or several at a time). As opposed to the enbloc dialing scheme in which a complete number is sent.

The VCX V7122 can optionally support ISDN overlap dialing for incoming ISDN calls for the entire gateway by setting ‘ISDNRxOverlap’ to 1, or per E1/T1 span by setting

‘ISDNRxOverlap_x’ to 1 (‘x’ represents the number of the trunk, 0–7).

To play a Dial tone to the ISDN user side when an empty called number is received, set

‘ISDNINCallsBehavior = 65536’ (bit #16) causing the Progress Indicator to be included in the

SetupAck ISDN message.

The VCX V7122 stops collecting digits (for ISDN IP calls) when:

The sending device transmits a "sending complete" IE in the ISDN Setup or the following

Info messages to signal that no more digits are going to be sent.

The inter-digit timeout (configured by the parameter ‘TimeBetweenDigits’) expires. The default for this timeout is 4 seconds.

The maximum allowed number of digits (configured by the parameter ‘MaxDigits’) is reached. The default is 30 digits.

Relevant parameters (described in Table 30 on page 133 ):

ISDNRxOverlap

ISDNRxOverlap_x

TimeBetweenDigits

MaxDigits

ISDNInCallsBehavior

Using ISDN NFAS

In regular (non-NFAS) T1 ISDN trunks, a single 64 kbps channel carries signaling for the other 23 B-channels of that particular T1 trunk. This channel is called the D-channel and usually resides on timeslot # 24.

144 3Com VCX V7122 SIP VoIP Gateway User Manual

The ISDN Non-Facility Associated Signaling (NFAS) feature enables use of a single Dchannel to control multiple PRI interfaces.

With NFAS it is possible to define a group of T1 trunks, called an NFAS group, in which a single D-channel carries ISDN signaling messages for the entire group. The NFAS group’s

B-channels are used to carry traffic, such as voice or data. The NFAS mechanism also enables definition of a backup D-channel on a different T1 trunk, to be used if the primary Dchannel fails.

The NFAS group comprises several T1 trunks. Each T1 trunk is called an ‘NFAS member’.

The T1 trunk whose D-channel is used for signaling is called the ‘Primary NFAS Trunk’. The

T1 trunk whose D-channel is used for backup signaling is called the ‘Backup NFAS Trunk’.

The primary and backup trunks each carry 23 B-channels while all other NFAS trunks each carry 24 B-channels.

The VCX V7122 supports multiple NFAS groups. Each group should contain different T1 trunks.

The NFAS group is identified by an NFAS GroupID number (possible values are 1, 2, 3 and

4). To assign a number of T1 trunks to the same NFAS group, use the parameter

‘NFASGroupNumber_x = groupID’. ‘x’ stands for the physical trunkID (0–7).

The parameter ‘DchConfig_x = Trunk_type’ is used to define the type of NFAS trunk.

Trunk_type is set to 0 for the primary trunk, to 1 for the backup trunk and to 2 for an ordinary

NFAS trunk. ‘x’ stands for the physical trunkID (0–7).

For example, to assign the first four VCX V7122 T1 trunks to NFAS group #1, in which trunk

#0 is the primary trunk and trunk #1 is the backup trunk, use the following configuration:

NFASGroupNumber_0 = 1

NFASGroupNumber_1 = 1

NFASGroupNumber_2 = 1

NFASGroupNumber_3 = 1

DchConfig_0 = 0

DchConfig_1 = 1

DchConfig_2 = 2

DchConfig_3 = 2

;Primary T1 trunk

;Backup T1 trunk

;24 B-channel NFAS trunk

;24 B-channel NFAS trunk

The NFAS parameters are described in Table 29 on page 127 .

NFAS Interface ID

Several ISDN switches require an additional configuration parameter per T1 trunk, that is called ‘Interface Identifier’. In NFAS T1 trunks the Interface Identifier is sent explicitly in

Q.931 Setup / Channel Identification IE for all NFAS trunks, except for the B-channels of the

Primary trunk (see note 1 below).

The Interface ID can be defined per each member (T1 trunk) of the NFAS group, and must be coordinated with the configuration of the Switch.

The default value of the Interface ID is identical to the number of the physical T1 trunk (0 for the first VCX V7122 trunk, 1 for the second VCX V7122 T1 trunk etc. up to 7).

To define an explicit Interface ID for a T1 trunk (that is different from the default), use the following parameters:

3Com VCX V7122 SIP VoIP Gateway User Manual 145

ISDNBehavior_x = 512 (x = 0–7 identifying the VCX V7122 physical trunk)

ISDNNFASInterfaceID_x = ID (x = 0 to 255)

Usually the Interface Identifier is included in the Q.931 Setup/Channel

Identification IE only on T1 trunks that do not contain the D-channel. Calls initiated on B-channels of the Primary T1 trunk, by default, don’t contain the Interface

Identifier. Setting the parameter ‘ISDNBehavior_x’ to 2048 forces the inclusion of the Channel Identifier parameter also for the Primary trunk.

The parameter ‘ISDNNFASInterfaceID_x = ID’ can define the ‘Interface ID’ for any

Primary T1 trunk, even if the T1 trunk is not a part of an NFAS group. However, to include the Interface Identifier in Q.931 Setup/Channel Identification IE configure

‘ISDNBehavior_x = 2048’ in the ini file.

Working with DMS-100 Switches

The DMS-100 switch requires the following NFAS Interface ID definitions:

InterfaceID #0 for the Primary trunk

InterfaceID #1 for the Backup trunk (see the note below)

InterfaceID #2 for a 24 B-channel T1 trunk

InterfaceID #3 for a 24 B-channel T1 trunk

Etc.

Note: In the current version, the VCX V7122 doesn’t support the DMS-100 Backup trunk.

Therefore, InterfaceID #1, should not be used.

For example, if four T1 trunks on a VCX V7122 are configured as a single NFAS group, that is used with a DMS-100 switch, the following parameters should be used:

ISDNNFASInterfaceID_0 = 0

ISDNNFASInterfaceID_1 = 2

ISDNNFASInterfaceID_2 = 3

ISDNNFASInterfaceID_3 = 4

NFASGroupNumber_0 = 1

NFASGroupNumber_1 = 1

NFASGroupNumber_2 = 1

NFASGroupNumber_3 = 1

DchConfig_0 = 0 ;Primary T1 trunk

DchConfig_2 = 2 ;24 B-channel NFAS trunk

DchConfig_3 = 2 ;24 B-channel NFAS trunk

DchConfig_4 = 2 ;24 B-channel NFAS trunk

Configuring the DTMF Transport Types

You can control the way DTMF digits are transported over the IP network to the remote endpoint. The following five modes are supported:

Using INFO message according to the Nortel IETF draft:

In this mode DTMF digits are carried to the remote side within INFO messages.

To enable this mode set:

‘Enable DTMF = Yes’ (IsDTMFUsed = 1)

146 3Com VCX V7122 SIP VoIP Gateway User Manual

‘OutofBandDTMFFormat 1’

‘RxDTMFOption = 0’

Note that in this mode DTMF digits are erased from the audio stream (DTMFTransportType is automatically set to 0).

Using INFO message according to Cisco’s style:

In this mode DTMF digits are carried to the remote side within INFO messages.

To enable this mode set:

‘Enable DTMF = Yes’ (IsDTMFUsed = 1)

‘OutofBandDTMFFormat 2’

‘RxDTMFOption = 0’

Note that in this mode DTMF digits are erased from the audio stream (DTMFTransportType is automatically set to 0).

Using NOTIFY messages according to <draft-mahy-sipping-signaled-digits-01.txt>:

In this mode DTMF digits are carried to the remote side using NOTIFY messages.

To enable this mode set:

‘Enable DTMF = Yes’ (IsDTMFUsed = 1)

‘OutofBandDTMFFormat 3’

‘RxDTMFOption = 0’

Note that in this mode DTMF digits are erased from the audio stream (DTMFTransportType is automatically set to 0).

Using RFC 2833 relay with Payload type negotiation:

In this mode, DTMF digits are carried to the remote side as part of the RTP stream in accordance with RFC 2833 standard.

To enable this mode set:

‘Enable DTMF = No’ (IsDTMFUsed = 0)

‘DTMF RFC 2833 Negotiation = Yes’ (TxDTMFOption=4)

‘RxDTMFOption = 3’

‘DTMFTransportType = 3’

Note that to set the RFC 2833 payload type with a different value (other than its default, 96) configure the ‘RFC2833PayloadType’ parameter. The gateway negotiates the RFC 2833 payload type using local and remote SDP and sends packets using the PT from the received

SDP. The gateway expects to receive RFC 2833 packets with the same PT as configured by the ‘RFC2833PayloadType’ parameter. The RFC 2833 packets are sent even if the remote side didn't include the send "telephone-event" parameter in its SDP, in which case the gateway uses the same PT for send and for receive.

Sending DTMF digits (in RTP packets) as part of the audio stream (DTMF Relay is disabled):

Note that this method is normally used with G.711 coders; with other Low Bit Rate (LBR) coders the quality of the DTMF digits is reduced.

To set this mode:

‘Enable DTMF = No’ (IsDTMFUsed = 0)

‘DTMF RFC 2833 Negotiation = No’ (TxDTMFOption=0)

3Com VCX V7122 SIP VoIP Gateway User Manual 147

‘RxDTMFOption = 0’

‘DTMFTransportType = 2’

The gateway is always ready to receive DTMF packets over IP, in all possible transport modes: INFO messages, Notify and RFC 2833 (in proper payload type) or as part of the audio stream.

To exclude RFC 2833 Telephony event parameter from the gateway’s SDP, set

‘RxDTMFOption = 0’ in the ini file.

The following parameters affect the way the VCX V7122 SIP handles the DTMF digits:

Table 34

Summary of DTMF Configuration Parameters

ini File Field Name

[Web Name]

IsDTMFUsed

[Use Out-of-Band DTMF]

OutOfBandDTMFFormat

[Out-of-Band DTMF

Format]

Valid Range and Description

Use out-of-band signaling to relay DTMF digits.

0 = Disable, DTMF digits are sent according to DTMFTransportType parameter (default).

1 = Enable sending DTMF digits within INFO or NOTIFY messages.

Note: When out-of-band DTMF transfer is used, DTMFTransportType is automatically set to 0 (erase the DTMF digits from the RTP stream).

The exact method to send out-of-band DTMF digits.

1 = INFO format (Nortel).

2 = INFO format (Cisco) - (default).

3 = NOTIFY format <draft-mahy-sipping-signaled-digits-01.txt>.

Note 1: To use out-of-band DTMF, set “IsDTMFUsed=1” or “Enable DTMF = yes”.

Note 2: When using out-of-band DTMF, the “DTMFTransportType” parameter is automatically set to 0, to erase the DTMF digits from RTP path.

TxDTMFOption

[DTMF RFC 2833

Negotiation]

0 = No negotiation, DTMF digit is sent according to the parameters ‘DTMFTransportType’ and ‘RFC2833PayloadType’.

4 = Enable RFC 2833 payload type (PT) negotiation.

Note 1: This parameter is applicable only if “IsDTMFUsed=0” (out of-band DTMF is not used).

Note 2: If enabled, the gateway:

Negotiates RFC 2833 payload type using local and remote SDPs.

Sends DTMF packets using RFC 2833 PT according to the PT in the received SDP.

Expects to receive RFC 2833 packets with the same PT as configured by the

“RFC2833PayloadType” parameter.

Note 3: If the remote party doesn’t include the RFC 2833 DTMF relay payload type in the

SDP, the gateway uses the same PT for send and for receive.

Note 4: If TxDTMFOption is set to 0, the RFC 2833 payload type is set according to the parameter ‘RFC2833PayloadType’ for both transmit and receive.

148 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name

[Web Name]

RxDTMFOption

Valid Range and Description

RFC2833PayloadType

Defines the supported Receive DTMF negotiation method.

0 = Don’t declare RFC 2833 Telephony-event parameter in SDP.

1 = n/a.

2 = n/a.

3 = Declare RFC 2833 “Telephony-event” parameter in SDP (default).

The gateway is designed to always be receptive to RFC 2833 DTMF relay packets.

Therefore, it is always correct to include the “Telephony-event” parameter as a default in the SDP. However some gateways use the absence of the “telephony-event” from the

SDP to decide to send DTMF digits inband using G.711 coder. If this is the case you can set “RxDTMFOption=0”.

The RFC 2833 DTMF relay dynamic payload type.

Range: 96 to 99, 106 to 127; Default = 96.

The 100, 102 to 105 range is allocated for proprietary usage.

Cisco is using payload type 101 for RFC 2833.

Note: When RFC 2833 payload type (PT) negotiation is used (TxDTMFOption=4), this payload type is used for the received DTMF packets. If negotiation isn’t used, this payload type is used for receive and for transmit.

MGCPDTMFDetectionPont 0 = Send out of-band DTMF message on starting point of DTMF digit.

1 = Send DTMF message on ending point of DTMF digit (default).

DTMFDigitLength Time in msec for generating DTMF tones to the PSTN side (if received in INFO).

The default value is 100 msec.

DTMFInterDigitInterval Time in msec between generated DTMFs to PSTN side (if received in INFO).

The default value is = 100 msec.

DTMFVolume

[DTMF Volume]

DTMFTransportType

[DTMF Transport Type]

DTMF level for regenerated digits to PSTN side (-31 to 0, corresponding to -31 dBm to 0 dBm in 1 dB steps, default = -11 dBm).

0 = Erase digits from voice stream, do not relay to remote.

2 = Digits remain in voice stream.

3 = Erase digits from voice stream, relay to remote according to RFC 2833.

Note: This parameter is automatically updated if one of the following parameters is configured: IsDTMFUsed, TxDTMFOption or RxDTMFOption.

3Com VCX V7122 SIP VoIP Gateway User Manual 149

Configuring the Gateway’s Alternative Routing (based on Connectivity and

QoS)

The Alternative Routing feature enables reliable routing of Tel to IP calls when Proxy isn’t used. The VCX V7122 gateway periodically checks the availability of connectivity and suitable Quality of Service (QoS) before routing. If the expected quality cannot be achieved, an alternative IP route for the prefix (phone number) is selected.

Note that if the alternative routing destination is the gateway itself, the call can be configured to be routed back to one of the gateway’s trunk groups and thus back into the PSTN (PSTN

Fallback).

Alternative Routing Mechanism

When a Tel IP call is routed through the VCX V7122 gateway, the call’s destination number is compared to the list of prefixes defined in the Tel to IP Routing table (described in Tel to IP

Routing Table on page 52 ). The Tel to IP Routing table is scanned for the destination number’s prefix starting at the top of the table. When an appropriate entry (destination number matches one of the prefixes) is found, the prefix’s corresponding destination IP address is checked. If the destination IP address is disallowed, an alternative route is searched for in the following table entries.

Destination IP address is disallowed if no ping to the destination is available (ping is continuously initiated every 7 seconds), when an inappropriate level of QoS (delay or packet loss, calculated according to previous calls) was detected, or when DNS host name is not resolved.

The VCX V7122 gateway matches the rules starting at the top of the table. For this reason, enter the main IP route above any alternative route.

Determining the Availability of Destination IP Addresses

To determine the availability of each destination IP address (or host name) in the routing table, one (or all) of the following (configurable) methods are applied:

Connectivity – The destination IP address is queried periodically (currently only by ping).

QoS – The QoS of an IP connection is determined according to RTCP (Real-Time

Control Protocol) statistics of previous calls. Network delay (in msec) and network packet loss (in percentage) are separately quantified and compared to a certain (configurable) threshold. If the calculated amounts (of delay or packet loss) exceed these thresholds the IP connection is disallowed.

DNS resolution – When host name is used (instead of IP address) for the destination route, it is resolved to an IP address by a DNS server. Connectivity and QoS are then applied to the resolved IP address.

PSTN Fallback as a Special Case of Alternative Routing

The purpose of the PSTN Fallback feature is to enable the VCX V7122 gateway to redirect

PSTN originated calls back to the legacy PSTN network if a destination IP route is found unsuitable (disallowed) for voice traffic at a specific time.

150 3Com VCX V7122 SIP VoIP Gateway User Manual

To enable PSTN fallback, assign the IP address of the gateway itself as an alternative route to the desired prefixes. Note that calls (now referred to as IP to Tel calls) can be re-routed to a specific trunk group using the Routing parameters.

Relevant Parameters

The following parameters (described in Table 28 on page 119 ) are used to configure the

Alternative Routing mechanism:

AltRoutingTel2IPEnable

AltRoutingTel2IPMode

IPConnQoSMaxAllowedPL

IPConnQoSMaxAllowedDelay

Working with Supplementary Services

The VCX V7122 SIP gateway supports the following supplementary services:

Hold / Retrieve

Transfer

Call Forward (doesn't initiate call forward, only responds to call forward request)

VCX V7122 SIP users are only required to enable the Hold and Transfer features. The call forward (supporting 30x redirecting responses) and call waiting (receive of 182 response) features are enabled by default. Note that all call participants must support the specific used method.

Call Hold and Retrieve Features

The party that initiates the hold is called the holding party, the other party is called the held party. The VCX V7122 gateway can't initiate the hold, but it can respond to hold request, and as such it is a held party.

After a successful hold, the held party should hear HELD_TONE, defined in gateway's

Call Progress Tones file.

Retrieve can be performed only by the holding party while the call is held and active.

After a successful retrieve the voice should be connected again.

The hold and retrieve functionalities are implemented by Reinvite messages. The IP address 0.0.0.0 as the connection IP address or the string “a=inactive” in the received

Reinvite SDP cause the gateway to enter Hold state and to play held tone (configured in the gateway) to the PBX/PSTN. If the string “a=recvonly” is received in the SDP message, the gateway stops sending RTP packets, but continues to listen to the incoming RTP packets. Usually, the remote party plays, in this scenario, Music on-hold

(MOH) and the VCX V7122 forwards the MOH to the held party.

Call Transfer

There are two types of call transfers:

3Com VCX V7122 SIP VoIP Gateway User Manual 151

The common way to perform a consultation transfer is as follows:

In the transfer scenario there are three parties:

Party A - transferring, Party B – transferred, Party C – transferred to.

A Calls B.

A presses the hookflash and puts B on-hold (party B hears a hold tone).

A dials C.

After A completes dialing C, A can perform the transfer by on-hooking the A phone.

After the transfer is completed, B and C parties are engaged in a call.

The transfer can be initiated at any of the following stages of the call A to C:

Just after completing dialing C phone number - Transfer from setup.

While hearing ring back – Transfer from alert.

While speaking to C – Transfer from active.

Blind transfer is performed after we have a call between A and B, and A party decides to transfer the call to C immediately without speaking with C.

The result of the transfer is a call between B and C (just like consultation transfer only skipping the consultation stage).

The VCX V7122 gateway doesn't initiate call transfer. It only can respond to call transfer request.

TDM Tunneling

The VCX V7122 TDM Tunneling feature allows you to tunnel groups of digital trunk spans or timeslots (B-channels) over the IP network. TDM Tunneling utilizes the internal routing capabilities of the VCX V7122 (working without Proxy control) to receive voice and data streams from TDM (1 to 16 E1/T1/J1) spans or individual timeslots, convert them into packets and transmit them automatically over the IP network (using point-to-point or point-tomultipoint gateway distributions). A VCX V7122 opposite it (or several VCX V7122 gateways, when point-to-multipoint distributions is used) converts the IP packets back into TDM traffic.

Each timeslot can be targeted to any other timeslot within a trunk in the opposite VCX

V7122.

Implementation

When TDM Tunneling is enabled (‘EnableTDMOverIP’ is set to 1 on the originating VCX

V7122), the originating VCX V7122 automatically initiates SIP calls from all enabled Bchannels belonging to the E1/T1/J1 spans that are configured with the ‘Transparent’ protocol. The called number of each call is the internal phone number of the B-channel that the call originates from. The IP to Trunk Group routing table is used to define the destination

IP address of the terminating VCX V7122. The terminating VCX V7122 gateway automatically answers these calls if its E1/T1 protocol is set to ‘Transparent’

(ProtocolType = 5) and parameter ‘ChannelSelectMode = 0’ (By Phone Number).

152 3Com VCX V7122 SIP VoIP Gateway User Manual

Note: It is possible to configure both gateways to also operate in symmetric mode. To do so, set ‘EnableTDMOverIP’ to 1 and configure the Tel to IP Routing tables in both VCX V7122 gateways. In this mode, each gateway (after it is reset) initiates calls to the second gateway.

The first call for each B-channel is answered by the second gateway.

The VCX V7122 monitors the established connections continuously. If for some reason one or more calls are released, the gateway automatically reestablishes these “broken“ connections. In addition, when a failure in a physical trunk or in the IP network occurs, the

VCX V7122 gateways reestablish the tunneling connections as soon as the network restores.

Note: It is recommended to use the keep-alive mechanism for each connection by activating

“session expires” timeout, and using Reinvite messages.

By utilizing the ‘Profiles’ mechanism (see Configuring the Profile Definitions on page 58 ) you can configure the TDM Tunneling feature to choose different settings, based on a timeslot or groups of timeslots. For example, you can use low-bit-rate vocoders to transport voice, and

‘Transparent’ coder to transport data (e.g., for D-channel). You can also use Profiles to assign ToS (for DiffServ) per source, a time-slot carrying data or signaling gets a higher priority value than a time-slot carrying voice.

For tunneling of E1/T1 CAS trunks enable RFC 2833 CAS relay mode

(CASTransportType = 1).

Figure 58 and Figure 59 show an example of ini files for two VCX V7122 gateways implementing TDM Tunneling for four E1 spans. Note that in this example both gateways are dedicated to TDM tunneling.

Figure 58

ini File Example for TDM Tunneling (Originating Side)

EnableTDMOverIP = 1

;E1_TRANSPARENT_31

ProtocolType_0 = 5

ProtocolType_1 = 5

ProtocolType_2 = 5

ProtocolType_3 = 5 prefix = '*,10.8.24.12' ;(IP address of the VCX V7122 in the opposite location)

; Channel selection by Phone number

ChannelSelectMode = 0

;Profiles can be used do define different coders per B-channels, such as Transparent

; coder for B-channels (time slot 16) that carries PRI signaling.

TrunkGroup = 0/1-31,1000,1

TrunkGroup = 1/1-31,2000,1

TrunkGroup = 2/1-31,3000,1

TrunkGroup = 3/1-31,4000,1

TrunkGroup = 0/16-16,7000,2

TrunkGroup = 1/16-16,7001,2

TrunkGroup = 2/16-16,7002,2

TrunkGroup = 3/16-16,7003,2

CoderName = 'g7231'

CoderName = 'Transparent'

3Com VCX V7122 SIP VoIP Gateway User Manual 153

CoderName_1 = 'g7231'

CoderName_2 = 'Transparent'

TelProfile_1 = voice,$$,1,$$,$$,$$,$$,$$,$$,$$

TelProfile_2 = data,$$,2,$$,$$,$$,$$,$$,$$,$$

Figure 59

ini File Example for TDM Tunneling (Terminating Side)

;E1_TRANSPARENT_31

ProtocolType_0 = 5

ProtocolType_1 = 5

ProtocolType_2 = 5

ProtocolType_3 = 5

; Channel selection by Phone number

ChannelSelectMode = 0

TrunkGroup = 0/1-31,1000,1

TrunkGroup = 1/1-31,2000,1

TrunkGroup = 2/1-31,3000,1

TrunkGroup = 3/1-31,4000,1

TrunkGroup = 0/16-16,7000,2

TrunkGroup = 1/16-16,7001,2

TrunkGroup = 2/16-16,7002,2

TrunkGroup = 3/16-16,7003,2

CoderName = 'g7231'

CoderName = 'Transparent'

CoderName_1 = 'g7231'

CoderName_2 = 'Transparent'

TelProfile_1 = voice,$$,1,$$,$$,$$,$$,$$,$$,$$

TelProfile_2 = data,$$,2,$$,$$,$$,$$,$$,$$,$$

Call Detail Report

The Call Detail Report (CDR) contains vital statistic information on calls made by the gateway. CDRs are generated at the end and (optionally) at the beginning of each call

(determined by the parameter ‘CDRReportLevel’). The destination IP address for CDR logs is determined by the parameter ‘CDRSyslogServerIP’.

The following CDR fields are supported:

Table 35

Supported CDR Fields

Field Name

Cid

Trunk

Description

Board’s Logic Channel Number

Call Identifier

Physical Trunk Number

154 3Com VCX V7122 SIP VoIP Gateway User Manual

Field Name

ConId

TG

Orig

SourceIp

DestIp

TON

NPI

SrcPhoneNum

TON

NPI

DstPhoneNum

DstNumBeforeMap

Description

H.323/SIP Conference ID

Trunk Group Number

Call Originator (IP, Tel)

Source IP Address

Destination IP Address

Source Phone Number Type

Source Phone Number Plan

Source Phone Number

Destination Phone Number Type

Destination Phone Number Plan

Destination Phone Number

Destination Number Before Manipulation

Port

TrmSd

Fax

OutPackets

PackLoss

UniqueId

SetupTime

ConnectTime

ReleaseTime

3Com VCX V7122 SIP VoIP Gateway User Manual

Remote RTP Port

Initiator of Call Release (IP, Tel, Unknown)

Fax Transaction during the Call

Number of Outgoing Packets

Number of Outgoing Lost Packets unique RTP ID

Call Setup Time

Call Connect Time

Call Release Time

155

Field Name Description

RTPssrc

RemoteRTPssrc

Local RTP SSRC

Remote RTP SSRC

TON

NPI

RedirectPhonNum

Redirection Phone Number Type

Redirection Phone Number Plan

Redirection Phone Number

Trunk to Trunk Routing Example

This example describes two VCX V7122 gateways, each interface with the PSTN through four E1 spans. Gateway "A" is configured to route all incoming Tel IP calls to gateway "B".

Gateway "B" generates calls to PSTN on the same E1 Trunk as the call was originally received (in gateway "A").

Gateway "A" IP address is 192.168.3.50.

Gateway "B" IP address is 192.168.3.51.

Ini File Parameters of Gateways "A" and "B”:

Define, for both gateways, four trunk groups; each with 30 B-channels:

TrunkGroup_1 = 0/1-31,1000

TrunkGroup_2 = 1/1-31,2000

TrunkGroup_3 = 2/1-31,3000

TrunkGroup_4 = 3/1-31,4000

In gateway "A”, add the originating Trunk Group ID, as a prefix, to the destination number, for Tel IP calls:

AddTrunkGroupAsPrefix=1

In gateway "A", route all incoming PSTN calls, starting with the prefixes 1, 2, 3 and 4, to gateway’s "B" IP address:

Prefix = 1, 192.168.3.51

Prefix = 2, 192.168.3.51

Prefix = 3, 192.168.3.51

Prefix = 4, 192.168.3.51

Note: It is also possible to define "Prefix = *,192.168.3.51" instead of the four lines above.

In gateway "B", route IP PSTN calls to Trunk Group ID according to the first digit of the called number:

156 3Com VCX V7122 SIP VoIP Gateway User Manual

PSTNPrefix = 1,1

PSTNPrefix = 2,2

PSTNPrefix = 3,4

PSTNPrefix = 4,4

In gateway "B", remove the first digit from each IP PSTN number, before it is used in an outgoing call:

NumberMapIP2Tel = *,1

SIP Call Flow Example

The Call Flow, shown in Figure 60 , describes SIP messages exchanged between VCX

V7122 gateway and an VCX V7111 gateway during a simple call.

VCX V7111 with phone number “8000”, calls VCX V7122 with phone number “1000”:

Figure 60

SIP Call Flow Example

Mediant 2000

10.8.201.10

MP-108

10.8.201.108

INVITE F1

Trying F2

Ringing F3

200 OK F4

Ack F5

BYE

200 OK

F6

F7

F1 10.8.201.108 ==> 10.8.201.10 INVITE

INVITE sip:[email protected];user=phone SIP/2.0

Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd

From: <sip:[email protected]>;tag=1c5354

To: <sip:[email protected]>

Call-ID: [email protected]

CSeq: 18153 INVITE

Contact: <sip:[email protected];user=phone>

3Com VCX V7122 SIP VoIP Gateway User Manual 157

User-Agent: Audiocodes-Sip-Gateway/MP-108 FXS/v.4.20.299.410

Supported: 100rel,em

Accept-Language: en

Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO

Content-Type: application/sdp

Content-Length: 208 v=0 o=AudiocodesGW 18132 74003 IN IP4 10.8.201.108 s=Phone-Call c=IN IP4 10.8.201.108 t=0 0 m=audio 4000 RTP/AVP 8 96 a=rtpmap:8 pcma/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20

F2 10.8.201.10 ==> 10.8.201.108 Trying

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd

From: <sip:[email protected]>;tag=1c5354

To: <sip:[email protected]>

Call-ID: [email protected]

Server: Audiocodes-Sip-Gateway/TrunkPack 1610/v.4.20.299.412

CSeq: 18153 INVITE

Content-Length: 0

F3 10.8.201.10 ==> 10.8.201.108 180 Ringing

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd

From: <sip:[email protected]>;tag=1c5354

To: <sip:[email protected]>;tag=1c7345

Call-ID: [email protected]

Server: Audiocodes-Sip-Gateway/TrunkPack 1610/v.4.20.299.412

CSeq: 18153 INVITE

Supported: 100rel,em

Content-Length: 0

Phone "1000" answers the call, and sends "200 OK" message to MP gateway

10.8.201.108.

F4 10.8.201.10 ==> 10.8.201.108 200 OK

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd

From: <sip:[email protected]>;tag=1c5354

To: <sip:[email protected]>;tag=1c7345

Call-ID: [email protected]

CSeq: 18153 INVITE

Contact: <sip:[email protected];user=phone>

Server: Audiocodes-Sip-Gateway/TrunkPack 1610/v.4.20.299.412

Supported: 100rel,em

Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO

158 3Com VCX V7122 SIP VoIP Gateway User Manual

Content-Type: application/sdp

Content-Length: 206 v=0 o=AudiocodesGW 30221 87035 IN IP4 10.8.201.10 s=Phone-Call c=IN IP4 10.8.201.10 t=0 0 m=audio 7210 RTP/AVP 8 96 a=rtpmap:8 pcma/8000 a=ptime:20 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15

F5 10.8.201.108 ==> 10.8.201.10 ACK

ACK sip:[email protected];user=phone SIP/2.0

Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacZYpJWxZ

From: <sip:[email protected]>;tag=1c5354

To: <sip:[email protected]>;tag=1c7345

Call-ID: [email protected]

User-Agent: Audiocodes-Sip-Gateway/MP-108 FXS/v.4.20.299.410

CSeq: 18153 ACK

Supported: 100rel,em

Content-Length: 0

Phone "8000" goes on-hook; gateway 10.8.201.108 sends "BYE" to gateway

10.8.201.10. Voice path is established.

F6 10.8.201.108 ==> 10.8.201.10 BYE

BYE sip:[email protected]0;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacRKCVBud

From: <sip:[email protected]>;tag=1c5354

To: <sip:[email protected]>;tag=1c7345

Call-ID: [email protected]

User-Agent: Audiocodes-Sip-Gateway/MP-108 FXS/v.4.20.299.410

CSeq: 18154 BYE

Supported: 100rel,em

Content-Length: 0F7 10.2.37.10 ==> 10.2.37.20 200 OK

F7 10.8.201.10 ==> 10.8.201.108 200 OK

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacRKCVBud

From: <sip:[email protected]>;tag=1c5354

To: <sip:[email protected]>;tag=1c7345

Call-ID: [email protected]

Server: Audiocodes-Sip-Gateway/TrunkPack 1610/v.4.20.299.412

CSeq: 18154 BYE

Supported: 100rel,em

Content-Length: 0

3Com VCX V7122 SIP VoIP Gateway User Manual 159

SIP Authentication Example

VCX V7122 gateway supports basic and digest authentication types, according to SIP RFC

3261 standard. A proxy server might require authentication before forwarding an INVITE message. A Registrar/Proxy server may also require authentication for client registration. A proxy replies to an unauthenticated INVITE with a 407 Proxy Authorization Required response, containing a Proxy-Authenticate header with the form of the challenge. After sending an ACK for the 407, the User Agent can then resend the INVITE with a Proxy-

Authorization header containing the credentials.

User Agent, Redirect or Registrar servers typically use 401 Unauthorized responses to challenge authentication containing a WWW-Authenticate header, and expect the re-INVITE to contain an Authorization header.

The following example describes the Digest Authentication procedure including computation of User Agent credentials.

The REGISTER request is sent to Registrar/Proxy server for registration, as follows:

REGISTER sip:10.2.2.222 SIP/2.0

Via: SIP/2.0/UDP 10.1.1.200

From: <sip: [email protected]>;tag=1c17940

To: <sip: [email protected]>

Call-ID: [email protected]

User-Agent: Audiocodes-Sip-Gateway/TrunkPack 1610/v.4.20.299.412

CSeq: 1 REGISTER

Contact: sip:[email protected]:

Expires:3600

On receiving this request the Registrar/Proxy returns 401 Unauthorized response.

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 10.2.1.200

From: <sip:[email protected] >;tag=1c17940

To: <sip:[email protected] >

Call-ID: [email protected]

Cseq: 1 REGISTER

Date: Mon, 30 Jul 2001 15:33:54 GMT

Server: Columbia-SIP-Server/1.17

Content-Length: 0

WWW-Authenticate: Digest realm="audiocodes.com", nonce="11432d6bce58ddf02e3b5e1c77c010d2", stale=FALSE, algorithm=MD5

According to the sub-header present in the WWW-Authenticate header the correct

REGISTER request is formed.

Since the algorithm used is MD5, take:

The username from the ini file: VCX V7122 -3Com

The realm return by the proxy: 3Com.com

The password from the ini file: 3Com.

The equation to be evaluated: (according to RFC this part is called A1).

160 3Com VCX V7122 SIP VoIP Gateway User Manual

“M2K-3Com:3Com.com:3Com”.

The MD5 algorithm is run on this equation and stored for future usage.

The result is: “a8f17d4b41ab8dab6c95d3c14e34a9e1”

Next we need to evaluate the par called A2. We take:

The method type “REGISTER”

Using SIP protocol “sip”

Proxy IP from ini file “10.2.2.222”

The equation to be evaluated:

“REGISTER:sip:10.2.2.222”.

The MD5 algorithm is run on this equation and stored for future usage.

The result is:”a9a031cfddcb10d91c8e7b4926086f7e”

The final stage:

The A1 result

The nonce from the proxy response: “11432d6bce58ddf02e3b5e1c77c010d2”

The A2 result

The equation to be evaluated:

“A1:11432d6bce58ddf02e3b5e1c77c010d2:A2”.

The MD5 algorithm is run on this equation. The outcome of the calculation is the response needed by the gateway to be able top register with the Proxy.

The response is: “b9c45d0234a5abf5ddf5c704029b38cf”

3Com VCX V7122 SIP VoIP Gateway User Manual 161

At this time a new REGISTER request is issued with the response:

REGISTER sip:10.2.2.222 SIP/2.0

Via: SIP/2.0/UDP 10.1.1.200

From: <sip: [email protected]>;tag=1c23940

To: <sip: [email protected]>

Call-ID: [email protected]

Server: Audiocodes-Sip-Gateway/TrunkPack 1610/v.4.20.299.412

CSeq: 1 REGISTER

Contact: sip:[email protected]:

Expires:3600

Authorization: Digest, Username: MP108-AudioCodes, realm="audiocodes.com”, nonce="11432d6bce58ddf02e3b5e1c77c010d2", uri=”10.2.2.222”, response=“ b9c45d0234a5abf5ddf5c704029b38cf”

On receiving this request, if accepted by the Proxy, the proxy returns a 200 OK response closing the REGISTER transaction.

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.1.1.200

From: <sip: [email protected]>;tag=1c23940

To: <sip: [email protected]>

Call-ID: [email protected]

Cseq: 1 REGISTER

Date: Thu, 26 Jul 2001 09:34:42 GMT

Server: Columbia-SIP-Server/1.17

Content-Length: 0

Contact: <sip:[email protected]>; expires="Thu, 26 Jul 2001 10:34:42 GMT"; action=proxy; q=1.00

Contact: <[email protected]:>; expires="Tue, 19 Jan 2038 03:14:07 GMT"; action=proxy; q=0.00

Expires: Thu, 26 Jul 2001 10:34:42 GMT

162 3Com VCX V7122 SIP VoIP Gateway User Manual

C

HAPTER

9: D

IAGNOSTICS

Several diagnostic tools are provided, enabling you to identify correct functioning of the VCX

V7122, or an error condition with a probable cause and a solution or workaround.

Front panel indicator LEDs on the VCX V7122. The location and functionality of the front panel LEDs is shown in TP-1610 Front Panel LED Indicators on page 22 .

VCX V7122 Self-Testing on hardware initialization (see VCX V7122 Self-Testing below).

Syslog Syslog Support below).

VCX V7122 Self-Testing

The VCX V7122 features two self-testing modes: rapid and detailed.

Rapid self-test mode is invoked each time the media gateway completes the initialization process. In this short test phase the only error detected and reported is failure in initializing hardware components. All Status and Error reports in this self-test phase are reported through Network Interface ports, as well as indicated by the LED Status Indicators.

Detailed self-test mode is invoked when initialization of the media gateway is completed and if the configuration parameter EnableDiagnostics is set to 1 (this parameter can be configured through the ini file mechanism). In this mode, the media gateway tests all the hardware components (memory, DSP, etc.), outputs the status of the test results, and ends the test. To continue operational running, reset the media gateway again but this time configure the EnableDiagnostics parameter to 0.

Syslog Support

Syslog protocol is an event notification protocol that enables a machine to send event notification messages across IP networks to event message collectors, also known as

Syslog servers. Syslog protocol is defined in the IETF RFC 3164 standard.

Since each process, application and operating system was written independently, there is little uniformity to Syslog messages. For this reason, no assumption is made on the contents of the messages other than the minimum requirements of its priority.

Syslog uses UDP as its underlying transport layer mechanism. The UDP port that was assigned to Syslog is 514.

The Syslog message is transmitted as an ASCII (American Standard Code for Information

Interchange) message. The message starts with a leading "<" (less-than character), followed by a number, which is followed by a ">" (greater-than character). This is optionally followed by a single ASCII space.

The number described above is known as the Priority and represents both the Facility and

Severity as described below. The Priority number consists of one, two, or three decimal integers.

3Com VCX V7122 SIP VoIP Gateway User Manual 163

For example:

<37>

Note that when NTP is enabled, a timestamp string [hour:minutes:seconds] is added to all

Syslog messages (for information on NTP see Simple Network Time Protocol Support on page 67 ).

Syslog Servers

Users can use the provided Syslog server (ACSyslog08.exe) or other third-party Syslog servers.

Examples of Syslog servers available as shareware on the Internet:

Enterprises:

The US CMS Server: http://uscms.fnal.gov/hanlon/uscms_server/

Software:

Netal SL4NT 2.1 Syslog Daemon: http://www.netal.com

A typical Syslog server application enables filtering of the messages according to priority, IP sender address, time, date, etc.

Operation

Sending the Syslog Messages

The Syslog client, embedded in the firmware of the VCX V7122, sends error reports/events generated by the VCX V7122 unit application to a Syslog server, using IP/UDP protocol.

Setting the Syslog Server

To set the Syslog server:

Use the VCX V7122 Embedded Web Server (Advanced Configuration>Network

Settings>screen section Syslog Settings) to enable the Syslog Server (Enable Syslog) and to enter its IP address (Syslog Server IP address); see Figure 61 below.

Figure 61

Setting the Syslog Server IP Address

Alternately, use the Embedded Web Server or the BootP/TFTP utility to load the ini configuration file containing both the IP address and the enabling parameters:

SyslogServerIP and EnableSyslog respectively. For detailed information on the

BootP/TFTP utility, see Appendix B: The BootP/TFTP Configuration Utility on page 193 .

For an ini file example showing these parameters, see The ini File Example for Syslog and to Figure 62 .

164 3Com VCX V7122 SIP VoIP Gateway User Manual

The ini File Example for Syslog

Figure 62 shows an ini file section with an example configuration for the address parameter

SyslogServerIP and an example configuration for the client activation parameter

EnableSyslog.

Figure 62

The ini File Example for Syslog

[Syslog]

SyslogServerIP = 10.2.0.136

EnableSyslog = 1

GWDebugLevel = 5

3Com VCX V7122 SIP VoIP Gateway User Manual 165

166 3Com VCX V7122 SIP VoIP Gateway User Manual

C

HAPTER

10: B

OOT

P/DHCP S

UPPORT

The startup process (illustrated in Figure 63 on page 168 ) begins when the gateway is reset

(physically or from the Web / SNMP) and ends when the operational software is running. In the startup process, the network parameters, software and configuration files are obtained.

Startup Process

After the gateway powers up or after it is physically reset, it broadcasts a BootRequest message to the network. If it receives a reply (from a BootP server), it changes its network parameters (IP address, subnet mask and default gateway address) to the values provided.

If there is no reply from a BootP server and if DHCP is enabled (DHCPEnable = 1), the gateway initiates a standard DHCP procedure to configure its network parameters.

After changing the network parameters, the gateway attempts to load the cmp and various configuration files from the TFTP server’s IP address, received from the BootP/DHCP servers. If a TFTP server’s IP address isn’t received, the gateway attempts to load the software (cmp) file and / or configuration files from a preconfigured TFTP server (see the parameters ‘IniFileURL’ and ‘CmpFileURL’ described in Table 24 on page 94 ). Thus, the gateway can obtain its network parameters from BootP or DHCP servers and its software and configuration files from a different TFTP server (preconfigured in ini file).

If BootP/DHCP servers are not found or when the gateway is reset from the Web / SNMP, it retains its network parameters and attempts to load the software (cmp) file and / or configuration files from a preconfigured TFTP server.

If a preconfigured TFTP server doesn’t exist, the gateway operates using the existing software and configuration files loaded on its non-volatile memory.

Note that after the operational software runs, if DHCP is configured, the gateway attempts to renew its lease with the DHCP server.

Though DHCP and BootP servers are very similar in operation, the DHCP server includes some differences that could prevent its operation with BootP clients.

However, many DHCP servers, such as Windows

NT DHCP server, are backward-compatible with BootP protocol and can be used for gateway configuration.

The time duration between BootP/DHCP requests is set to 1 second by default.

This can be changed by the BootPDelay ini file parameter. Also, the number of

requests is 3 by default and can be changed by BootPRetries ini file parameter.

(Both parameters can also be set using the BootP command line switches).

3Com VCX V7122 SIP VoIP Gateway User Manual 167

Figure 63

VCX V7122 Startup Process

Physical Reset

Reset from the Web

Interface or SNMP

BootP x times

No

Response

DHCP x times

BootP Response

Update network parameters from

BootP/DHCP reply

DHCP Response

BootP/DHCP reply contains firmware file name?

Yes

Download firmware via

TFTP

No

No

Response

168

No

BootP/DHCP reply contains ini file name?

Yes

Yes

Yes

BootP/DHCP reply contains ini file name?

No

Preconfigured firmware

URL?

No

Yes

Device reset

Download firmware via

TFTP

Preconfigured ini file

URL?

Download configuration files via TFTP

No

Run operational software

3Com VCX V7122 SIP VoIP Gateway User Manual

DHCP Support

When the gateway is configured to use DHCP (DHCPEnable = 1), it attempts to contact the enterprise’s DHCP server to obtain the networking parameters (IP address, subnet mask, default gateway, primary/secondary DNS server and SIP server address). These network parameters have a "time limit". After the time limit expires, the gateway must "renew" its lease from the DHCP server.

Note that if the DHCP server denies the use of the gateway's current IP address and specifies a different IP address (according to RFC 1541), the gateway must change its networking parameters. If this happens while calls are in progress, they are not automatically rerouted to the new network address (since this function is beyond the scope of a VoIP gateway). Therefore, administrators are advised to configure DHCP servers to allow renewal of IP addresses.

Note: If the gateway's network cable is disconnected and reconnected, a DHCP renewal is performed (to verify that the gateway is still connected to the same network).

When DHCP is enabled, the gateway also includes its product name (e.g., ‘VCX V7122’) in the DHCP ‘option 60’ Vendor Class Identifier. The DHCP server can use this product name to assign an IP address accordingly.

Note: After power-up, the gateway issues two DHCP requests. Only in the second request, the DHCP ‘option 60’ is contained. If the gateway is reset from the Web/SNMP, only a single

DHCP request containing ‘option 60’ is sent.

If DHCP procedure is used, the new gateway IP address, allocated by the DHCP server, must be detected.

If, during operation, the IP address of the gateway is changed as a result of a

DHCP renewal, the gateway is automatically reset.

To detect the gateway’s IP address, follow one of the procedures below:

Starting with Bootload software version 1.92, the gateway can use host name in the

DHCP request. The host name is set to acl_nnnnn, where nnnnn stands for the gateway’s serial number (the serial number is equal to the last 6 digits of the MAC address converted from Hex to decimal). If the DHCP Server registers this host name to a DNS server, the user can access the gateway (through a Web browser) using a URL of http://acl_<serial number> (instead of using the gateway’s IP address). For example, if the gateway’s MAC address is 00908f010280, the DNS name is acl_66176.

After physically resetting the gateway its IP address is displayed in the ‘Client Info’ column in the BootP/TFTP configuration utility (see Figure 65 on page 195 ).

Contact your System Administrator.

BootP Support

Upgrading the VCX V7122

When upgrading the VCX V7122 (loading new software onto the gateway) using the

BootP/TFTP configuration utility:

3Com VCX V7122 SIP VoIP Gateway User Manual 169

From version 4.2 to version 4.4, the device loses its configuration. Therefore, to retain the previous gateway configuration you must save the ini file before you replace the cmp file, and reload it to the device. For information on backing up and restoring the gateway’s configuration see Restoring and Backing Up the Gateway Configuration on page 72 .

From version 4.4 to version 4.4 or to any higher version, the device retains its configuration (ini file), however, the auxiliary files (CPT, logo, etc.) may be erased.

When using the Software Upgrade wizard, available through the Web Interface (see

Software Upgrade Wizard on page 81 ), the auxiliary files are saved as well.

Note: To save the cmp file to non-volatile memory, use the -fb command line switches. If the file is not saved, the gateway reverts to the old version of software after the next reset. For information on using command line switches, see Using Command Line Switches on page 202 .

Vendor Specific Information Field

The VCX V7122 uses the vendor specific information field in the BootP request to provide device-related initial startup information. The BootP/TFTP configuration utility displays this information in the ‘Client Info’ column (see Figure 65 on page 195 ).

Note: This option is not available on DHCP servers.

The Vendor Specific Information field is disabled by default. To enable / disable this feature: set the ini file parameter ‘ExtBootPReqEnable’ (see Table 24 on page 94 ) or use the ‘-be’ command line switch (see Table 40 on page 203 ).

Table 36 details the vendor specific information field according to device types:

Table 36

Vendor Specific Information Field

Tag #

221

222

225

Description

Current IP Address

Burned Boot Software Version

Chassis Geographical Address

Value

#02 = IPM-1610/TP-1610

#03 = IPMTP260

#05 = IPMTP260

XXX.XXX.XXX.XXX

X.XX

XXXXXXXXXXXX

0 – 31

(TP-260 Only)

0 – 31

(TP-260 Only)

N/A

Length

1

4

4

12

1

1

229 E&M 1

Table 37 exemplifies the structure of the vendor specific information field for a TP-1610 slave module with IP Address 10.2.70.1.

170 3Com VCX V7122 SIP VoIP Gateway User Manual

Table 37

Structure of the Vendor Specific Information Field

Vendor-

Specific

Informati on Code

42 12 220 1 2 225 1 1 221 4 10 2 70 1 255

3Com VCX V7122 SIP VoIP Gateway User Manual 171

172 3Com VCX V7122 SIP VoIP Gateway User Manual

C

HAPTER

11: SNMP-B

ASED

M

ANAGEMENT

Simple Network Management Protocol (SNMP) is a standard-based network control protocol used to manage elements in a network. The SNMP Manager (usually implemented by a

Network Manager (NM) or an Element Manager (EM)) connects to an SNMP Agent

(embedded on a remote Network Element (NE)) to perform network element Operation,

Administration and Maintenance (OAM).

Both the SNMP Manager and the NE refer to the same database to retrieve information or configure parameters. This database is referred to as the Management Information Base

(MIB), and is a set of statistical and control values. Apart from the standard MIBs documented in IETF RFCs, SNMP additionally enables the use of private MIBs, containing a non-standard information set (specific functionality provided by the NE).

Directives, issued by the SNMP Manager to an SNMP Agent, consist of the identifiers of

SNMP variables (referred to as MIB object identifiers or MIB variables) along with instructions to either get the value for that identifier, or set the identifier to a new value

(configuration). The SNMP Agent can also send unsolicited events towards the EM, called

SNMP traps.

The definitions of MIB variables supported by a particular agent are incorporated in descriptor files, written in Abstract Syntax Notation (ASN.1) format, made available to EM client programs so that they can become aware of MIB variables and their use.

The device contains an embedded SNMP Agent supporting both general network MIBs

(such as the IP MIB), VoP-specific MIBs (such as RTP) and our proprietary MIBs (acBoard, acGateway, acAlarm and other MIBs), enabling a deeper probe into the inter-working of the device. All supported MIB files are supplied to customers as part of the release.

About SNMP

SNMP Message Standard

Four types of SNMP messages are defined:

Get - A request that returns the value of a named object.

Get-Next - A request that returns the next name (and value) of the ‘next’ object supported by a network device given a valid SNMP name.

Set - A request that sets a named object to a specific value.

Trap - A message generated asynchronously by network devices. It is an unsolicited message from an agent to the manager.

Each of these message types fulfills a particular requirement of Network Managers:

Get Request - Specific values can be fetched via the ‘get’ request to determine the performance and state of the device. Typically, many different values and parameters

3Com VCX V7122 SIP VoIP Gateway User Manual 173

can be determined via SNMP without the overhead associated with logging into the device, or establishing a Transmission Control Protocol (TCP) connection with the device.

Get Next Request - Enables the SNMP standard network managers to ‘walk’ through all

SNMP values of a device (via the ‘get-next’ request) to determine all names and values that an operant device supports. This is accomplished by beginning with the first SNMP object to be fetched, fetching the next name with a ‘get-next’, and repeating this operation.

Set Request - The SNMP standard provides a method of effecting an action associated with a device (via the ‘set’ request) to accomplish activities such as disabling interfaces, disconnecting users, clearing registers, etc. This provides a way of configuring and controlling network devices via SNMP.

Trap Message - The SNMP standard furnishes a mechanism by which devices can

‘reach out’ to a Network Manager on their own (via a ‘trap’ message) to notify or alert the manager of a problem with the device. This typically requires each device on the network to be configured to issue SNMP traps to one or more network devices that are awaiting these traps.

The above message types are all encoded into messages referred to as Protocol Data Units

(PDUs) that are interchanged between SNMP devices.

SNMP MIB Objects

The SNMP MIB is arranged in a tree-structured fashion, similar in many ways to a disk directory structure of files. The top level SNMP branch begins with the ISO ‘internet’ directory, which contains four main branches:

The ‘mgmt’ SNMP branch - Contains the standard SNMP objects usually supported (at least in part) by all network devices.

The ‘private’ SNMP branch - Contains those ‘extended’ SNMP objects defined by network equipment vendors.

The ‘experimental’ and ‘directory’ SNMP branches - Also defined within the ‘internet’ root directory, these branches are usually devoid of any meaningful data or objects.

The ‘tree’ structure described above is an integral part of the SNMP standard, though the most pertinent parts of the tree are the ‘leaf’ objects of the tree that provide actual management data regarding the device. Generally, SNMP leaf objects can be partitioned into two similar but slightly different types that reflect the organization of the tree structure:

Discrete MIB Objects - Contain one precise piece of management data. These objects are often distinguished from ‘Table’ items (below) by adding a ‘.0’ (dot-zero) extension to their names. The operator must merely know the name of the object and no other information.

Table MIB Objects - Contain multiple sections of management data. These objects are distinguished from ‘Discrete’ items (above) by requiring a ‘.’ (dot) extension to their names that uniquely distinguishes the particular value being referenced. The ‘.’ (dot) extension is the ‘instance’ number of an SNMP object. For ‘Discrete’ objects, this instance number is zero. For ‘Table’ objects, this instance number is the index into the

SNMP table. SNMP tables are special types of SNMP objects which allow parallel arrays of information to be supported. Tables are distinguished from scalar objects, so that tables can grow without bounds. For example, SNMP defines the ‘ifDescr’ object (as a standard SNMP object) that indicates the text description of each interface supported by

174 3Com VCX V7122 SIP VoIP Gateway User Manual

a particular device. Since network devices can be configured with more than one interface, this object can only be represented as an array.

By convention, SNMP objects are always grouped in an ‘Entry’ directory, within an object with a ‘Table’ suffix. (The ‘ifDescr’ object described above resides in the ‘ifEntry’ directory contained in the ‘ifTable’ directory).

SNMP Extensibility Feature

One of the principal components of an SNMP manager is a MIB Compiler which allows new

MIB objects to be added to the management system. When a MIB is compiled into an SNMP manager, the manager is made ‘aware’ of new objects that are supported by agents on the network. The concept is similar to adding a new schema to a database.

Typically, when a MIB is compiled into the system, the manager creates new folders or directories that correspond to the objects. These folders or directories can typically be viewed with a MIB Browser, which is a traditional SNMP management tool incorporated into virtually all Network Management Systems.

The act of compiling the MIB allows the manager to know about the special objects supported by the agent and access these objects as part of the standard object set.

Carrier Grade Alarm System

The basic alarm system has been extended to a carrier-grade alarm system. A carrier-grade alarm system provides a reliable alarm reporting mechanism that takes into account EMS outages, network outages, and transport mechanism such as SNMP over UDP.

A carrier-grade alarm system is characterized by the following:

The device has a mechanism that allows a manager to determine which alarms are currently active in the device. That is, the device maintains an active alarm table.

The device has a mechanism to allow a manager to detect lost alarm raise and clear notifications [sequence number in trap, current sequence number MIB object].

The device has a mechanism to allow a manager to recover lost alarm raise and clear notifications [maintains a log history].

The device sends a cold start trap to indicate that it is starting. This allows the EMS to synchronize its view of the device's active alarms.

The SNMP alarm traps are sent as in previous releases. This system provides the mechanism for viewing of history and current active alarm information.

Active Alarm Table

The device maintains an active alarm table to allow a manager to determine which alarms are currently active in the device. Two views of the active alarm table are supported by the agent: alarmActiveTable and alarmActiveVariableTable in the IETF standard ALARM-MIB

(rooted in the AC tree)

The acActiveAlarmTable is a simple, one-row per alarm table that is easy to view with a MIB browser.

3Com VCX V7122 SIP VoIP Gateway User Manual 175

The ALARM-MIB is currently a draft standard and therefore has no OID assigned to it. In the current software release, the MIB is rooted in the experimental MIB subtree. In a future release, after the MIB has been ratified and an OID assigned, it is to move to the official OID.

Alarm History

The device maintains a history of alarms that have been raised and traps that have been cleared to allow a manager to recover any lost, raised or cleared traps. Two views of the alarm history table are supported by the agent: nlmLogTable and nlmLogVariableTable in the standard NOTIFICATION-LOG-MIB

As with the acActiveAlarmTable, the acAlarmHistoryTable is a simple, one-row-per-alarm table that is easy to view with a MIB browser.

Cold Start Trap

VCX V7122 technology supports a cold start trap to indicate that the device is starting. This allows the manager to synchronize its view of the device's active alarms. Two different traps are sent at start-up:

The standard coldStart trap - iso(1).org(3).dod(6).internet(1). snmpV2(6). snmpModules(3). snmpMIB(1). snmpMIBObjects(1). snmpTraps(5). coldStart(1) - sent at system initialization.

The enterprise acBoardEvBoardStarted which is generated at the end of system initialization. This is more of an ‘application-level’ cold start sent after the entire initializing process is complete and all the modules are ready.

Third-Party Performance Monitoring Measurements

Performance measurements are available for a third-party performance monitoring system through an SNMP interface. These measurements can be polled at scheduled intervals by an external poller or utility in a media server or other off-device system.

The device provides two types of performance measurements:

Gauges: Gauges represent the current state of activities on the device. Gauges, unlike counters, can decrease in value, and like counters, can increase. The value of a gauge is the current value or a snapshot of the current activity on the device.

Counters: Counters always increase in value and are cumulative. Counters, unlike gauges, never decrease in value unless the off-device system is reset. the counters are then zeroed.

Performance measurements are provided by three proprietary MIBs (acPerfMediaGateway, acPerfMediaServices and acPerfH323SIPGateway). The first MIB is a generic-type of performance measurements MIB available on all VCX V7122 and related devices. The second is specific to the media server, and the third is for H.323/SIP media gateways.

The generic performance measurements MIB covers:

System packets statistics

176 3Com VCX V7122 SIP VoIP Gateway User Manual

Performance measurement enterprise MIB supports statistics which apply to the

Proxy/Gatekeeper routing tables.

Supported MIBs

The VCX V7122 contains an embedded SNMP Agent supporting the following MIBs:

Standard MIB (MIB-II) - The various SNMP values in the standard MIB are defined in

RFC 1213. The standard MIB includes various objects to measure and monitor IP activity, TCP activity, UDP activity, IP routes, TCP connections, interfaces and general system indicators.

RTP MIB - The RTP MIB is supported in conformance with the IETF’s RFC 2959. It contains objects relevant to the RTP streams generated and terminated by the device and to RTCP information related to these streams.

Trunk MIB - The Trunk MIB contains objects relevant to E1/T1 Trunk interfaces.

NOTIFICATION-LOG-MIB - This standard MIB (RFC 3014 - iso.org.dod.internet.mgmt.mib-2) is supported as part of our implementation of carrier grade alarms.

ALARM-MIB - This is an IETF proposed MIB also supported as part of our implementation of carrier grade alarms. This MIB is still not standard and is therefore under the 3Com.acExperimental branch.

SNMP-TARGET-MIB - This MIB is partially supported (RFC 2273). It allows for the configuration of trap destinations and trusted managers only.

SNMP Research International Enterprise MIBs - VCX V7122 supports two SNMP

Research International MIBs: SR-COMMUNITY-MIB and TGT-ADDRESS-MASK-MIB.

These MIBs are used in the configuration of SNMPv2c community strings and trusted managers.

In addition to the standard MIBs, the complete series contains several proprietary MIBs: acBoard MIB - This proprietary MIB contains objects related to configuration of the device and channels, as well as to run-time information. Through this MIB, users can set up the device configuration parameters, reset the device, monitor the device’s operational robustness and Quality of Service during run-time, and receive traps.

The acBoard MIB is still supported but is being replaced by five newer proprietary

MIBs.

The acBoard MIB has the following groups:

boardConfiguration

boardInformation

channelConfiguration

channelStatus

reset

acTrap

3Com VCX V7122 SIP VoIP Gateway User Manual 177

As noted above, five new MIBs cover the device’s general parameters. Each contains a

Configuration subtree for configuring related parameters. In some, there also are Status and

Action subtrees.

The 5 MIBs are:

AC-ANALOG-MIB

AC-CONTROL-MIB

AC-MEDIA-MIB

AC-PSTN-MIB

AC-SYSTEM-MIB

Other proprietary MIBs are: acGateway MIB - This proprietary MIB contains objects related to configuration of the device when applied as a SIP or H.323 media gateway only. This MIB complements the other proprietary MIBs.

The acGateway MIB has the following groups:

Common - for parameters common to both SIP and H.323

SIP - for SIP parameters only

H.323 - for H.323 parameters only acAtm - This proprietary MIB contains objects related to configuration and status of the device when applied as an ATM media gateway only. This MIB complements the other proprietary MIBs.

The acAtm MIB has the following groups: acAtmConfiguration - for configuring ATM related parameters acAtmStatus - for the status of ATM connections acAlarm - This is 3Com's proprietary carrier-grade alarm MIB. It is a simpler implementation of the notificationLogMIB and the IETF suggested alarmMIB (both also supported in all 3Com devices).

The acAlarm MIB has the following groups:

ActiveAlarm - straightforward (single-indexed) table, listing all currently active alarms, together with their bindings (the alarm bindings are defined in acAlarm. acAlarmVarbinds and also in acBoard.acTrap. acBoardTrapDefinitions. oid_1_3_6_1_4_1_5003_9_10_1_21_2_0). acAlarmHistory - straightforward (single-indexed) table, listing all recently raised alarms together with their bindings (the alarm bindings are defined in acAlarm. acAlarmVarbinds and also in acBoard.acTrap. acBoardTrapDefinitions. oid_1_3_6_1_4_1_5003_9_10_1_21_2_0).

The table size can be altered via notificationLogMIB.notificationLogMIBObjects.nlmConfig.nlmConfigGlobalEntryLimit or notificationLogMIB.notificationLogMIBObjects.nlmConfig.nlmConfigLogTable.nlm

ConfigLogEntry.nlmConfigLogEntryLimit.

The table size can be any value between 50 to 1000 and is 500 by default.

178 3Com VCX V7122 SIP VoIP Gateway User Manual

Traps

Full proprietary trap definitions and trap Varbinds are found in the acBoard MIB and acAlarm

MIB.

The following proprietary traps are supported in the device: acBoardEvResettingBoard - Sent after the device is reset. acBoardEvBoardStarted - Sent after the device is successfully restored and initialized following reset. acBoardTemperatureAlarm - Sent when a board exceeds its temperature limits. acBoardConfigurationError - Sent when a device’s settings are illegal - the trap contains a message stating/detailing/explaining the illegality of the setting. acBoardFatalError - Sent whenever a fatal device error occurs. acFeatureKeyError - Development pending. Intended to relay Feature Key errors, etc. acgwAdminStateChange - Sent when Graceful Shutdown commences and ends. acBoardCallResourcesAlarm - Indicates that no free channels are available. acBoardControllerFailureAlarm - The Gatekeeper/Proxy is not found or registration failed. Internal routing table can be used for routing. acBoardEthernetLinkAlarm - Ethernet link or links are down. acBoardOverloadAlarm - Overload in one or some of the system's components. acActiveAlarmTableOverflow - An active alarm could not be placed in the active alarm table because the table is full.

acAtmPortAlarm

1

- ATM Port Alarm.

acAudioProvisioningAlarm

1

- Raised if the Media Server is unable to provision its audio.

In addition to the listed traps, the device also supports the following standard traps:

dsx1LineStatusChange.

coldStart.

authenticationFailure.

The following are special notes pertaining to MIBs:

A detailed explanation of each parameter can be viewed in an SNMP browser

in the ‘MIB Description’ field.

Not all groups in the MIB are functional. See the version release notes.

Certain parameters are non-functional. Their MIB status is marked 'obsolete'.

When a parameter is set to a new value via SNMP, the change may affect device functionality immediately or may require that the device be soft reset for the change to take effect. This depends on the parameter type.

The current (updated) device configuration parameters are programmed into the

device provided that the user does not load an ini file to the device after reset.

Loading an ini file after reset overrides the updated parameters.

3Com VCX V7122 SIP VoIP Gateway User Manual 179

Additional MIBs are to be supported in future releases.

SNMP Interface Details

This section describes details of the SNMP interface that is required when developing an

Element Manager (EM) for any of the TrunkPack-VoP Series products, or to manage a device with a MIB browser.

Currently, both SNMP and ini file commands and downloads are not encrypted. For ini file encoding, see Secured ini File on page 91 .

SNMP Community Names

By default, the device uses a single, read-only community string of ‘public’ and a single readwrite community string of ‘private’.

Users can configure up to 5 read-only community strings and up to 5 read-write community strings, and a single trap community string is supported:

Configuration of Community Strings via the ini File

SNMPREADONLYCOMMUNITYSTRING_<x> = '#######'

SNMPREADWRITECOMMUNITYSTRING_<x> = '#######' where <x> is a number between 0 and 4, inclusive. Note that the '#' character represents any alphanumeric character. The maximum length of the string is 20 characters.

Configuration of Community Strings via SNMP

To configure read-only and read-write community strings, the EM must use the srCommunityMIB. To configure the trap community string, the EM must also use the snmpVacmMIB and the snmpTargetMIB.

To add a read-only community string (v2user):

Add a new row to the srCommunityTable with CommunityName v2user and GroupName

ReadGroup.

To delete the read-only community string (v2user), follow these steps:

1 If v2user is being used as the trap community string, follow the procedure for changing the trap community string (see below).

2 Delete the srCommunityTable row with CommunityName v2user.

To add a read-write community string (v2admin):

Add a new row to the srCommunityTable with CommunityName of v2admin and

GroupName ReadWriteGroup.

To delete the read-write community string (v2admin), follow these steps:

1 If v2admin is being used as the trap community string, follow the procedure for changing the trap community string. (See below.)

2 Delete the srCommunityTable row with a CommunityName of v2admin and GroupName of ReadWriteGroup.

180 3Com VCX V7122 SIP VoIP Gateway User Manual

To change the only read-write community string from v2admin to v2mgr, follow these steps:

1 Follow the procedure above to add a read-write community string to a row for v2mgr.

2 Set up the EM so that subsequent ‘set’ requests use the new community string, v2mgr.

3 If v2admin is being used as the trap community string, follow the procedure to change the trap community string (see below).

4 Follow the procedure above to delete a read-write community name in the row for v2admin.

To change the trap community string, follow these steps:

(The following procedure assumes that a row already exists in the srCommunityTable for the new trap community string. The trap community string can be part of the TrapGroup,

ReadGroup or ReadWriteGroup. If the trap community string is used solely for sending traps

(recommended), it should be made part of the TrapGroup).

1 Add a row to the vacmSecurityToGroupTable with these values: SecurityModel=2,

SecurityName=the new trap community string, GroupName=TrapGroup, ReadGroup or

ReadWriteGroup. The SecurityModel and SecurityName objects are row indices.

You must add GroupName and RowStatus on the same set.

2 Modify the SecurityName field in the sole row of the snmpTargetParamsTable.

Trusted Managers

By default, the agent accepts ‘get’ and ‘set’ requests from any IP address, as long as the correct community string is used in the request. Security can be enhanced via the use of

Trusted Managers. A Trusted Manager is an IP address from which the SNMP Agent accepts and processes ‘get’ and ‘set’ requests. An EM can be used to configure up to 5

Trusted Managers.

If Trusted Managers are defined, all community strings work from all Trusted

Managers. That is, there is no way to associate a community string with particular trusted managers.

Configuration of Trusted Managers via ini File

To set the Trusted Mangers table from start-up, write the following in the ini file:

SNMPTRUSTEDMGR_X = D.D.D.D where X is any integer between 0 and 4 (0 sets the first table entry, 1 sets the second, and so on), and D is an integer between 0 and 255.

Configuration of Trusted Managers via SNMP

To configure Trusted Managers, the EM must use the srCommunityMIB, the snmpTargetMIB and the TGT-ADDRESS-MASK-MIB.

3Com VCX V7122 SIP VoIP Gateway User Manual 181

To add the first Trusted Manager, follow these steps:

(The following procedure assumes that there is at least one configured read-write community. There are currently no Trusted Managers. The taglist for columns for all srCommunityTable rows are currently empty).

1 Add a row to the snmpTargetAddrTable with these values: Name=mgr0, TagList=MGR,

Params=v2cparams.

2 Add a row to the tgtAddressMaskTable table with these values: Name=mgr0, tgtAddressMask=255.255.255.255:0. The agent does not allow creation of a row in this table unless a corresponding row exists in the snmpTargetAddrTable.

3 Set the value of the TransportLabel field on each non-TrapGroup row in the srCommunityTable to MGR.

To add a subsequent Trusted Manager, follow these steps:

(The following procedure assumes that there is at least one configured read-write community. There are currently one or more Trusted Managers. The taglist for columns for all rows in the srCommunityTable are currently set to MGR. This procedure must be performed from one of the existing Trusted Managers).

1 Add a row to the snmpTargetAddrTable with these values: Name=mgrN, TagList=MGR,

Params=v2cparams, where N is an unused number between 0 and 4.

2 Add a row to the tgtAddressMaskTable table with these values: Name=mgrN, tgtAddressMask=255.255.255.255:0.

An alternative to the above procedure is to set the tgtAddressMask column while you are creating other rows in the table.

To delete a Trusted Manager (not the final one), follow this steps:

(The following procedure assumes that there is at least one configured read-write community. There are currently two or more Trusted Managers. The taglist for columns for all rows in the srCommunityTable are currently set to MGR. This procedure must be performed from one of the existing trusted managers, but not the one that is being deleted.

Remove the appropriate row from the snmpTargetAddrTable.

The change takes effect immediately. The deleted trusted manager cannot access the device. The agent automatically removes the row in the tgtAddressMaskTable.

To delete the final Trusted Manager, follow these steps:

(The following procedure assumes that there is at least one configured read-write community. There is currently only one Trusted Manager. The taglist for columns for all rows in the srCommunityTable are currently set to MGR. This procedure must be performed from the final Trusted Manager.

1 Set the value of the TransportLabel field on each row in the srCommunityTable to the empty string.

2 Remove the appropriate row from the snmpTargetAddrTable

The change takes effect immediately. All managers can now access the device.

182 3Com VCX V7122 SIP VoIP Gateway User Manual

SNMP Ports

The SNMP Request Port is 161 and the Trap Port is 162. These ports can be changed by setting parameters in the device ini file. The parameter name is:

SNMPPort = <port_number>

Valid UDP port number; default = 161

This parameter specifies the port number for SNMP requests and responses. Usually, it should not be specified. Use the default.

Multiple SNMP Trap Destinations

An agent can now send traps to up to five managers. For each manager, set the following parameters defined in the snmpManagersTable in the acBoardMIB:

snmpTrapManagerSending

snmpManagerIsUsed

snmpManagerTrapPort

snmpManagerIP

When snmpManagerIsUsed is set to zero (not used), the other three parameters are set to zero. snmpManagerIsUsed (Default = Disable(0))

The allowed values are 0 (disable or no) and 1 (enable or yes). snmpManagerIp (Default = 0.0.0.0)

This is known as SNMPMANAGERTABLEIP in the ini file and is the IP address of the manager.

SnmpManagerTrapPort (Default = 162)

The valid port range for this is 100-4000. snmpManagerTrapSendingEnable (Default = Enable(1))

The allowed values are 0 (disable) and 1 (enable).

Each of these MIB objects is independent and can be set regardless of the state of snmpManagerIsUsed.

If the parameter IsUsed is set to 1, the IP address for that row should be supplied in the same SNMP PDU.

Configuration via the ini File

In the VCX V7122 ini file, the parameters below can be set to enable or disable the sending of SNMP traps. Multiple trap destinations can be supported on the device by setting multiple trap destinations in the ini file.

SNMPMANAGERTRAPSENDINGENABLE_<x> = 0 or 1 indicates if traps are to be sent to the specified SNMP trap manager. A value of ‘1’ means that it is enabled, while a value of ‘0’ means disabled.

3Com VCX V7122 SIP VoIP Gateway User Manual 183

<x> = a number 0, 1, 2 which is the array element index. Currently, up to 5 SNMP trap managers can be supported.

Figure 64 presents an example of entries in a device ini file regarding SNMP. The device can be configured to send to multiple trap destinations. The lines in the file below are commented out with the ‘;’ at the beginning of the line. All of the lines below are commented out since the first line character is a semi-colon.

Figure 64

Example of Entries in a Device ini file Regarding SNMP

; SNMP trap destinations

; The board maintains a table of trap destinations containing 5 ;rows. The rows are numbered 0..4. Each block of 4 items below ;apply to a row in the table.

; To configure one of the rows, uncomment all 4 lines in that ;block. Supply an IP address and if necessary, change the port ;number.

; To delete a trap destination, set ISUSED to 0.

; -change these entries as needed

;SNMPManagerTableIP_0=

;SNMPManagerTrapPort_0=162

;SNMPManagerIsUsed_0=1

;SNMPManagerTrapSendingEnable_0=1

;

;SNMPManagerTableIP_1=

;SNMPManagerTrapPort_1=162

;SNMPManagerIsUsed_1=1

;SNMPManagerTrapSendingEnable_1=1

;

;SNMPManagerTableIP_2=

;SNMPManagerTrapPort_2=162

;SNMPManagerIsUsed_2=1

;SNMPManagerTrapSendingEnable_2=1

;

;SNMPManagerTableIP_3=

;SNMPManagerTrapPort_3=162

;SNMPManagerIsUsed_3=1

;SNMPManagerTrapSendingEnable_3=1

;

;SNMPManagerTableIP_4=

;SNMPManagerTrapPort_4=162

;SNMPManagerIsUsed_4=1

;SNMPManagerTrapSendingEnable_4=1

The same information configurable in the ini file can also be configured via the

acBoardMIB.

Configuration via SNMP

To configure trap destinations, the EM must use the snmpTargetMIB. Up to 5 trap destinations can be configured.

To add a trap destination:

Add a row to the snmpTargetAddrTable with these values:

Name=trapN, TagList=AC_TRAP, Params=v2cparams, where N is an unused number between 0 and 4.

184 3Com VCX V7122 SIP VoIP Gateway User Manual

All changes to the trap destination configuration take effect immediately.

To delete a trap destination:

Remove the appropriate row from the snmpTargetAddrTable.

To modify a trap destination:

(You can change the IP address and/or port number for an existing trap destination. The same effect can be achieved by removing a row and adding a new row).

Modify the IP address and/or port number for the appropriate row in the snmpTargetAddrTable.

To disable a trap destination:

Change TagList on the appropriate row in the snmpTargetAddrTable to the empty string.

To enable a trap destination:

Change TagList on the appropriate row in the snmpTargetAddrTable to ‘AC_TRAP’.

SNMP Manager Backward Compatibility

With support for the Multi Manager Trapping feature, the older acSNMPManagerIP MIB object, synchronized with the first index in the snmpManagers MIB table, is also supported.

This is translated in two features:

SET/GET to either of the two MIB objects is identical. i.e., as far as the SET/GET are concerned

OID 1.3.6.1.4.1.5003.9.10.1.1.2.7 is identical to

OID 1.3.6.1.4.1.5003.9.10.1.1.2.21.1.1.3.

When setting ANY IP to the acSNMPManagerIP (this is the older parameter, not the table parameter), two more parameters are SET to ENABLE. snmpManagerIsUsed.0 and snmpManagerTrapSendingEnable.0 are both set to 1.

Element Management System

Using the Element Management System (EMS) is recommended to Customers requiring large deployments (multiple media gateways in globally distributed enterprise offices, for example), that need to be managed by central personnel.

The EMS is not included in the device’s supplied package. Contact 3Com for detailed information on the 3Com EMS and Enterprise VoIP Network solution for large VoIP deployments.

3Com VCX V7122 SIP VoIP Gateway User Manual 185

186 3Com VCX V7122 SIP VoIP Gateway User Manual

C

HAPTER

12: S

ELECTED

T

ECHNICAL

S

PECIFICATIONS

Table 38

VCX V7122 Selected Technical Specifications

Function Specification

Trunk & Channel Capacity

1

Capacity with E1

Capacity with T1

1, 2, 4, 8 or 16 E1 spans, 30, 60, 120, 240 or 480 digital channels

1, 2, 4, 8 or 16 T1 spans, 24, 48, 96, 192 or 384 digital channels

Voice & Tone Characteristics

Voice Compression G.711 PCM at 64 kbps µ-law/A-law

G.723.1 MP-MLQ at 5.3 or 6.3 kbps

G.726 at 32 kbps ADPCM

G.729 CS-ACELP 8 Kbps Annex A / B

NetCoder at 6.4, 7.2, 8.0 and 8.8 kbps

Transparent coder

Silence Suppression

Packet Loss Concealment

(10, 20, 30, 40, 50, 60, 80, 100,120 msec)

(30, 60, 90, 120, 150 msec)

(10, 20, 30, 40, 50, 60, 80, 100,120 msec)

(10, 20, 30, 40, 50, 60, 80, 100,120 msec)

(20, 40, 60, 80, 100, 120 msec).

G.723.1 Annex A

G.729 Annex B

PCM and ADPCM: Standard Silence Descriptor (SID) with Proprietary Voice Activity

Detection (VAD) and Comfort Noise Generation (CNG)

NetCoder

G.711 appendix 1

G.723.1

G.729 a/b

Echo Cancellation

DTMF Detection and

Generation

DTMF Transport (in-band)

G.168, configurable tail length per gateway from 32 to 128 msec

Dynamic range 0 to -25 dBm compliant with TIA 464B and Bellcore TR-NWT-000506.

Mute, transfer in RTP payload or relay in compliance with RFC 2833

Call Progress Tone Detection and Generation

16 tones: single tone or dual tones, programmable frequency & amplitude; 15 frequencies in the range 300 to 1980 Hz, 1 or 2 cadences per tone, up to 2 sets of

ON/OFF periods.

Output Gain Control -32 dB to +31 dB in steps of 1 dB

1

Capacity Limitations:

When the Echo Canceller’s length is set to 64 msec or more, the number of available gateway channels is reduced by a factor of 5/6. For detailed information see the parameter ‘MaxEchoCancellerLength’ (see

Table 24 on page 94 ).

3Com VCX V7122 SIP VoIP Gateway User Manual 187

Function Specification

Input Gain Control -32 dB to +31 dB in steps of 1 dB

Fax and Modem Transport Modes

Real time Fax Relay Group 3 real-time fax relay up to 14400 bps with auto fallback

Tolerant network delay (up to 9 seconds round trip delay)

Fax Transparency

Modem Transparency

Protocols

VoIP Signaling Protocol

Communication Protocols

Telephony Protocols

In-Band Signaling

T.30 (PSTN) and T.38 (IP) compliant (real-time fax)

CNG tone detection & Relay per T.38

Answer tone (CED or AnsAm) detection & Relay per T.38

Automatic fax bypass (pass-through) to G.711, ADPCM or NSE bypass mode

Automatic switching (pass-through) to PCM, ADPCM or NSE bypass mode for modem signals (V.34 or V.90 modem detection)

SIP - RFC 3261

RTP/RTCP packetization.

IP stack (UDP, TCP, RTP).

Remote Software load (TFTP & BootP support).

PRI (EuroISDN, NI2, 4/5ESS, DMS 100, QSIG)

E1/T1 CAS protocols: MFC R2, E&M wink start,

Immediate start, delay start, loop start, ground start,

Feature Group B, D for E1/T1

DTMF (TIA 464A)

MF-R1, MFC R2

User-defined Call Progress Tones

1, 2, 4, 8 or 16 E1/T1/J1 Balanced 120/100 ohm

Two 10/100 Base-TX, half or full duplex with auto-negotiation

Interfaces

Telephony Interface

Network Interface

LED Indicators

LED Indications on Front

Panel

Connectors & Switches

Rear Panel

Trunks

1 to 8 and 9 to 16

Ethernet 1 and 2

AC Power

Power, ACT/Fail, T1/E1 status, LAN status, Swap ready indication

Two 50-pin female Telco connectors (DDK57AE-40500-21D) or 8 RJ-48c connectors for trunks 1 to 8 only

Two 10/100 Base-TX, RJ-45 shielded connectors

Standard IEC320 Appliance inlet.

Option for a dual (fully redundant) power supply.

188 3Com VCX V7122 SIP VoIP Gateway User Manual

Function Specification

DC Power 2-pin terminal block (screw connection type) suitable for field wiring applications connecting DC Power connector: MSTB2.5/2-STF (5.08 mm) from Phoenix Contact.

Bonding and grounding: A 6-32-UNC screw is provided. Correct ring terminal and 16

AWG wire minimum must be used for connection.

Or crimp connection shown below.

Physical

Note that to meet UL approvals, users must fulfill the criteria below.

2-pin terminal block (crimp connection type) comprising a Phoenix Contact Adaptor:

Shroud: MSTBC2,5/2-STZF-5,08.

Contacts: MSTBC-MT0,5-1,0

Cable requirement: 18 AWG x 1.5 m length.

AC Power Supply

DC Power Supply (optional) 36 to 72 VDC (nominal 48 VDC), 4A max, floating input

Environmental (DC) Operation Temp: 0° to 40° C / 32° to 104° F

Short Term Operation Temp (per NEBS): 0° to 55° C / 32° to 131° F

Storage: -40° to 70° C / -40° to 158° F

Humidity: 10 non-condensing

Environmental (AC)

Operation Temp:

Storage:

0° to 40° C / 32° to 104° F

-40° to 70° C / -40° to 158° F

Humidity: 10 non-condensing

Hot Swap

Enclosure Dimensions cPCI cards are full hot swap supported

Power supplies are redundant but not hot swappable

445 x 44 x 300 mm; 17.5 x 1.75 x 12 inch.

Installation

Type Approvals

Universal 90 to 260 VAC 1A max, 47-63 Hz

Option for a dual redundant power supply.

1U 19-inch 2-slot cPCI chassis, Rack mount, shelf or desk top.

Rack mount with 2 side brackets, option 2 extra (rear) side brackets.

Telecommunication Standards IC CS03; FCC part 68

Chassis and Host telecom card are approved to the following telecom standards: IC

CS03; FCC part 68; CTR 4, CTR 12 & CTR 13; JATE; TS-016, TS-038, Anatel, Mexico

Telecom.

Safety and EMC Standards UL 60 950, FCC part 15 Class B, (Class A with Sun 2080 CPU card)

CE Mark (EN 55022 Class B (Class A with Sun 2080 CPU card), EN 60950, EN 55024,

EN 300 386).

Environmental NEBS Level 3: GR-63-Core, GR-1089-Core, Type 1 & 3. Approved for DC powered version.

Complies with ETS 301019; ETS 300019-1, -2, -3. (T 1.1, T 2.3, T3.2). Approved for

AC or DC powered versions.

3Com VCX V7122 SIP VoIP Gateway User Manual 189

Function Specification

Diagnostics

Front panel Status LEDs E1/T1 status

LAN status

Gateway status (Fail, ACT, Power, and Swap Ready).

Syslog events Supported by Syslog Server, per RFC 3164 IETF standard.

SNMP MIBs and Traps SNMP v2c

All specifications in this document are subject to change without prior notice.

190 3Com VCX V7122 SIP VoIP Gateway User Manual

A

PPENDIX

A: VCX V7122 SIP S

OFTWARE

K

IT

Table 39 describes the standard supplied software kit for VCX V7122 SIP gateways. The supplied documentation includes this User’s Manual, the VCX V7122 Fast Track and the

VCX V7122 & TP-1610 SIP Release Notes.

Table 39

VCX V7122 SIP Software Kit

File Name Description

Ram.cmp file

VCX V7122_SIP_xxx.cmp Image file containing the software for the VCX V7122 gateway.

ini files and utilities

VCX V7122_T1.ini

VCX V7122_E1.ini

Usa_tones_xx.dat

Sample ini file for VCX V7122 E1 gateways.

Sample ini file for VCX V7122 T1 gateways.

Default loadable Call Progress Tones dat file.

Usa_tones_xx.ini voice_prompts.dat

DConvert240.exe

Call progress Tones ini file (used to create dat file).

Default loadable Voice Prompts dat file.

TrunkPack Downloadable Conversion Utility.

CAS Protocol Files

MIB Files

CAS Capture Tool

ISDN Capture Tool

Used for various signaling types, such as E_M_WinkTable.dat.

MIB library for SNMP browser.

Utility that is used to convert CAS traces to textual form.

Utility that is used to convert ISDN traces to textual form.

3Com VCX V7122 SIP VoIP Gateway User Manual 191

192 3Com VCX V7122 SIP VoIP Gateway User Manual

A

PPENDIX

B: T

HE

B

OOT

P/TFTP

C

ONFIGURATION

U

TILITY

The BootP/TFTP utility enables you to easily configure and provision our boards and media gateways. Similar to third-party BootP/TFTP utilities (which are also supported) but with added functionality; our BootP/TFTP utility can be installed on Windows 98 or Windows

NT/2000/XP. The BootP/TFTP utility enables remote reset of the device to trigger the initialization procedure (BootP and TFTP). It contains BootP and TFTP utilities with specific adaptations to our requirements.

When to Use the BootP/TFTP

The BootP/TFTP utility can be used with the device as an alternative means of initializing the gateways. Initialization provides a gateway with an IP address, subnet mask, and the default gateway IP address. The tool also loads default software, ini and other configuration files.

BootP Tool can also be used to restore a gateway to its initial configuration, such as in the following instances:

The IP address of the gateway is not known.

The Web browser has been inadvertently turned off.

The Web browser password has been forgotten.

The gateway has encountered a fault that cannot be recovered using the Web browser.

The BootP is normally used to configure the device’s initial parameters. Once this information has been provided, the BootP is no longer needed. All parameters are stored in non-volatile memory and used when the BootP is not accessible.

An Overview of BootP

BootP is a protocol defined in RFC 951 and RFC 1542 that enables an internet device to discover its own IP address and the IP address of a BootP on the network, and to obtain the files from that utility that need to be loaded into the device to function.

A device that uses BootP when it powers up broadcasts a BootRequest message on the network. A BootP on the network receives this message and generates a BootReply. The

BootReply indicates the IP address that should be used by the device and specifies an IP address from which the unit may load configuration files using Trivial File Transfer Protocol

(TFTP) described in RFC 906 and RFC 1350.

Key Features

Internal BootP supporting hundreds of entities.

Contains all required data for our products in predefined format.

3Com VCX V7122 SIP VoIP Gateway User Manual 193

Provides a TFTP address, enabling network separation of TFTP and BootP utilities.

Tools to backup and restore the local database.

Templates.

Option for changing MAC address.

Protection against entering faulty information.

Unicast BootP response.

User-initiated BootP respond, for remote provisioning over WAN.

Filtered display of BootP requests.

Location of other BootP utilities that contain the same MAC entity.

Common log window for both BootP and TFTP sessions.

Works with Windows 98, Windows NT, Windows 2000 and Windows XP.

Specifications

BootP standards: RFC 951 and RFC 1542

TFTP standards: RFC 1350 and RFC 906

Operating System: Windows 98, Windows NT, Windows 2000 and Windows XP

Max number of MAC entries: 200

Installation

To install the BootP/TFTP on your computer, follow these steps:

1 Locate the BootP folder on the VoIP gateway supplied CD ROM and open the file

Setup.exe.

2 Follow the prompts from the installation wizard to complete the installation.

To open the BootP/TFTP, follow these steps:

1 From the Start menu on your computer, navigate to Programs and then click on BootP.

2 The first time that you run the BootP/TFTP, the program prompts you to set the user preferences. See Setting the Preferences on page 197 for information on setting the preferences.

Loading the cmp File, Booting the Device

Once the application is running, and the preferences set (see Setting the Preferences on page 197 ), for each unit that is to be supported, enter parameters into the tool to set up the network configuration information and initialization file names. Each unit is identified by a

MAC address. For information on how to configure (add, delete and edit) units see

Configuring the BootP Clients on page 199 .

To load the software and configuration files, follow these steps:

1 Create a folder on your computer that contains all software and configuration files that are needed as part of the TFTP process.

194 3Com VCX V7122 SIP VoIP Gateway User Manual

2 Set the BootP and TFTP preferences (see Setting the Preferences on page 197 ).

3 Add client configuration for the VoIP gateway that you want to initialize by the BootP,see

Configuring the BootP Clients on page 199 .

4 Reset the VoIP gateway, either physically or remotely, causing the device to use BootP to access the network and configuration information.

BootP/TFTP Application User Interface

Figure 65 shows the main application screen for the BootP/TFTP utility.

Figure 65

Main Screen

Function Buttons on the Main Screen

Pause: Click this button to pause the BootP Tool so that no replies are sent to

BootP requests. Click the button again to restart the BootP Tool so that it responds to all BootP requests. The Pause button provides a depressed graphic when the feature is active.

Edit Clients: Click this button to open a new window that enables you to enter configuration information for each supported VoIP gateway. Details on the Clients window are provided in Configuring the BootP Clients on page 199 .

Edit Templates: Click this button to open a new window that enables you to create or edit standard templates. These templates can be used when configuring new clients that share most of the same settings. Details on the

Templates window are provided in Managing Client Templates on page 203 .

Clear Log: Click this button to clear all entries from the Log Window portion of the main application screen. Details on the log window are provided in

Log Window on page 196 .

3Com VCX V7122 SIP VoIP Gateway User Manual 195

Filter Clients: Click this button to prevent the BootP Tool from logging BootP requests received from disabled clients or from clients which do not have entries in the Clients table.

Reset: Click this button to open a new window where you enter an IP address requests for a gateway that you want to reset. See Figure 66 below.

Figure 66

Reset Screen

When a gateway resets, it first sends a BootRequest. Therefore, Reset can be used to force a BootP session with a gateway without needing to power cycle the gateway. As with any

BootP session, the computer running the BootP Tool must be located on the same subnet as the controlled VoIP gateway.

Log Window

The log window (see Figure 65 on the previous page) records all BootP request and BootP reply transactions, as well as TFTP transactions. For each transaction, the log window displays the following information:

Client: shows the Client address of the VoIP gateway, which is the MAC address of the client for BootP transactions or the IP address of the client for TFTP transactions.

Date: shows the date of the transaction, based on the internal calendar of the computer.

Time: shows the time of day of the transaction, based on the internal clock of the computer.

Status: indicates the status of the transaction.

Client Not Found: A BootRequest was received but there is no matching client entry in the BootP Tool.

Client Found: A BootRequest was received and there is a matching client entry in the

BootP Tool. A BootReply is sent.

Client’s MAC Changed: There is a client entered for this IP address but with a different MAC address.

Client Disabled: A BootRequest was received and there is a matching client entry in the BootP tool but this entry is disabled.

Listed At: Another BootP utility is listed as supporting a particular client when the

Test Selected Client button is clicked (for details on Testing a client see Testing the

Client on page 201 ).

Download Status: Progress of a TFTP load to a client, shown in %.

New IP / File: shows the IP address applied to the client as a result of the BootP transaction, as well as the file name and path of a file transfer for a TFTP transaction.

Client Name: shows the client name, as configured for that client in the Client

Configuration screen.

196 3Com VCX V7122 SIP VoIP Gateway User Manual

Use right-click on a line in the Log Window to open a pop-up window with the following options:

Reset: Selecting this option results in a reset command being sent to the client VoIP gateway. The program searches its database for the MAC address indicated in the line. If the client is found in that database, the program adds the client MAC address to the ARP table for the computer. The program then sends a reset command to the client. This enables a reset to be sent without knowing the current IP address of the client, as long as the computer sending the reset is on the same subnet.

Note: In order to use reset as described above, the user must have administrator privileges on the computer. Attempting to perform this type of reset without administrator privileges on the computer results in an error message. ARP Manipulation Enable must also be turned on in the Preferences window.

View Client: Selecting this option, or double clicking on the line in the log window, opens the Client Configuration window. If the MAC address indicated on the line exists in the client database, it is highlighted. If the address is not in the client database, a new client is added with the MAC address filled out. You can enter data in the remaining fields to create a new client entry for that client.

Setting the Preferences

The Preferences window, Figure 67 , is used to configure the BootP Tool parameters.

Figure 67

Preferences Screen

3Com VCX V7122 SIP VoIP Gateway User Manual 197

BootP Preferences

ARP is a common acronym for Address Resolution Protocol, and is the method used by all

Internet devices to determine the link layer address, such as the Ethernet MAC address, in order to route Datagrams to devices that are on the same subnet.

When ARP Manipulation is enabled on this screen, the BootP Tool creates an ARP cache entry on your computer when it receives a BootP BootRequest from the VoIP gateway. Your computer uses this information to send messages to the VoIP gateway without using ARP again. This is particularly useful when the gateway does not yet have an IP address and, therefore, cannot respond to an ARP.

Because this feature creates an entry in the computer ARP cache, Administrator Privileges are required. If the computer is not set to allow administrator privileges, ARP Manipulation cannot be enabled.

ARP Manipulation Enabled: Enable ARP Manipulation to remotely reset a gateway that does not yet have a valid IP address.

If ARP Manipulation is enabled, the following two commands are available.

Reply Type: Reply to a BootRequest can be either Broadcast or Unicast. The default for the BootP Tool is Broadcast. In order for the reply to be set to Unicast, ARP

Manipulation must first be enabled. This then enables the BootP Tool to find the MAC address for the client in the ARP cache so that it can send a message directly to the requesting device. Normally, this setting can be left at Broadcast.

ARP Type: The type of entry made into the ARP cache on the computer, once ARP

Manipulation is enabled, can be either Dynamic or Static. Dynamic entries expire after a period of time, keeping the cache clean so that stale entries do not consume computer resources. The Dynamic setting is the default setting and the setting most often used.

Static entries do not expire.

Number of Timed Replies: This feature is useful for communicating to VoIP gateways that are located behind a firewall that would block their BootRequest messages from getting through to the computer that is running the BootP Tool. You can set this value to any whole digit. Once set, the BootP Tool can send that number of BootReply messages to the destination immediately after you send a remote reset to a VoIP gateway at a valid

IP address. This enables the replies to get through to the VoIP gateway even if the

BootRequest is blocked by the firewall. To turn off this feature, set the Number of Timed

Replies = 0.

TFTP Preferences

Enabled: To enable the TFTP functionality of the BootP Tool, check the box beside this heading. If you want to use another TFTP application, other than the one included with the BootP Tool, unselect the box.

On Interface: This pull down menu displays all network interfaces currently available on the computer. Select the interface that you want to use for the TFTP. Normally, there is only one choice.

Directory: This option is enabled only when the TFTP is enabled. Use this parameter to specify the folder that contains the files for the TFTP utility to manage (cmp, ini, Call

Progress Tones, etc.).

198 3Com VCX V7122 SIP VoIP Gateway User Manual

Boot File Mask: Boot File Mask specifies the file extension used by the TFTP utility for the boot file that is included in the BootReply message. This is the file that contains VoIP gateway software and normally appears as cmp.

ini File Mask: ini File mask specifies the file extension used by the TFTP utility for the configuration file that is included in the BootReply message. This is the file that contains

VoIP gateway configuration parameters and normally appears as ini.

Timeout: This specifies the number of seconds that the TFTP utility waits before retransmitting TFTP messages. This can be left at the default value of 5 (the more congested your network, the higher the value you should define in these fields).

Maximum Retransmissions: This specifies the number of times that the TFTP utility tries to resend messages after timing out. This can be left at the default value of 10 (the more congested your network, the higher the value you should define in these fields).

Configuring the BootP Clients

The Clients window, shown in Figure 68 below, is used to set up the parameters for each specific VoIP gateway.

Figure 68

Client Configuration Screen

Adding Clients

Adding a client creates an entry in the BootP Tool for a specific gateway.

3Com VCX V7122 SIP VoIP Gateway User Manual 199

To add a client to the list without using a template, follow these steps:

1 Click on the Add New Client Icon; a client with blank parameters is displayed.

2 Enter values in the fields on the right side of the window, using the guidelines for the fields in Setting Client Parameters on page 201 .

3 Click Apply to save this entry to the list of clients, or click Apply & Reset to save this entry to the list of clients and send a reset message to that gateway to immediately implement the settings.

Note: To use Apply & Reset you must enable ARP Manipulation in the Preferences window. Also, you must have administrator privileges for the computer you are using.

An easy way to create several clients that use similar settings is to create a template. For information on how to create a template see Managing Client Templates on page 203 .

To add a client to the list using a template, follow these steps:

1 Click on the Add New Client Icon; a client with blank parameters is displayed.

2 In the field Template, located on the right side of the Client Configuration Window, click on the down arrow to the right of the entry field and select the template that you want to use.

3 The values provided by the template are automatically entered into the parameter fields on the right side of the Client Configuration Window. To use the template parameters, leave the check box next to that parameter selected. The parameter values appear in gray text.

4 To change a parameter to a different value, unselect the check box to the right of that parameter. This clears the parameter provided by the template and enables you to edit the entry. Clicking the check box again restores the template settings.

5 Click Apply to save this entry to the list of clients or click Apply & Reset to save this entry to the list of clients and send a reset message to that gateway to immediately implement the settings.

Note: To use Apply & Reset you must enable ARP Manipulation in the Preferences window. Also, you must have administrator privileges for the computer you are using.

Deleting Clients

To delete a client from the BootP Tool, follow these steps:

1 Select the client that you wish to delete by clicking on the line in the window for that client.

2 Click the Delete Current Client button

3 A warning pops up. To delete the client, click Yes.

Editing Client Parameters

To edit the parameters for an existing client, follow these steps:

1 Select the client that you wish to edit by clicking on the line in the window for that client.

2 Parameters for that client display in the parameter fields on the right side of the window.

3 Make the changes required for each parameter.

200 3Com VCX V7122 SIP VoIP Gateway User Manual

4 Click Apply to save the changes, or click Apply & Reset to save the changes and send a reset message to that gateway to immediately implement the settings.

Note: To use Apply & Reset you must enable ARP Manipulation in the Preferences window. Also, you must have administrator privileges for the computer you are using.

Testing the Client

There should only be one BootP utility supporting any particular client MAC active on the network at any time.

To check if other BootP utilities support this client, follow these steps:

1 Select the client that you wish to test by clicking on the client name in the main area of the Client Configuration Window.

2 Click the Test Selected Client button

3 Examine the Log Window on the Main Application Screen. If there is another BootP utility that supports this client MAC, there is a response indicated from that utility showing the status Listed At along with the IP address of that utility.

4 If there is another utility responding to this client, you must remove that client from either this utility or the other one.

Setting Client Parameters

Client parameters are listed on the right side of the Client Configuration Window.

Client MAC: The Client MAC is used by BootP to identify the VoIP gateway. The MAC address for the VoIP gateway is printed on a label located on the VoIP gateway hardware. Enter the Ethernet MAC address for the VoIP gateway in this field. Click the box to the right of this field to enable this particular client in the BootP tool (if the client is disabled, no replies are sent to BootP requests).

Note: When the MAC address of an existing client is edited, a new client is added, with the same parameters as the previous client.

Client Name: Enter a descriptive name for this client so that it is easier to remember which VoIP gateway the record refers to. For example, this name could refer to the location of the gateway.

Template: Click the pull down arrow if you wish to use one of the templates that you configured. This applies the parameters from that template to the remaining fields.

Parameter values that are applied by the template are indicated by a check mark in the box to the right of that parameter. Uncheck this box if you want to enter a different value.

If templates are not used, the box to the right of the parameters is colored gray and is not selectable.

IP: Enter the IP address you want to apply to the VoIP gateway. Use the normal dotted decimal format.

Subnet: Enter the subnet mask you want to apply to the VoIP gateway. Use the normal dotted decimal format. Ensure that the subnet mask is correct. If the address is incorrect, the VoIP gateway may not function until the entry is corrected and a BootP reset is applied.

Gateway: Enter the IP address for the data network gateway used on this subnet that you want to apply to the VoIP gateway. The data network gateway is a device, such as a router, that is used in the data network to interface this subnet to the rest of the enterprise network.

3Com VCX V7122 SIP VoIP Gateway User Manual 201

TFTP Server IP: This field contains the IP address of the TFTP utility that is used for file transfer of software and initialization files to the gateway. When creating a new client, this field is populated with the IP address used by the BootP Tool. If a different TFTP utility is to be used, change the IP address in this field to the IP address used by the other utility.

Boot File: This field specifies the file name for the software (cmp) file that is loaded by the TFTP utility to the VoIP gateway after the VoIP gateway receives the BootReply message. The actual software file is located in the TFTP utility directory that is specified in the BootP Preferences window. The software file can be followed by command line switches. For information on available command line switches see Using Command Line

Switches on page 202 .

Once the software file loads into the gateway, the gateway begins functioning from

that software. In order to save this software to non-volatile memory, (only the cmp

file, i.e., the compressed firmware file, can be burned to your device's flash memory), the -fb flag must be added to the end of the file name. If the file is not saved, the gateway reverts to the old version of software after the next reset.

The Boot file field can contain up to two file names: cmp file name to be used for

load of application image and ini file name to be used for gateway provisioning.

Either one, two or no file names can appear in the Boot file field. To use both file

names use the ";" separator (without blank spaces) between the xxx.cmp and the

yyy.ini files (e.g., ram.cmp;SIPgw.ini).

ini File: This field specifies the configuration ini file that the gateway uses to program its various settings. Enter the name of the file that is loaded by the TFTP utility to the VoIP gateway after it receives the BootReply message. The actual ini file is located in the

TFTP utility directory that is specified in the BootP Preferences window.

Call Agent: This field specifies the IP address of the MGCP Call Agent that is controlling the gateway. This field can be ignored for all other control/signaling protocols.

Using Command Line Switches

You can add command line switches in the field Boot File.

To use a Command Line Switch, follow these steps:

1 In the field Boot File, leave the file name defined in the field as it is (e.g., ramxxx.cmp).

2 Place your cursor after cmp

3 Press the space bar

4 Type in the switch you require.

Examples: “ramxxx.cmp -fb” to burn flash memory.

“ramxxx.cmp -fb -em 4” to burn flash memory and for Ethernet Mode 4 (auto-negotiate).

Table 40 lists and describes the switches that are available:

202 3Com VCX V7122 SIP VoIP Gateway User Manual

Table 40

Command Line Switch Descriptions

Switch Description

-em #

-br

-bd

-bs

-be

Use this switch to set Ethernet mode.

0 = 10 Base-T half-duplex

1 = 10 Base-T full-duplex

2 = 100 Base-TX half-duplex

3 = 100 Base-TX full-duplex

4 = auto-negotiate (default)

Auto-negotiate falls back to half-duplex mode when the opposite port is not in auto-negotiate but the speed (10 Base-T or 100 Base-TX) in this mode is always configured correctly.

BootP retries. Sets the number of BootP requests the device sends during start-up. The device stops sending BootP requests when either BootP reply is received or Number of Retries is reached. This switch takes effect only from the next device reset.

1 = 1 BootP retry, 1 second

2 = 2 BootP retries, 3 seconds

3 = 3 BootP retries, 6 seconds

4 = 10 BootP retries, 30 seconds

5 = 20 BootP retries, 60 seconds

6 = 40 BootP retries, 120 seconds

7 = 100 BootP retries, 300 seconds

15 = BootP retries indefinitely

BootP delays. Sets the interval between the device’s start-up and the first BootP/DHCP request that is issued by the device. The switch only takes effect from the next reset of the device.

1 = 1 second delay (default).

2 = 10 second delay.

3 = 30 second delay.

4 = 60 second delay.

5 = 120 second delay.

Use –bs 1 to enable the Selective BootP mechanism.

Use –bs 0 to disable the Selective BootP mechanism.

The Selective BootP mechanism enables the gateway’s integral BootP client to filter unsolicited

BootP/DHCP replies (accepts only BootP replies that contain the text “AUDC" in the vendor specific information field). This option is useful in environments where enterprise BootP/DHCP servers provide undesired responses to the gateway’s BootP requests.

Use -be 1 for the device to send device-related initial startup information (such as board type, current IP address, software version, etc.) in the vendor specific information field (in the BootP request). This information can be viewed in the main screen of the BootP/TFTP, under column

'Client Info‘ (see Figure 65 on page 195 showing BootP/TFTP main screen with the column

'Client Info' on the extreme right). For a full list of the vendor specific Information fields see

BootP Support on page 169 .

Note: This option is not available on DHCP servers.

Managing Client Templates

Templates can be used to simplify configuration of clients when most of the parameters are the same.

3Com VCX V7122 SIP VoIP Gateway User Manual 203

Figure 69

Templates Screen

To create a new template, follow these steps:

1 Click on the Add New Template button

2 Fill in the default parameter values in the parameter fields.

3 Click Apply to save this new template.

4 Click OK when you are finished adding templates.

To edit an existing template, follow these steps:

1 Select the template by clicking on its name from the list of templates in the window.

2 Make changes to the parameters, as required.

3 Click Apply to save this new template.

4 Click OK when you are finished editing templates.

To delete an existing template, follow these steps:

1 Select the template by clicking its name from the list of templates in the window.

2 Click on the Delete Current Template button.

3 A warning pop up message appears. To delete the template, click Yes.

Note that if this template is currently in use, the template cannot be deleted.

204 3Com VCX V7122 SIP VoIP Gateway User Manual

A

PPENDIX

C: RTP/RTCP P

AYLOAD

T

YPES

AND

P

ORT

A

LLOCATION

RTP Payload Types are defined in RFC 1889 and RFC 1890. The gateways also have new payload types to enable advanced use of other coder types. These types are reportedly not used by other applications.

See the VCX V7122 & TP-1620 SIP Release Notes for the supported coders.

Payload Types Defined in RFC 1890

Table 41

Packet Types Defined in RFC 1890

Payload Type Description Basic Packet Rate [msec]

18 G.729A 20

200 RTCP Sender Report

Randomly, approximately every 5 seconds (when packets are sent by channel)

201 RTCP Receiver Report Randomly, approximately every 5 seconds (when channel is only receiving)

202

203

204

RTCP SDES packet

RTCP BYE packet

RTCP APP packet

Defined Payload Types

Table 42

Defined Payload Types

Payload Type

35

36

38

Description

G.726 32 kbps

G.726 24 kbps

G.726 40 kbps

Basic Packet Rate [msec]

20

20

20

3Com VCX V7122 SIP VoIP Gateway User Manual 205

50

51

52

45

46

47

49

53

54

55

96

41

42

43

Payload Type

39

40

Description

G.727 16 kbps

G.727 24-16 kbps

G.727 24 kbps

G.727 32-16 kbps

G.727 32-24 kbps

G.727 40-16 kbps

G.727 40-24 kbps

G.727 40-32 kbps

NetCoder 4.8 kbps

NetCoder 5.6 kbps

NetCoder 6.4 kbps

NetCoder 7.2 kbps

NetCoder 8.0 kbps

NetCoder 8.8 kbps

NetCoder 9.6 kbps

20

20

20

20

20

20

20

20

20

20

DTMF relay per RFC 2833

20

20

20

Basic Packet Rate [msec]

20

20

104 RFC 2198 (Redundancy) Same as channel’s voice coder.

Default RTP/RTCP/T.38 Port Allocation

Table 43

Default RTP/RTCP/T.38 Port Allocation

Channel Number RTP Port RTCP Port T.38 Port

206 3Com VCX V7122 SIP VoIP Gateway User Manual

Channel Number RTP Port RTCP Port T.38 Port

: :

n 6000 + 10(n-1)

: :

:

6001 + 10(n-1)

:

:

6002 + 10(n-1)

:

: :

: :

:

:

:

:

: :

: :

: :

:

:

:

:

:

:

To configure the gateway to use the same port for both RTP and T.38 packets, set the parameter ‘T38UseRTPPort’ to 1.

3Com VCX V7122 SIP VoIP Gateway User Manual 207

208 3Com VCX V7122 SIP VoIP Gateway User Manual

A

PPENDIX

D: F

AX AND

M

ODEM

T

RANSPORT

M

ODES

Users can choose for fax, and for each modem type (V.22/V.23/Bell/V.32/V.34), one of the following transport methods:

Fax relay mode (demodulation / remodulation, not applicable to Modem),

Bypass (using a high bit rate coder to pass the signal), or

Transparent (passing the signal in the current voice coder).

When any of the relay modes are enabled, distinction between fax and modem is not immediately possible at the beginning of a session. The channel is therefore in “Answer

Tone” mode until a decision is made The packets sent to the network at this stage are T.38complaint fax relay packets.

Configuring Fax Relay Mode

When FaxTransportType = 1 (relay mode), then when fax is detected the channel automatically switches from the current voice coder to answer tone mode, and then to T.38complaint fax relay mode.

When fax transmission ends, the reverse is carried out, and fax relay switches to voice. This mode switch occurs automatically, both at the local and remote endpoints.

Users can limit the fax rate using the FaxRelayMaxRate parameter and can enable/disable

ECM fax mode using the FaxRelayECMEnable parameter.

When using T.38 mode, the User can define a redundancy feature to improve Fax transmission over congested IP network. This feature is activated by

“FaxRelayRedundancyDepth” and “EnhancedFaxRelayRedundancyDepth” parameters.

Although this is a proprietary redundancy scheme, it should not create problems when working with other T.38 decoders.

T.38 mode currently supports only the T.38 UDP syntax.

Configuring Fax/Modem ByPass Mode

When VxxTransportType=2 (FaxModemBypass, Vxx can be either V32/V22/Bell/V34/Fax), then when fax/modem is detected, the channel automatically switches from the current voice coder to a high bit-rate coder, as defined by the User, with the FaxModemBypassCoderType configuration parameter.

During the bypass period, the coder uses the packing factor (by which a number of basic coder frames are combined together in the outgoing WAN packet) set by the User in the

FaxModemBypassM configuration parameter. The network packets to be generated and

3Com VCX V7122 SIP VoIP Gateway User Manual 209

received during the bypass period are regular voice RTP packets (per the selected bypass coder) but with a different RTP Payload type.

When fax/modem transmission ends, the reverse is carried out, and bypass coder is switched to regular voice coder.

Supporting V.34 Faxes

V.34 fax machine support is available only in bypass mode (fax relay is not supported) when the channel is configured in one of the configurations described below:

FaxTransportMode = 2 (Bypass)

V34ModemTransportType = 2 (Modem bypass)

In this configuration, both T.30 and V.34 faxes work in Bypass mode

Or

FaxTransportMode = 1 (Relay)

V34ModemTransportType = 2 (Modem bypass)

In this configuration, T.30 faxes use Relay mode (T.38) while V.34 Fax uses Bypass mode.

In order to use V.34 fax in Relay mode (T.38), you must configure:

FaxTransportMode = 1 (Relay)

V34ModemTransportType = 0 (Transparent)

V32ModemTransportType = 0

V23ModemTransportType = 0

V22ModemTransportType = 0

This configuration forces the V.34 fax machine to work in T.30 mode.

210 3Com VCX V7122 SIP VoIP Gateway User Manual

A

PPENDIX

E: VCX V7122 C

LOCK

S

ETTINGS

The gateway can either generate its own timing signals, using an internal clock, or recover them from one of the E1/T1 trunks.

To use the internal gateway clock source configure the following parameters:

TDMBusClockSource 1

ClockMaster = 1 (for all gateway trunks)

To use the recovered clock option configure the following parameters:

TDMBusClockSource = 4

ClockMaster_x = 0 (for all "slave" gateway trunks connected to PBX#1)

ClockMaster_x = 1 (for all "master" gateway trunks connected to PBX#2)

Assuming that the gateway recovers its internal clock from one of the "slave" trunks connected to PBX#1, and provides clock to PBX#2 on its "master" trunks.

In addition it is necessary to define from which of the "slave" trunks the gateway recovers its clock:

TDMBusPSTNAutoClockEnable = 1 (the gateway automatically selects one of the connected "slave" trunks)

Or

TDMBusLocalReference = # (Trunk index: 0 to 7, default = 0)

To configure the TDM Bus Clock Source parameters see Configuring the TDM Bus

Settings on page 72 .

3Com VCX V7122 SIP VoIP Gateway User Manual 211

212 3Com VCX V7122 SIP VoIP Gateway User Manual

A

PPENDIX

F: C

USTOMIZING THE

VCX

V7122 W

EB

I

NTERFACE

Customers incorporating the VCX V7122 into their portfolios can customize the device’s Web

Interface to suit their specific corporate logo and product naming conventions.

Customers can customize the Web Interface’s title bar (3Com’s title bar is shown in

Figure 70 ; a customized title bar is shown in Figure 71 ).

Figure 70

User-Customizable Web Interface Title Bar

Corporate logo can be OEMcustomized

Figure 71

Customized Web Interface Title Bar

Background image can be

OEM-customized

Product name can be

OEM-customized

To customize the title bar via the Web Interface, follow these steps:

1 Replace the main corporate logo (see

Replacing the Main Corporate Logo below).

2 Replace the title bar’s background image file (see Replacing the Background Image File on page 215 ).

3 Customize the product’s name (see Customizing the Product Name on page 216 ).

Replacing the Main Corporate Logo

The main corporate logo can be replaced either with a different logo image file (see

Replacing the Main Corporate Logo with an Image File below) or with a text string (see

Replacing the Main Corporate Logo with a Text String on page 215 ). Note that when the main corporation logo is replaced, the 3Com logo on the left bar (see Figure 17 on page 44 ) and in the Software Upgrade Wizard (see Software Upgrade Wizard on page 81 ) disappear.

Also note that the browser’s title bar is automatically updated with the string assigned to the

WebLogoText parameter when the 3Com default logo is not used.

3Com VCX V7122 SIP VoIP Gateway User Manual 213

Replacing the Main Corporate Logo with an Image File

Use a gif, jpg or jpeg file for the logo image. It is important that the image file has a fixed height of 59 pixels (the width can be configured). The combined size of the image files (logo and background) is limited to 64 kbytes.

To replace the default logo with your own corporate image via the Web Interface, follow these steps:

1 Access the VCX V7122 Embedded Web Server (see Accessing the Embedded Web

Server on page 42 ).

2 In the URL field, append the suffix ‘AdminPage’ (note that it’s case-sensitive) to the IP address, e.g., http://10.1.229.17/AdminPage.

3 Click Image Load to Device; the Image Download screen is displayed (see

Figure 72 ).

Figure 72

Image Download Screen

4 Click the Browse button in the Send Logo Image File from your computer to the

Device box. Navigate to the folder that contains the logo image file you want to load.

5 Click the Send File button; the file is sent to the device. When loading is complete, the screen is automatically refreshed and the new logo image is displayed.

6 Note the appearance of the logo. If you want to modify the width of the logo (the default width is 339 pixels), in the Logo Width field, enter the new width (in pixels) and press the Set Logo Width button.

7 To save the image to flash memory so it is available after a power fail see Save

Configuration on page 88 .

The new logo appears on all Web Interface screens.

If you encounter any problem during the loading of the files, or you want to restore the default images, click the Restore Default Images button.

214 3Com VCX V7122 SIP VoIP Gateway User Manual

To replace the default logo with your own corporate image via the ini file, follow these steps:

1 Place your corporate logo image file in the same folder as where the device’s ini file is located (i.e., the same location defined in the BootP/TFTP configuration utility). For detailed information on the BootP/TFTP, see Appendix B: The BootP/TFTP

Configuration Utility on page 193 .

2 Add/modify the two ini file parameters in Table 44 according to the procedure described in Modifying an ini File on page 91 .

Note that loading the device’s ini file via the ‘Configuration File’ screen in the Web Interface doesn’t load the corporate logo image files as well.

Table 44

Customizable Logo ini File Parameters

Parameter Description

LogoFileName The name of the image file containing your corporate logo.

Use a gif, jpg or jpeg image file.

The default is 3Com’s logo file.

Note: The length of the name of the image file is limited to 47 characters.

LogoWidth Width (in pixels) of the logo image.

Note: The optimal setting depends on the resolution settings.

The default value is 339, which is the width of 3Com’s displayed logo.

Replacing the Main Corporate Logo with a Text String

The main corporate logo can be replaced with a text string.

To replace the 3Com default logo with a text string via the Web Interface, modify the two

ini file parameters in according to the procedure described in Modifying ini File

Parameters via the Web AdminPage on page 217 .

To replace the 3Com default logo with a text string via the ini file, add/modify the two ini file parameters in Table 45 according to the procedure described in Modifying an ini File on page 91 .

Table 45

Web Appearance Customizable ini File Parameters

Parameter Description

UseWebLogo 0 = Logo image is used (default).

1 = Text string is used instead of a logo image.

WebLogoText Text string that replaces the logo image.

The string can be up to 15 characters.

Replacing the Background Image File

The background image file is duplicated across the width of the screen. The number of times the image is duplicated depends on the width of the background image and screen resolution. When choosing your background image, keep this in mind.

Use a gif, jpg or jpeg file for the background image. It is important that the image file has a fixed height of 59 pixels. The combined size of the image files (logo and background) is limited to 64 kbytes.

3Com VCX V7122 SIP VoIP Gateway User Manual 215

To replace the background image via the Web, follow these steps:

1 Access the VCX V7122 Embedded Web Server (see Accessing the Embedded Web

Server on page 42 ).

2 In the URL field, append the suffix ‘AdminPage’ (note that it’s case-sensitive) to the IP address, e.g., http://10.1.229.17/AdminPage.

3 Click the Image Load to Device, the Image Download screen is displayed (see

Figure 72 on page 214 ).

4 Click the Browse button in the Send Background Image File from your computer to

gateway box. Navigate to the folder that contains the background image file you want to load.

5 Click the Send File button; the file is sent to the device. When loading is complete, the screen is automatically refreshed and the new background image is displayed.

6 To save the image to flash memory so it is available after a power fail see

Save Configuration on page 88 .

The new background appears on all Web Interface screens.

If you encounter any problem during the loading of the files, or you want to restore the default images, click the Restore Default Images button.

When replacing both the background image and the logo image, first load the logo image followed by the background image.

To replace the background image via the ini file, follow these steps:

1 Place your background image file in the same folder as where the device’s ini file is located (i.e., the same location defined in the BootP/TFTP configuration utility). For detailed information on the BootP/TFTP, see Appendix B: The BootP/TFTP

Configuration Utility on page 193 .

2 Add/modify the ini file parameters in Table 46 according to the procedure described in

Modifying an ini File on page 91 .

Note that loading the device’s ini file via the ‘Configuration File’ screen in the Web Interface doesn’t load the logo image file as well.

Table 46

Customizable Logo ini File Parameters

Parameter Description

BkgImageFileName The name (and path) of the file containing the new background.

Use a gif, jpg or jpeg image file.

The default is 3Com’s background file.

Note: The length of the name of the image file is limited to 47 characters.

Customizing the Product Name

The Product Name text string can be modified according to OEMs specific requirements.

To replace 3Com’s default product name with a text string via the Web Interface, modify the two ini file parameters in Table 48 according to the procedure described in Modifying ini File Parameters via the Web AdminPage on page 217 .

216 3Com VCX V7122 SIP VoIP Gateway User Manual

To replace 3Com’s default product name with a text string via the ini file, add/modify the two ini file parameters in Table 47 according to the procedure described in Modifying an ini File on page 91 .

Table 47

Web Appearanc Customizable ini File Parameters

Parameter Description

UseProductName

UserProductName

0 = Don’t change the product name (default).

1 = Enable product name change.

Text string that replaces the product name.

The default is “VCX V7122”.

The string can be up to 29 characters.

Modifying ini File Parameters via the Web AdminPage

To modify ini file parameters via the AdminPage, follow these steps:

1 Access the VCX V7122 Embedded Web Server (see Accessing the Embedded Web

Server on page 42).

2 In the URL field, append the suffix ‘AdminPage’ (note that it’s case-sensitive) to the IP address, e.g., http://10.1.229.17/AdminPage.

3 Click the INI Parameters option, the INI Parameters screen is displayed (see

Figure 73 ).

Figure 73

INI Parameters Screen

4 In the Parameter Name dropdown list, select the required ini file parameter.

3Com VCX V7122 SIP VoIP Gateway User Manual 217

5 In the Enter Value field to the right, enter the parameter’s new value.

6 Click the Apply new value button to the right; the INI Parameters screen is refreshed, the parameter name with the new value appears in the fields at the top of the screen and the Output Window displays a log displaying information on the operation.

You cannot load the image files (e.g., logo/background image files) to the device by choosing a file name parameter in this screen.

218 3Com VCX V7122 SIP VoIP Gateway User Manual

A

PPENDIX

G: A

CCESSORY

P

ROGRAMS AND

T

OOLS

The accessory applications and tools shipped with the device provide you with friendly interfaces that enhance device usability and smooth your transition to the new VoIP infrastructure. The following applications are available:

TrunkPack Downloadable Conversion Utility (see TrunkPack Downloadable Conversion

Utility below).

PSTN Trace Utility (see Creating a Loadable Prerecorded Tones File on page 224 ).

TrunkPack Downloadable Conversion Utility

Use the TrunkPack Downloadable Conversion Utility to:

Create a loadable Call Progress Tones file (see Converting a CPT ini File to a Binary dat

File on page 220 ).

Create a loadable Voice Prompts file from pre-recorded voice messages (see Creating a

Loadable Voice Prompts File on page 221 ).

Encode / decode an ini file (see Encoding/Decoding an ini File on page 223 ).

Create a loadable Prerecorded Tones file (see Encoding/Decoding an ini File on page 223 ).

3Com VCX V7122 SIP VoIP Gateway User Manual 219

Figure 74

TrunkPack Downloadable Conversion Utility Opening Screen

Converting a CPT ini File to a Binary dat File

For detailed information on creating a CPT ini file see Configuring the Call Progress Tones on page 139 .

To convert a CPT ini file to a binary dat file, follow these steps:

1 Execute the TrunkPack Downloadable Conversion Utility, DConvert240.exe (supplied with the software package); the utility’s main screen opens (see Figure 74 ).

2 Click the Process Call Progress Tones File(s) button; the Call Progress Tones screen, shown in Figure 75 , opens.

220 3Com VCX V7122 SIP VoIP Gateway User Manual

Figure 75

Call Progress Tones Conversion Screen

3 Click the Select File… button that is in the ‘Call Progress Tone File’ box.

4 Navigate to the folder that contains the CPT ini file you want to convert.

5 Click the ini file and click the Open button; the name and path of both the ini file and the

(output) dat file appears in the fields below the Select File button.

6 Enter the Vendor Name, Version Number and Version Description in the corresponding required fields under the ‘User Data’ section.

7 Set ‘CPT Version’ to ‘Version 1’ only if you use this utility with a version released before version 4.4 of the device software (this field is used to maintain backward compatibility).

8 Check the ‘Use dBm units for Tone Levels’ check box. Note that the levels of the Call

Progress Tones (in the CPT file) must be in -dBm units.

9 Click the Make File button; you’re prompted that the operation (conversion) was successful.

10 Close the application.

Creating a Loadable Voice Prompts File

For detailed information on the Voice Prompts file see Prerecorded Tones (PRT) File on page 141 .

To create a loadable Voice Prompts dat file from your voice recording files, follow these steps:

1 Execute the TrunkPack Downloadable Conversion Utility, DConvert240.exe (supplied with the software package); the utility’s main screen opens (see Figure 74 on page 220).

2 Click the Process Voice Prompts File(s) button; the Voice Prompts screen, shown in

Figure 76 , opens.

3Com VCX V7122 SIP VoIP Gateway User Manual 221

Figure 76

Voice Prompts Screen

3 To add the pre-recorded voice files to the ‘Voice Prompts’ screen follow one of these procedures:

Select the files and drag them to the ‘Voice Prompts’ screen.

Prompt files and press the Add>> button. Close the ‘Select Files’ screen.

4 Arrange the files according to your requirements by dragging and dropping them from one location in the list to another. Note that the sequence of the files determines their assigned Voice Prompt ID.

Use the Play button to play wav files through your PC speakers.

Use the Remove and Remove all buttons to delete files from the list.

222 3Com VCX V7122 SIP VoIP Gateway User Manual

5 For each of the raw files, select a coder that corresponds with the coder it was originally recorded in by completing the following steps:

Double-click or right-click the required file(s); the ‘File Data’ window (shown in Figure 77 ) appears.

From the ‘Coder’ drop-down list, select the required coder type.

In the ‘Description’ field, enter additional identifying information.

Close the ‘File Data’ window.

Note that for wav files, a coder is automatically selected from the wav file’s header.

Figure 77

File Data Window

6 In the ‘Output’ field, specify the output directory in which the Voice Prompts file is generated followed by the name of the Voice Prompts file (the default name is

voiceprompts.dat).

7 Press the Make File(s) button; the Voice Prompts loadable file is produced.

Encoding/Decoding an ini File

For detailed information on secured ini file see Secured ini File on page 91 .

To encode an ini file, follow these steps:

1 Execute the TrunkPack Downloadable Conversion Utility, DConvert240.exe (supplied with the software package); the utility’s main screen opens (see Figure 74 on page 220 ).

2 Click the Process Encoded/Decoded ini file(s) button; the ‘Encode/Decode ini File(s)’ screen, shown in Figure 78 , opens.

3Com VCX V7122 SIP VoIP Gateway User Manual 223

Figure 78

Encode/Decode ini File(s) Screen

3 Click the Select File… button under the ‘Encode ini File(s)’ section.

4 Navigate to the folder that contains the ini file you want to encode.

5 Click the ini file and click the Open button; the name and path of both the ini file and the output encoded file appear in the fields under the Select File button. Note that the name and extension of the output file can be modified.

6 Click the Encode File(s) button; an encoded ini file with the name and extension you specified is created.

To decode an encoded ini file, follow these steps:

1 Click the Select File… button under the ‘Decode ini File(s)’ section.

2 Navigate to the folder that contains the file you want to decode.

3 Click the file and click the Open button. the name and path of both the encode ini file and the output decoded file appear in the fields under the Select File button. Note that the name of the output file can be modified.

4 Click the Decode File(s) button; a decoded ini file with the name you specified is created.

Note that the decoding process verifies the input file for validity. Any change made to the encoded file causes an error and the decoding process is aborted.

Creating a Loadable Prerecorded Tones File

For detailed information on the PRT file, see Prerecorded Tones (PRT) File on page 141 .

To create a loadable PRT dat file from your raw data files, follow these steps:

224 3Com VCX V7122 SIP VoIP Gateway User Manual

1 Prepare the prerecorded tones (raw data PCM or L8) files you want to combine into a single dat file using standard recording utilities.

2 Execute the TrunkPack Downloadable Conversion utility, DConvert240.exe (supplied with the software package); the utility’s main screen opens (shown Figure 74 on page 220 ).

3 Click the Process Prerecorded Tones File(s) button; the Prerecorded Tones File(s) screen, shown in Figure 79 , opens.

Figure 79

Prerecorded Tones Screen

4 To add the prerecorded tone files (you created in Step

1 ) to the ‘Prerecorded Tones’ screen follow one of these procedures:

Select the files and drag them to the ‘Prerecorded Tones’ screen.

Prerecorded Tone files and press the Add>> button. Close the ‘Select Files’ screen.

3Com VCX V7122 SIP VoIP Gateway User Manual 225

5 For each raw data file, define a Tone Type, a Coder and a Default Duration by completing the following steps:

Double-click or right-click the required file; the ‘File Data’ window (shown in Figure 77 on page 223 ) appears.

From the ‘Type’ drop-down list, select the tone type this raw data file is associated with.

From the ‘Coder’ drop-down list, select the coder that corresponds to the coder this raw data file was originally recorded with.

In the ‘Description’ field, enter additional identifying information (optional).

In the ’Default’ field, enter the default duration this raw data file is repeatedly played.

Close the ‘File Data’ window (press the Esc key to cancel your changes); you are returned to the Prerecorded Tones File(s) screen.

Figure 80

File Data Window

6 In the ‘Output’ field, specify the output directory in which the PRT file is generated followed by the name of the PRT file (the default name is prerecordedtones.dat).

Alternatively, use the Browse button to select a different output file. Navigate to the desired file and select it; the selected file name and its path appear in the ‘Output’ field.

7 Click the Make File(s) button; the Progress bar at the bottom of the window is activated.

The dat file is generated and placed in the directory specified in the ‘Output’ field. A message box informing you that the operation was successfu.

PSTN Trace Utility

These utilities are designed to convert PSTN trace binary files to textual form. The binary

PSTN trace files are generated when the user sets the PSTN interface to trace mode.

Operation

Generating textual trace/audit file for CAS protocols -

To generate a readable text file out of the binary trace file when using CAS protocols, rename the PSTN trace binary file to CASTrace0.dat and copy it to the same directory in which the translation utility CAS_Trace.exe is located. Then, run CAS_Trace.exe (no arguments are required). As a result, the textual file CASTrace0.txt is created.

Generating textual trace/audit file for ISDN PRI protocols -

To generate a readable text file out of the binary trace file when using ISDN protocols, copy the PSTN trace binary file to the same directory in which the translation utility

Convert_Trace.bat is located. The following files should reside in the same directory:

Dumpview.exe, Dumpview.cfg and ReadMe.txt. Please read carefully the ReadMe.txt in order to understand the usage of the translation utility. Next, run the Convert_Trace.bat. As a result, the textual file is created.

226 3Com VCX V7122 SIP VoIP Gateway User Manual

To start and collect the PSTN trace via the Web, please use the following instructions. (See

Figure 81 for a view of the Trunk Traces). Also, please note if the PSTN trace was of a PRI or CAS collection based on the framer involved in the trace. This information is needed to properly parse the captured data.

1 Run the UDP2File utility.

2 Determine the trace file name.

3 Determine the UDP port.

4 Mark the PSTN Trace check box.

5 Push the Run button=> the UDP2File utility starts to collect the trace messages.

6 Activate the Web page by entering <M2K IP address>/TrunkTraces, such as: http://10.8.8.101/TrunkTraces. The user and password is the same for the unit.

7 In the Web page set the trace level of each trunk.

8 Enable the trace via the Web.

9 Determine the UDP port (the same as in step 3).

10 Push the Submit button => the board starts to send the trace messages.

In the UDP2File utility (see Figure 82 ) you should see the number in the packets counter increasing.

Figure 81

Trunk Traces

Figure 82

UDP2File Utility

3Com VCX V7122 SIP VoIP Gateway User Manual 227

228 3Com VCX V7122 SIP VoIP Gateway User Manual

A

PPENDIX

H: S

OFTWARE

U

PGRADE

K

EY

3Com supplies VCX V7122 devices to customers with a Software Upgrade Key already preconfigured.

Customers can later upgrade their VCX V7122 features and capabilities by specifying what upgrades they require, and purchasing a new key from 3Com to match their specification.

The Software Upgrade Key is sent to customers as a string in an ini file, to be loaded into the

VCX V7122. Stored in the device’s non-volatile flash memory, the string defines the features and capabilities allowed by the specific key purchased by the customer. The device allows customers to utilize only these features and capabilities. A new key overwrites a previously installed key.

The Software Upgrade Key is an encrypted key. Each device utilizes a unique key.

The Software Upgrade Key is provided by 3Com only.

Loading the Software Upgrade Key

Customers can load a Software Upgrade Key using:

The Embedded Web Server (see Loading the Software Upgrade Key Using the

Embedded Web Server below).

The BootP/TFTP configuration utility (see Loading the Software Upgrade Key Using

BootP/TFTP on page 231 ).

Access the Syslog server to verify that the key was successfully loaded (see Verifying that the Key was Successfully Loaded on page 231 ). If the Syslog server indicates that a

Software Upgrade Key was unsuccessfully loaded, see Troubleshooting an Unsuccessful

Loading of a Key on page 231 to troubleshoot the issue.

After verifying that the Software Upgrade Key was successfully loaded, reset the

device and reload all other related configuration files: ini, CAS and Call Progress

Tones. To keep your previous configuration, follow the Software Upgrade procedure described in Software Upgrade Wizard on page 81 .

Loading the Software Upgrade Key Using the Embedded Web Server

To load a Software Upgrade Key using the Web Server, follow these steps:

1 After receiving the Software Upgrade Key ini file from 3Com, it’s recommended to save it to the same location where you saved the device’s ini file, cmp file and other configuration files.

3Com VCX V7122 SIP VoIP Gateway User Manual 229

2 Open the file to check its contents. Do not modify the contents of the file in any way.

Verify that the ini file you’ve opened is the Software Upgrade Key ini file and none other; its first line must be [LicenseKeys]. Close the file.

3 Access the devices Embedded Web Server (see Configuring the Web Interface via the ini File on page 42 ).

4 Click the Software Update button.

5 Click the Software Upgrade Key tab; the Software Upgrade Key screen is displayed

(shown in Figure 83 ).

6 When loading a single key S/N line (Serial Number) to a device:

Open the Software Upgrade Key ini file (it should open in Notepad), select and copy the key string of the device’s S/N and paste it into the Web field New Key. If the string is sent in the body of an email, copy and paste it from there. Press the Add Key button.

7 When loading a Software Upgrade Key ini file containing multiple S/N (Serial Number) lines to a device (see Figure 84 ):

Click the Browse button in the Send “Upgrade Key” file from your computer to the

device field, and navigate to the Software Upgrade Key ini file.

Click the Send File button.

The new key is displayed in the Current Key field. Information on the key and available features and capabilities are displayed in the Status pane.

Figure 83

Software Upgrade Key Screen

230 3Com VCX V7122 SIP VoIP Gateway User Manual

Figure 84

Example of a Software Upgrade Key ini File Containing Multiple S/N Lines

Loading the Software Upgrade Key Using BootP/TFTP

To load the Software Upgrade Key file using BootP/TFTP, follow these steps:

1 Place the ini file in the same location you’ve saved the device’s cmp file.

2 Start the BootP/TFTP configuration utility and from the Services menu in the main screen, choose option Clients; the Client Configuration screen is displayed (see

Figure 68 on page 199 ).

3 From the drop-down list in the INI File field, select the Software Upgrade Key ini file instead of the device’s ini file. Note that the device’s cmp file must be specified in the

Boot File field.

4 Configure the initial BootP/TFTP parameters required, and click OK (see

Appendix B:

The BootP/TFTP Configuration Utility on page 193 ).

5 Reset the device; the device’s cmp and Software Upgrade Key files are loaded to the device.

Verifying That the Key was Successfully Loaded

Verify that the key was successfully loaded to the device by accessing the Syslog server. For detailed information on the Syslog server see Syslog Support on page 163 . When a key is successfully loaded, the following message is issued in the Syslog server:

"S/N___ Key Was Updated. The Board Needs to be Reloaded with ini file\n"

(S/N is the device’s serial number).

After installing the key, customers can determine in the Embedded Web Server’s read-only

Status pane (Software Update menu > Software Upgrade Key) (see Figure 83 on page 230 ) that the features and capabilities activated by the installed string match those that they ordered.

Troubleshooting an Unsuccessful Loading of a Key

If the Syslog server indicates that a Software Upgrade Key ini file was unsuccessfully loaded

(the SN_ line is blank), take the following preliminary actions to troubleshoot the issue:

Open the Software Upgrade Key ini file and check that the S/N line of the specific device whose key you want to update is listed in it. If it isn’t, contact 3Com.

3Com VCX V7122 SIP VoIP Gateway User Manual 231

Verify that you’ve loaded the correct ini file and that you haven’t loaded the device’s ini file or the CPT ini file by mistake. Open the file and ensure that the first line is

[LicenseKeys].

Verify that you didn’t alter in any way the contents of the ini file.

232 3Com VCX V7122 SIP VoIP Gateway User Manual

A

PPENDIX

I: R

ELEASE

R

EASON

M

APPING

Table 48 below describes the mapping of ISDN release reason to SIP response. Table 49 on page 235 describes the mapping of SIP response to ISDN release reason.

Table 48

Mapping of ISDN Release Reason to SIP

2

3

6

ISDN Release

Reason

1

7

16

Description

Unallocated number

No route to network

No route to destination

Channel unacceptable

Call awarded and being delivered in an established channel

Normal call clearing

SIP

Response

404

404

404

406*

-*

Description

Not found

Not found

Not found

Not acceptable

BYE

18

19

No user responding

No answer from the user

408

480

Request timeout

Temporarily unavailable

28

29

30

34

38

22

22

23

26

Number changed w/o diagnostic

Number changed with diagnostic

410

410

Redirection to new destination 480

Non-selected user clearing 404

Gone

Gone

Not found

Address incomplete

Facility rejected

Response to status enquiry

No circuit available

Network out of order

484

501

501*

503

503

Address incomplete

Not implemented

Not implemented

Service unavailable

Service unavailable

3Com VCX V7122 SIP VoIP Gateway User Manual 233

63

65

66

83

84

85

86

70

79

81

82

50

55

57

44

47

49

ISDN Release

Reason

41

42

43

58

69

87

Description

Temporary failure

Switching equipment congestion

Access information discarded

Requested channel not available

Resource unavailable

QoS unavailable

Facility not subscribed 503*

Incoming calls barred within CUG 403

403 Bearer capability not authorized

Bearer capability not presently available

Service/option not available 503*

Bearer capability not implemented 501

480* Channel type not implemented

Requested facility not implemented

Only restricted digital information bearer capability is available

Service or option not implemented 501

SIP

Response

503

503

502*

503*

503

503*

Invalid call reference value

Identified channel does not exist

502*

502*

Suspended call exists, but this call identity does not

Call identity in use 503*

503* No call suspended

Call having the requested call identity has been cleared

User not member of CUG 503

Description

Service unavailable

Service unavailable

Bad gateway

Service unavailable

Service unavailable

Service unavailable

Service unavailable

Forbidden

Forbidden

Service unavailable

Temporarily unavailable

Not implemented

Bad gateway

Bad gateway

Service unavailable

Service unavailable

Service unavailable

91

95

Invalid transit network selection

Invalid message

502*

503

Bad gateway

Service unavailable

234 3Com VCX V7122 SIP VoIP Gateway User Manual

ISDN Release

Reason

96

97

98

99

100

101

Description

SIP

Response

Description

Mandatory information element is missing

Message type non-existent or not implemented

Message not compatible with call state or message type nonexistent or not implemented

Information element non-existent or not implemented

Invalid information elements contents

Message not compatible with call state

Recovery of timer expiry

Protocol error

Interworking unspecified

409* Conflict

409* Conflict

408

500

500

Request timeout

Server internal error

Server internal error

102

111

127

Table 49

Mapping of SIP Response to ISDN Release Reason

SIP

Response

Description

400* Bad request

401 Unauthorized

402 Payment required

403 Forbidden

404 Not found

405

406

Method not allowed

Not acceptable

ISDN Release

Reason

31

21

21

21

1

63

79

407

Proxy authentication required

Request timeout 408

409 Conflict

410 Gone

102

41

22

411 Length 127

413 Request entity too long 127

Description

Normal, unspecified

Call rejected

Unallocated number

Service/option unavailable

Service/option not implemented

Recovery on timer expiry

Number changed w/o diagnostic

Interworking

Interworking

3Com VCX V7122 SIP VoIP Gateway User Manual 235

SIP

Response

414

415

Description

Request URI too long

Unsupported media type

ISDN Release

Reason

127

79

127

18

Description

Interworking

Service/option not implemented

Interworking

No user responding 480 Temporarily unavailable

488

500

501

502

503

504

505*

481*

483

484

Call leg/transaction doesn’t exist

127 Interworking

127 Interworking

Too many hops

Address incomplete

485 Ambiguous

25

28

1

Exchange – routing error

Invalid number format

Not acceptable here

Server internal error

Not implemented

Bad gateway

Service unavailable

Server timeout

Version not supported

17

31

41

38

38

41

102

127

Normal, unspecified

Temporary failure

Network out of order

Network out of order

Temporary failure

Recovery on timer expiry

Interworking

603 Decline

604 Does not exist anywhere

606* Not acceptable

21

1

38

Unallocated number

Network out of order

236 3Com VCX V7122 SIP VoIP Gateway User Manual

A

PPENDIX

J: RADIUS B

ILLING AND

C

ALLING

C

ARD

A

PPLICATION

The VCX V7122 calling card application capability (included in its IVR - Interactive Voice

Response - feature) enables Internet Telephony Service Providers (ITSPs) to provide a VoIP telephone service to subscribers who have purchased calling cards in advance.

The subscriber market for calling cards is growing exponentially worldwide. Calling cards are often much cheaper than collect calls and operator-assisted calls made through long distance providers and local phone companies. VoIP service providers, using the VCX

V7122, can further reduce costs for calling card subscribers and substantially reduce implementation time, making the service extraordinarily attractive both for them and their subscribers.

Benefits

Using the VCX V7122, telephony service providers can offer the calling card service over a

VoIP network and thereby:

Lower the cost and deployment time that a PSTN calling-card service requires

Achieve voice quality comparable to toll quality

Acquire a cost-effective, reliable VoIP network infrastructure

IVR functionality can be located at the edge of the network (distributed functionality) or in a central location

Connect with the PSTN over carrier interfaces

Interoperate with other VoIP service providers and other vendors' VoIP equipment

Become part of a world-wide network of other VoIP service providers interested in interconnecting

Features

PSTN IP and IP PSTN distributed IVR architecture

Authentication Dial In User Service/Service) - stored on a RADIUS server

Provides comprehensive management of accounting and billing support

CDRs (Call Detail Reports) over RADIUS (stored on a RADIUS server) disconnected (a short Prompt is played prior to disconnection)

Common internal, on-board, Voice Prompts (in flash memory) for all VoiceXML (Voice

Extensible Markup Language) scripts

Multiple VXML scripts stored on external HTTP server (up to 10 different scripts)

3Com VCX V7122 SIP VoIP Gateway User Manual 237

Caller can place multiple successive calls without re-entering the account and password numbers (authentication and authorization are applied without collecting the information from the user again)

Supports Cisco gateway RADIUS functionality

Interoperates with standard RADIUS-based (AAA) billing servers

Supports 240 concurrent calls running a VXML script

Loads the VXML scripts once and stores them in the RAM; scripts can be changed without the need of reset

Barge-in dialing (to shorten menu time), once prompt has started

Supported Architecture

Figure 85

VCX V7122 Supported Architecture

Figure 85 illustrates standard Calling Card IVR application architecture. The figure depicts in general terms an incoming PSTN IP call being conveyed to the IP network.

The architecture comprises the following components:

VCX V7122 - The PSTN gateway that includes the VoiceXML interpreter which generates events in response to user actions (e.g., spoken or character input received, disconnect) and system events (e.g., timer expiration). These events are acted on by the

VoiceXML interpreter itself, as specified by the VoiceXML document.

HTTP Server – (Document Server), sends out the VoiceXML script in response to the

VCX V7122 (the VoiceXML interpreter) request.

RADIUS Server – A centralized Authentication, Authorization and Accounting server for remote access users that communicate via the RADIUS protocol.

238 3Com VCX V7122 SIP VoIP Gateway User Manual

VoiceBrowser - (not shown in this diagram), responsible for the TTS (Text to Speech) and ASR (Automatic Speech Recognition) services (not supported in this version).

Proxy Server – Standard management tool for SIP networks. Performs essential control, administrative and managerial functions.

MediaPack – Analog media gateway, provides excellent voice quality and optimized packet voice, fax and modem streaming over the IP network.

Implementation

The VCX V7122 uses an embedded VoiceXML interpreter to interpret and execute standard

VoiceXML scripts, which are loaded from an outbound HTTP server and stored in the gateway’s volatile memory (RAM). The predefined VoiceXML scripts (up to 10 different scripts are supported) determine the development of the call according to the caller’s responses (DTMF digits) and AAA (Authentication, Authorization and Accounting) information exchanged with a RADIUS server. Interaction with the caller is conducted using a set of audio messages (stored in a single Voice Prompts file in the gateway’s flash/RAM memory) on the gateway’s side, and by pressing DTMF digits on the subscriber’s side.

Basic Calling Card IVR Scenario

Figure 86

Basic Call Scenario

3Com VCX V7122 SIP VoIP Gateway User Manual 239

Some messages have been omitted from the above drawing for the sake of clarity.

Call Flow Description

Figure 86 on the previous page depicts an example of a standard PSTN IP call (billingmodel: debit).

An incoming PSTN call with a published access number reaches the VCX V7122.

The VCX V7122 accepts the call (sends an Alert message).

The VCX V7122 searches its internal Tel IP Destination Number Manipulation table for the specific prefix. When it is detected, if the ‘Prefix to Add’ column corresponds to the predefined VXMLID parameter (in the example below: http), the VCX V7122 determines that the incoming call is an IVR call.

Figure 87

Basic ini File VXML Parameters

VXMLID = http

NumberMapTel2IP = 5394288,7,http://10.8.1.19/RadAAA01.txt

The VCX V7122 loads the VoiceXML file (RadAAA01.txt) from an outbound HTTP Server

(10.8.1.19) and stores it in its volatile memory. In Figure 86 it is assumed that the

VoiceXML already resides in the VCX V7122.

The VCX V7122 sends an Answer/Connect message.

The following steps are according to the supplied VXML script.

The VCX V7122 starts by playing an initial voice message. This message is composed of an opening greeting and an interactive menu asking the caller to choose one of the following options: 1 to make a call, 2 for help, 3 for operator service, 4 to exit.

After pressing the digit 1, the caller is immediately prompted to enter his account number

(usually the card number) and his password or PIN (personal identification number). The

VCX V7122 collects the input DTMF digits and sends an Authentication message to an outbound RADIUS server.

Only after the call is authenticated successfully (in this stage the RADIUS server returns the billing-model, in the above example: debit), the caller is asked to enter the number he wishes to reach, the final destination number. The gateway collects the input DTMF digits and sends an Authorization message to the RADIUS server.

The RADIUS server determines if the caller is authorized to proceed with the call and specifies the maximum duration of the call; the VCX V7122 conveys the call to the IP network and starts an internal timer.

One minute before credit is exhausted (see the VXML parameter finalalerttime on page

254 ); the finalalertaudio is played; finally, a minute later, the call is disconnected and the endaudio Voice Prompt is played.

240 3Com VCX V7122 SIP VoIP Gateway User Manual

After the conclusion of the call, the VCX V7122 sends an Accounting message to the

RADIUS server containing the call details (CDR) and prompts the user either to proceed with another call or to disconnect.

Operation and Configuration

To start working with the IVR system, follow these steps:

1 Install the VCX V7122 (see Chapter 3: Installing the VCX V7122 on page 25 and to

Chapter 4: Getting Started on page 35).

2 Create and load a Voice Prompts file to the VCX V7122 (see

Voice Prompts File on page 142 ).

3 Create the VXML scripts (see Voice XML Interpreter on page 247 ).

4 Install an HTTP Server, store the VXML scripts in it and provision the VCX V7122 relevant VoiceXML parameters.

5 In the VCX V7122 Destination Manipulation tables (PSTN

IP and IP PSTN), convert the predefined number to the HTTP server IP address and the VXML file name.

Assuming that the incoming number is 5394288, the IP address of the HTTP server is

10.8.1.19 and the name of the VXML file is RadAAA01.txt, then the ini file entry should look like: NumberMapTel2IP = 5394288,7,http://10.8.1.19/RadAAA01.txt. For the VCX V7122 to identify that the incoming call is designated to the IVR system, assign the string “http” to the

VXMLID parameter (see Table 50 for extended provisioning information).

To change a VXML script on-the-fly, access the Destination Number Manipulation table via the Embedded Web Server and replace the name of the file with the name of the new script (existing calls continue to run on the old script, while new calls run on the new script).

6 Install a RADIUS server, define the VCX V7122 in it (specify a common ‘SharedSecret’ parameter) and configure the VCX V7122 relevant RADIUS parameters (see Voice XML

Interpreter on page 247 ).

You are now ready to start using the Calling Card Application.

3Com VCX V7122 SIP VoIP Gateway User Manual 241

Configuration Parameters

Table 50

General VCX V7122 Parameters

ini File Field Name Valid Range and Description

NumberMapTel2IP

NumberMapIP2Tel

IVR Reference

Manipulates the destination number for Tel to IP calls.

NumberMapTel2IP = a,b,c,d,e,f,g a = Destination number prefix b = Number of stripped digits from the left, or (if brackets are used) from the right. A combination of both options is allowed. c = String to add as prefix, or (if brackets are used) as suffix. A combination of both options is allowed. d e

= Number of remaining digits from the right

= Number Plan used in RPID header f g

= Number Type used in RPID header

= Source number prefix

The ‘b’ to ‘f’ manipulations rules are applied if the called and calling numbers match the ‘a’ and ‘g’ conditions.

The manipulation rules are executed in the following order:

‘b’, ‘d’ and ‘c’.

Parameters can be skipped by using the sign "$$", for example:

NumberMapTel2IP=01,2,972,$$,0,0,$$

NumberMapTel2IP=03,(2),667,$$,0,0,22

Replaces the incoming destination number with a URL that indicates where the VXML script is located. a = Calling Card number b = The length of the Calling

Card number c = HTTP server IP + name of

VXML file (maximum length of string is 50 characters)

Note: To update the VoiceXML file, change the name of the file in the ‘c’ field.

Manipulate the destination number for IP to Tel calls.

NumberMapIP2Tel = a,b,c,d,e,f,g,h,i a b

= Destination number prefix

= Number of stripped digits from the left, or (if brackets are used) from the right. A combination of both options is allowed. f g h i c = String to add as prefix, or (if brackets are used) as suffix. A combination of both options is allowed. d e

= Number of remaining digits from the right

= Q.931 Number Plan

= Q.931 Number Type

= Source number prefix

= Not applicable, set to $$

= Source IP address

The ‘b’ to ‘f’ manipulation rules are applied if the called and calling numbers match the ‘a’, ‘g’ and ‘i’ conditions.

The manipulation rules are executed in the following order:

‘b’, ‘d’ and ‘c’.

Parameters can be skipped by using the sign "$$", for example:

NumberMapIP2Tel =01,2,972,$$,0,$$,034

NumberMapIP2Tel =03,(2),667,$$,$$,0,22,$$,10.13.77.8

Note: The Source IP address can include the “x” wildcard to represent single digits. For example: 10.8.8.xx represents all the addresses between 10.8.8.10 to

10.8.8.99.

Replaces the incoming destination number with a URL that indicates where the VXML script is located. a = Calling Card number b = The length of the Calling

Card number c = HTTP server IP + name of

VXML file (maximum length of string is 50 characters)

Note: To update the VoiceXML file, change the name of the file in the ‘c’ field.

242 3Com VCX V7122 SIP VoIP Gateway User Manual

ini File Field Name Valid Range and Description IVR Reference

VoicePromptsFileName The name (and path) of the file containing the Voice

Prompts definitions.

SaveConfiguration Set to 1 to store the Voice Prompts file in the non-volatile memory (file size mustn’t exceed 1Mb).

EnableVoiceStreaming 0 = Disable voice streaming (default).

1 = Enable voice streaming.

Set to 1 to enable the load of the

VoiceXML file from the HTTP server.

Table 51

Table J-1: VoiceXML Related Parameters

ini File Field Name Valid Range and Description

EnableVxml

[Enable VXML]

VxmlID

[VXML ID]

VxmlCollectDigits

0 = Disable the VXML feature (default).

1 = Enable the VXML feature.

According to this string, the VCX V7122 recognizes that an incoming call is to be diverted to the IVR system.

Note: Set to “http” (the “http” string must also appear in the manipulation table).

Determines the destination to which the VXML script reports the collected number

(username).

0 = The collected number (username) is sent for authentication to the RADIUS server

(default).

1 = The collected number is sent (in INFO message) to an Application / Proxy server.

The following RADIUS related parameters are described in Table 24 on page 94 :

EnableRADIUS

MaxRADIUSSessions

SharedSecret

RADIUSRetransmission

RADIUSTo

RADIUSAuthServerIP

RADIUSAuthPort

RADIUSAccServerIP

RADIUSAccPort

AAAIndications

RADIUSAccountingType

Supported RADIUS Attributes

Use Table 52 below for explanations on the RADIUS attributes contained in the communication packets transmitted between the VCX V7122 and a RADIUS Server.

3Com VCX V7122 SIP VoIP Gateway User Manual 243

Table 52

Supported RADIUS Attributes

Attribute

Number

Attribute

Name

Request Attributes

VSA

No.

Purpose

1

User-

Name

Value

Format

Sample

AAA

2

Account number or calling party number or blank

String up to

15 digits long

5421385747

Start Acc

Stop Acc

Authe

Autho

2

User-

Passwor d

Up to 15 digits

Autho

4

6

26

26

NAS-IP-

Address

Service-

Type h323incomingconf-id

1 h323remoteaddress

23

IP address of the requesting VCX V7122

Type of service requested

Start Acc

Stop Acc

Numeric 192.168.14.43

Authe

Autho

Start Acc

Stop Acc

H.323/SIP call identifier

Up to 32 octets

IP address of the remote gateway

Numeric

Start Acc

Stop Acc

26 h323conf-id

24 H.323/SIP

Up to 32 octets

Start Acc

Stop Acc

Authe

Autho

26 h323setuptime

25

Setup time in NTP format 1

String

Start Acc

Stop Acc

26

26 h323call-origin

26 h323call-type

27

The call’s originator:

Answering (IP) or

Originator (PSTN)

Protocol type or family used on this leg of the call

String

Answer,

Originate etc

Start Acc

Stop Acc

String VoIP

Start Acc

Stop Acc

26 h323connecttime

28

Connect time in NTP format

String

2

The values in column ‘AAA’ are as follows:

’Start Acc’ - Start Accounting

’Stop Acc’ - Stop Accounting

’Authe’ -

’Autho’ -

Authentication

Authorization

244 3Com VCX V7122 SIP VoIP Gateway User Manual

Calling-

Station-Id

Acct-

Status-

Type

Acct-

Delay-

Time

Acct-

Input-

Octets

Acct-

Output-

Octets

Acct-

Session-

Id

Acct-

Session-

Time

Acct-

Input-

Packets

Acct-

Output-

Packets

NAS-

Port-

Type

Attribute

Number

26

26

26

Attribute

Name

VSA

No.

Purpose

h323disconne ct-time h323disconne ct-cause

29

30 h323-gwid

33

Disconnect time in NTP format

Q.931 disconnect cause code

Name of the gateway

Value

Format

String

Numeric

String

Sample

SIPIDString

AAA

2

Start Acc

Stop Acc

30

Called-

Station-Id

31

40

41

42

43

44

46

47

48

61

Destination phone number

Calling Party Number

(ANI)

Account Request Type

(start or stop)

Stop Acc

Autho

Start Acc

String 5135672127

Authe

Autho

Numeric

1: start, 2: stop

Start Acc

Stop Acc

No. of seconds tried in sending a particular record

Number of octets received for that call duration

Number of octets sent for that call duration

Numeric 5

Numeric

Numeric

Start Acc

Stop Acc

A unique accounting identifier - match start & stop

For how many seconds the user received the service

String 34832

Start Acc

Stop Acc

Numeric

Number of packets received during the call

Numeric

Number of packets sent during the call

Numeric

VCX V7122 physical port type on which the call is active

String

0:

Asynchronous

Start Acc

Stop Acc

Authe

Autho

3Com VCX V7122 SIP VoIP Gateway User Manual 245

Attribute

Number

Attribute

Name

Response Attributes

VSA

No.

Purpose

26

26 h323crdit-time h323returncode

102

103

Number of seconds for which the call is authorized

The reason for failing authentication (0 = ok, other number failed)

26

44 h323billingmodel

Acct-

Session-

Id

109

Type of billing service for a specific call

A unique accounting identifier – match start & stop

Value

Format

Numeric

Numeric

Sample

Numeric 360

0 Request accepted

1:debit/prepai d

String

AAA

2

Autho

Authe

Autho

Stop Acc

Authe

RADIUS Server Messages

In Figure 88 , Figure 89 , and Figure 90 non-standard parameters are preceded with brackets.

Authentication

Figure 88

Authentication Example

Access-Request (116) user-name = 111 user-password = (encrypted) nas-ip-address = 212.179.22.213 nas-port-type = 0 calling-station-id = 202

// Authentication non-standard parameters:

(4923 24) h323-conf-id = 02102944 600a1899 3fd61009 0e2f3cc5

In the Access-Accept response, the RADIUS server sends the billing model:

(4923 109) h323-billing-model = 1/0

The billing model is a non-standard parameter and can be one of the following:

1 = credit (prepaid)

0 = debit (postpaid)

When a billing model isn’t received, the VCX V7122 assumes a prepaid billing model (1).

246 3Com VCX V7122 SIP VoIP Gateway User Manual

Authorization

Figure 89

Authorization Example

Access-Request (121) user-name = 111 user-password = (encrypted) nas-ip-address = 212.179.22.213 nas-port-type = 0 called-station-id = 201 calling-station-id = 202

// Authorization non-standard parameters:

(4923 24) h323-conf-id = 02102944 600a1899 3fd61009 0e2f3cc5

In the Access-Accept response, the RADIUS server sends the credit time:

(4923 102) h323-credit-time = 6000

The credit-time is a non-standard parameter which is measured in seconds.

Accounting

Figure 90

Accounting Example

Accounting-Request (361) user-name = 111 acct-session-id = 1 nas-ip-address = 212.179.22.213 nas-port-type = 0 acct-status-type = 2 acct-input-octets = 4841 acct-output-octets = 8800 acct-session-time = 1 acct-input-packets = 122 acct-output-packets = 220 called-station-id = 201 calling-station-id = 202

// Accounting non-standard parameters:

(4923 33) h323-gw-id =

(4923 23) h323-remote-address = 212.179.22.214

(4923 1) h323-ivr-out = h323-incoming-conf-id:02102944 600a1899

3fd61009 0e2f3cc5

(4923 30) h323-disconnect-cause = 22 (0x16)

(4923 27) h323-call-type = VOIP

(4923 26) h323-call-origin = Originate

(4923 24) h323-conf-id = 02102944 600a1899 3fd61009 0e2f3cc5

Voice XML Interpreter

VoiceXML (Voice Extensible Markup Language) is designed for creating audio dialogs that feature synthesized speech, digitized audio, recognition of speech and DTMF inputs, recording of spoken input, telephony, and mixed initiative conversations. Its major goal is to bring the advantages of Web-based development and content delivery to interactive voice response applications.

3Com VCX V7122 SIP VoIP Gateway User Manual 247

Features

Supports DTMF recognition.

Executes audio dialogs between the gateway and a user, supporting mixed initiative applications.

Audio prompt recording (currently not supported).

Transfer Support - Using the <Transfer> element, the VCX V7122 places a call to different destinations.

JavaScript Expression Support - Supports ECMA script specification 3.0 (standard

ECMA-262).

Supports definition of the end-dial key (‘*’ or ‘#’) that terminates the DTMF collection.

To define whether * or # are used to terminate the DTMF collection, add the following line to each script:

<property name="EndDialKey" value="*">,

OR

<property name="EndDialKey" value="#">

Supports number concatenation, enabling number modification per VXML script.

<filled>

<assign name="user_passwd" expr="'domain'+user_passwd + '.com'"/>

<return namelist="user_account_num user_passwd"/>

</filled>

The user_passwd parameter (that initially contained the user password collected from the user) is being assigned the value ‘domain'+user_passwd + '.com’.

Calling number (recived form SIP incoming call) can optionally be used for authentication instead of the user name. form id="GetCallerId">

<log label="VXML--> getting caller id from SIP..." />

<object name="FCallerId" classid="builtin://com.audiocodes.ulp.input">

<filled>

<if cond="FCallerId.Result != 'fail'">

<assign name="CallerId" expr="FCallerId.Result" />

<goto next="#PerformAuthenWithoutUserName" />

<else />

<goto next="#PerformAuthen" />

</if>

</filled>

</object>

</form>

248 3Com VCX V7122 SIP VoIP Gateway User Manual

Supported Elements and Attributes

Table 53

VoiceXML Supported Elements and Attributes

Element

<assign>

<audio>

<block>

Element’s

Description

Assign value to variable

Plays an audio clip within a prompt

A container of (noninteractive) executable code

Catch an event

Parameters Parameter’s Description

name expr expr

The name of the modified variable

The new value of the variable.

Dynamically determines the URI to fetch by evaluating this ECMAScript expression.

<catch>

event count cond

The event or events to catch.

The occurrence of the event (default 1).

An expression which must evaluate to true after conversion to Boolean in order for the event to be caught, (defaults true).

The DTMF sequence for this choice. <choice> Defines a menu item

dtmf accept next expr

Override the setting for accept in <menu> for this particular choice.

URI of the next dialog or doc.

Specifies an expression to evaluate as a URI to transition to instead of specifying a next.

event eventexpr message

Specify an event to be thrown instead of specifying a next.

An ECMAScript expression evaluating to the name of the event to be thrown.

A message string providing additional context about the event being thrown.

messageexpr

An ECMAScript expression evaluating to the message string.

<clear> Clear one or more form item variables

<disconnect> Disconnect a session

namelist

The list of variables to be reset.

Supported

<else>

<elseif>

<error>

Used in <if> elements

Used in <if> elements

Catch an error event count The event count

3Com VCX V7122 SIP VoIP Gateway User Manual 249

Element

Element’s

Description

Parameters Parameter’s Description Supported

<exit>

<field>

Exit a session

Declares an input field in a form

cond expr namelist name expr

<filled>

<form>

<goto>

modal

An action executed when fields are filled

mode namelist

A dialog for presenting information and collecting data

id scope

Go to another dialog in the same or different document

next expr nextitem

<grammar> Specify a speech recognition or DTMF grammar

version xml:lang cond type slot mode root tag-format

An optional condition to test to see if the event is caught by this element. Defaults to true.

A return expression.

Variable names to be returned to interpreter context.

The form item variable in the dialog scope that holds the result.

The initial value of the form item variable;

An expression that must evaluate to true after conversion to Boolean in order for the form item to be visited.

The type of field.

The name of the grammar slot used to populate the variable. (NOT SUPPORTED)

If this is false (the default) all active grammars are turned on while collecting this field.

Either all (the default), or any.

The input items to trigger on.

The name of the form.

The default scope of the form’s grammars.

The URI to which to transition.

An ECMAScript expression that yields the URI.

The name of the next form item to visit in the current form.

Defines the version of the grammar. (NOT

SUPPORTED)

The language identifier of the contained or referenced grammar. (NOT SUPPORTED)

Defines the mode of the contained or referenced grammar following the modes of the W3C Speech

Recognition Grammar Specification [SRGS].

Defined values are "voice" and "dtmf" for DTMF input. (NOT SUPPORTED)

Defines the public rule which acts as the root rule of the grammar.

Defines the tag content format for all tags within the grammar. (NOT SUPPORTED)

250 3Com VCX V7122 SIP VoIP Gateway User Manual

Element

<help>

Element’s

Description

Catch a help event

Parameters Parameter’s Description

base src scope type count cond

Supported

Declares the base URI from which relative URIs are resolved. (NOT SUPPORTED)

The URI specifying the location of the grammar and optionally a rulename within that grammar, if it is external.

Either "document", which makes the grammar active in all dialogs of the current document or "dialog", to make the grammar active throughout the current form.

The media type of the grammar.

The event count.

An optional condition to test to see if the event is caught by this element.

<if> Simple conditional logic

<link> Specify a transition common to all dialogs in the link’s

next

scope

expr

The URI to go to.

event eventexpr message

Like next, except that the URI is dynamically determined.

The event to throw when the user matches one of the link grammars.

An ECMAScript expression evaluating to the name of the event to throw when the user matches one of the link grammars.

A message string providing additional context about the event being thrown.

<log>

<menu>

<noinput>

messageexpr

An ECMAScript expression evaluating to the message string.

dtmf

The DTMF sequence for this link.

Generate a debug message

label expr

A dialog for choosing amongst alternative destinations

id scope

A string which may be used.

An ECMAscript expression evaluating to a string.

The identifier of the menu.

Catch a noinput

Dtmf count

The menu’s grammar scope.

When set to true, the first nine choices that have not explicitly specified a value for the dtmf attribute are given the implicit ones "1", "2", etc.

The event count.

3Com VCX V7122 SIP VoIP Gateway User Manual 251

Element

Element’s

Description

event

<nomatch> Catch a nomatch event

cond count cond

<object> Interact with a custom extension

name

<param> Parameter in

<object> or

<subdialog>

<property> Control implementation platform settings.

<record> Record an audio sample

Parameters Parameter’s Description

expr cond classid codebase codetype data type

Name expr value valuetype type name expr

An optional condition to test to see if the event is caught by this element.

The event count.

An optional condition to test to see if the event is caught by this element.

When the object is evaluated, it sets this variable to an ECMAScript value whose type is defined by the object.

The initial value of the form item variable.

An expression that must evaluate to true after conversion to Boolean in order for the form item to be visited.

Supported

The URI specifying the location of the object’s implementation.

The base path used to resolve relative URIs specified by classid, data, and archive. (NOT

SUPPORTED)

The content type of data expected when loading the object specified by classid. (NOT SUPPORTED)

The URI specifying the location of the object’s data.

The content type of the data specified by the data attribute. (NOT SUPPORTED)

The name to be associated with this parameter when the object or sub-dialog is invoked.

An expression that computes the value associated with name.

Associates a literal string value with name.

One of data or ref, by default data; used to indicate to an object if the value associated with name is data or a URI (ref).

The media type of the result provided by a URI if the valuetype is ref;

For a list of available properties, see Table 54 on page 255 .

The input item variable that holds the recording.

The initial value of the form item variable.

252 3Com VCX V7122 SIP VoIP Gateway User Manual

Element

Element’s

Description

Parameters Parameter’s Description Supported

cond modal beep finalsilence dtmfterm type

An expression that must evaluate to true after conversion to Boolean in order for the form item to be visited.

If this is true all non-local speech and DTMF grammars are not active while making the recording.

If true, a tone is emitted just prior to recording. (NOT

SUPPORTED)

The interval of silence that indicates end of speech.

(NOT SUPPORTED)

If true, any DTMF keypress not matched by an active grammar is treated as a match of an active local DTMF grammar. (NOT SUPPORTED)

The media format of the resulting recording.

<return> Return from a subdialog.

event

Return, and then throw this event.

eventexpr

Return, and then throw the event to which this

ECMAScript expression evaluates.

message

A message string providing additional context about the event being thrown.

messageexpr

An ECMAScript expression evaluating to the message string.

namelist

<subdialog> Invoke another dialog as a subdialog of the current one

name

Expr

Variable names to be returned to calling dialog.

The result returned from the sub-dialog,

cond namelist src

The initial value of the form item variable.

An expression that must evaluate to true after conversion to Boolean in order for the form item to be visited.

The list of variables to submit. The default is to submit no variables. If a namelist is supplied, it may contain individual variable references which are submitted with the same qualification used in the namelist. Declared VoiceXML and ECMAScript variables can be referenced.

The URI of the sub-dialog.

srcexpr method

An ECMAScript expression yielding the URI of the sub-dialog.

See Section 5.3.8.

3Com VCX V7122 SIP VoIP Gateway User Manual 253

Element

<submit>

Element’s

Description

Submit values to a document server

Parameters Parameter’s Description

next expr namelist method event eventexpr cond

The URI reference.

Like next, except that the URI reference is dynamically determined.

The list of variables to submit.

The request method: get (the default) or post.

The event being thrown.

An ECMAScript expression evaluating to the name of the event being thrown.

message

A message string providing additional context about the event being thrown.

messageexpr

An ECMAScript expression evaluating to the message string.

<transfer>

Transfer the caller to another destination

name expr

Stores the outcome of a bridge transfer attempt.

The initial value of the form item variable.

An expression that must evaluate to true in order for the form item to be visited.

Supported

<var> Declare a variable

dest destexpr bridge

The URI of the destination (telephone, IP telephony address).

An ECMAScript expression yielding the URI of the destination.

Determines whether the platform remains in the connection with the caller and callee.

transferaudio

The URI of audio source to play while the transfer attempt is in progress (before far-end answer).

(NOT SUPPORTED)

endaudio

An internal Voice Prompt ID that determines the VP that is played when the maximal time allowed for a call expires.

finalalertaudio

An internal Voice Prompt ID that determines the VP that is played finalalerttime seconds before the maximal time allowed for a call expires.

finalalerttime

Determines how many seconds before the maximal time allowed for a call expires, the finalalertaudio VP is played.

name

The name of the variable that holds the result.

expr

The initial value of the variable.

254 3Com VCX V7122 SIP VoIP Gateway User Manual

Element

<vxml>

Element’s

Description

Parameters Parameter’s Description

Top-level element in each VoiceXML document

version xmlns

The version of VoiceXML of this document

(required). (NOT SUPPORTED)

xml:base xml:lang application

The designated namespace for VoiceXML

(required). (NOT SUPPORTED)

The base URI for this document as defined in [XML-

BASE]. (NOT SUPPORTED)

The language identifier for this document as defined in [RFC3066]. (NOT SUPPORTED)

The URI of this document’s application root document, if any. (NOT SUPPORTED)

Supported

Table 54

VoiceXML Supported Properties

Property Name Property description Default

Value

FetchTimeout

The maximum time (in seconds) to wait from the time the script is fetched.

5 seconds

TimeoutExpirationTime The maximum time to wait for the first digit in the user’s input. 5 seconds

EndDialKey The user’s input that terminates the current collection. #

Provided Calling Card System

Voice Prompts

0 “Welcome to the calling card service”.

1 “To make a call press 1, for help press 2, for operator assistance press 3, to exit press 4.

2 “Invalid selection. Please try again”.

3 “Unable to recognize your entry. Please try again” .

4 “Please call again later”.

5 “Please enter your account number followed by the pound key”.

6 “Please enter your password followed by the pound key”.

7 “Please wait while we verify your account number and password”.

8 “Your account number and password do not match”.

9 “We are having technical difficulties, please call again later”.

10 “Please enter the number that you wish to call followed by the pound key”.

11 “The number you are calling is busy”.

12 “The number you are calling is not answering, please call again later”.

13 “The person you called has hung up”.

3Com VCX V7122 SIP VoIP Gateway User Manual 255

14 “We are unable to complete your call”.

15 “Transferring your call”.

16 “Thank you for using the calling card service”.

17 “The menu is currently empty”.

18 “Operator assistance is currently unavailable”.

19 “You are unauthorized to access the number you are attempting to call”.

20 Final alert (one beep at 640 Hz for 0.5 second duration).

21 Time of call expired (one beep at 640 Hz for 1.5 second duration).

256 3Com VCX V7122 SIP VoIP Gateway User Manual

VXML Flow Chart

Figure 91

VXML Script Opening Menu

Play VP 0

Start

Play VP 1

DTMF Pressed?

Yes

No

Play VP 3

Yes

Pressed 1?

Go To 1

No

Yes

Pressed 2?

No

Go To 2

Go To 3

Yes

Pressed 3?

No

Yes

Pressed 4?

Go To Start

Go To 4

No

No Match

Counter = x?

No

Play VP 2

Yes

Play VP 4

Go To End

Go To Start

3Com VCX V7122 SIP VoIP Gateway User Manual 257

Figure 92

VXML Script Option 1, Make a Call

1

Play VP 5

DTMF Pressed?

Yes

Play VP 6

No

Play VP 3

Go To 1

DTMF Pressed?

No

Yes

Play VP 7

Play VP 3

Waits for

Authentication

Yes

Authentication

Finished?

Yes

Authentication

OK?

No

Go To Transfer

Yes

Play VP 4

Go To End

No Match

Counter = x?

No

Play VP 7

No

Go To 1

No

258 3Com VCX V7122 SIP VoIP Gateway User Manual

Figure 93

VXML Script, Call Transfer Procedure

Transfer

Play VP 10

DTMF Pressed?

No

Yes

Play VP 15

Play VP 3

Go to Transfer

Call for transfer service

Yes

No

Debit?

No

Transfer Failed?

Yes

Play either VP

11, 12 or 13 according to

Transfer failure reason

Final alert time expired?

No

Go to End

Yes

Play VP 20

Time for call expired?

Yes

Play VP 21

End

No

3Com VCX V7122 SIP VoIP Gateway User Manual 259

Figure 94

VXML Script, Options 2, 3 and 4

2

Play VP 17

Go To Start

3

Transfer Call To an Operator

End

4

Play VP 16

End

260 3Com VCX V7122 SIP VoIP Gateway User Manual

Figure 95

VXML Script, Call Termination

End

VXML Script

Terminated?

No

Go To Start

Yes

Play VP 16

Free Resources

End

VXML Script Example

Figure 96

VXML Script Example

<?xml version="1.0" encoding="UTF-8"?>

<vxml version="1.0" application="http://phoenix1.iperia.com:8080/sa3/jsp/sa.jsp">

<var name="AAStatus" expr="0"/>

<form id="main_mc">

<log label="starting main form"/>

<prompt>

<audio src="/0.wav">

wellcome to the pre-paid call service

</audio>

</prompt>

<goto next="#main_menu"/>

</block>

</form>

<form id="main_menu">

<log label="starting main_menu form"/>

<field name="option" type="number">

<dtmf>

1 | 2 | 3 | 4

</dtmf>

<prompt bargein="true">

<audio src="/1.wav">

for making a call press 1

for help press 2

for human service press 3

to exit press 4

3Com VCX V7122 SIP VoIP Gateway User Manual 261

</audio>

</prompt>

<nomatch>

<log label="try again interupt"/>

<prompt>

<audio src="/2.wav"> that is an invalid selection. Please try again

</audio>

</prompt>

</nomatch>

<noinput>

<log label="no try interupt"/>

<prompt>

<audio src="/3.wav"> we did not get your input. Please try again

</audio>

</prompt>

</noinput>

<catch event="noinput nomatch" count="4">

<prompt>

<audio src="/4.wav">

</audio>

</prompt>

<log label="please try again later"/>

<goto next="#disconnect"/>

</catch>

<filled>

<if cond="option == '1'">

<else/>

</if>

<elseif cond="option =='2'"/>

<prompt>

<audio src="/17.wav">

</audio>

</prompt>

<elseif cond="option =='3'"/>

<elseif cond="option == '4'"/>

<goto next="#disconnect"/>

</if>

</filled>

</field>

</form>

<log label="starting get user info form" cond="'1' == '1'"/>

<field name="user_account_num" type="digits">

262 3Com VCX V7122 SIP VoIP Gateway User Manual

bargein="true">

<audio src="/5.wav">

</audio>

</prompt>

</field>

<field name="user_passwd" type="digits"> bargein="true">

<audio src="/6.wav">

</audio>

</prompt>

</field>

<noinput>

<prompt>

<audio src="/3.wav"> we did not get your input. Please try again

</audio>

</prompt>

</noinput>

<catch event="noinput nomatch" count="3">

<prompt>

<audio src="/4.wav">

</audio>

</prompt>

<goto next="#disconnect"/>

</catch>

<filled>

<return namelist="user_account_num user_passwd"/>

</filled>

</form>

<log label="starting get called party telephone form" cond="'1' == '1'"/>

<field name="dest_number" type="digits"> bargein="true">

<audio src="/10.wav">

</audio>

</prompt>

<noinput>

<prompt>

<audio src="/3.wav">

</audio>

</prompt>

</noinput>

<catch event="noinput nomatch" count="3">

<goto next="#disconnect"/>

</catch>

3Com VCX V7122 SIP VoIP Gateway User Manual 263

<filled>

<return namelist="dest_number"/>

</filled>

</field>

</form>

<log label="performing help transfer"/>

<transfer name="mycall" destexpr="201" bridge="true">

<filled>

</filled>

</transfer>

</form>

<filled>

<prompt>

<audio src="/7.wav"> please wait while your account and pin numbers are cheked

</audio>

</prompt>

</filled>

</subdialog>

<nomatch>

<prompt>

<audio src="/8.wav"> your account and password did not match.

</audio>

</prompt>

</nomatch>

<catch event="nomatch" count="4">

<goto next="#disconnect"/>

</catch>

<filled>

264 3Com VCX V7122 SIP VoIP Gateway User Manual

</filled>

</object>

</form>

<log label="starting transfer form"/>

</subdialog>

<nomatch>

<prompt>

<audio src="/9.wav"/>

</prompt>

</nomatch>

<catch event="nomatch" count="3">

<goto next="#disconnect"/>

</catch>

<filled>

<prompt>

<audio src="/15.wav">

calling

</audio>

</prompt>

</filled>

</object>

<transfer name="mycall" destexpr="Call.dest_number" bridge="true" endaudio="7" finalalertaudio="4" finalalerttime="10">

<filled>

<prompt>

<audio src="/14.wav">

</audio>

</prompt>

<elseif cond="mycall.Result=='busy'"/>

<prompt>

<audio src="/11.wav">

</audio>

</prompt>

<elseif cond="mycall.Result=='maxtime'"/>

<prompt>

3Com VCX V7122 SIP VoIP Gateway User Manual 265

<audio src="/7.wav">

</audio>

</prompt>

<else/>

</if>

</filled>

</transfer>

</form>

<prompt>

goodbye

</audio>

</prompt>

<exit/>

</block>

</form>

</vxml>

266 3Com VCX V7122 SIP VoIP Gateway User Manual

Was this manual useful for you? yes no
Thank you for your participation!

* Your assessment is very important for improving the work of artificial intelligence, which forms the content of this project

Download PDF

advertisement