Alvarion SIP R2J User Manual

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Voice Gateways

System Manual

SW Version: SIP R2J July 2007 P/N: 214612

Document History

Topic Description Date Issued

1.2.8 Connectors The connectors' specifications were updated.

2.3 Installation and

Commissioning

2.4 Notes on Using the Voice

Gateway (VG) in Alvarion's Systems

3.2 Accessing the Web

Configuration Server

3.5.1 WAN Status Page Bridge Status new display. Version 1.0 November

3.6.5 VLAN Configuration

Example 2

3.7 Telephone Menu The Telephone Menu was modified to

3.12 Web Configuration Server's

Parameters Summary

Commissioning description added. Version 1.0 November

Using the VG with BreezeMAX and BreezeACCESS VL

Detailed instructions for accessing the web server via the WAN and via th e LAN were added.

The example was updated. Version 1.0 November

include the H323 Configuration page. Some of the parameters defaults have

been changed (also in their respective paragraphs)

Version 1.0 November, 2005

Version 1.0 November

2.3 Installation and

Commissioning

3.6.1 VLAN Tagging Page Default VLAN ID for LAN was removed,

3.6.6 VLAN Configuration

Example

3.7.1 SIP/H323 Configuration

Page

3.7.1.1 Codecs and Fax

Configuration

3.7.2 SIP Extensions Page SIP notify messages option added, to

3.7.4 STUN Client Page The STUN Client submenu was added to

Updated for VG management via WAN only.

and explanations were added. An additional example was provided Version 1.1 February

The page was updated (support of Message Waiting)

Jitter Buffer options added. Version 1.1 February

keep the SIP proxy connection alive.

the Telephone menu

Version 1.1 February 2006

2006 Version 1.1 February

2006

2006 Version 1.1 February

2006

ii Voice Gateways System Manual

Topic Description Date Issued

Legal Rights

3.7.6 Line Configuration Page The Line Configuration submenu was added to the Telephone menu

3.9.5 RTP Stats Page The RTP Statistics submenu was added to the System menu

3.10 Upgrade Page Download option from an HTTP server was added.

Version 1.1 February 2006

3.12 Logout Page Logout option added. Version 1.1 February

2006

General No H323 support Version 1.2 August

2006

2.3 (Installation and

Commissioning) and 3.2

2006 (Accessing the Web Configuration Server)

2.3 Installation and Commissioning

3.5.1 WAN Status Page and

3.5.2 WAN Configuration Page

Added access to the VG via LAN (in addition to WAN) using the WAN IP

Broadcast Limit and Multicast Limit deleted.

Version R2H276

December 2006

Version R2H276

December 2006

3.7.1 SIP/H323 Configuration Page

3.7.1.1 Codecs and Fax Configuration

Default dialplan changed Version R2H276

December 2006

Optional use of G711A/U codex enabled Version R2H276

December 2006

3.7.7.1 Hotline Hotline option added to the dialplan Version R2H276

December 2006

3.7.7.2 Adding/Removing Prefixes

Appendix C. New Features Added appendix C with a list of new

3.6.4 VLAN Configuration Example 1

Automatic addition and removal of prefixes options added to the dialplan

Version R2H276

December 2006

Version R2J259

features for R2J

May 2007

Step 7. LAN: NO (fixed) Version R2J259

May 2007

Voice Gateways System Manual iii

Legal Rights

The material contained herein is proprietary, privileged, and confidential and

owned by Alvarion or its third party licensors. No disclosure thereof shall be made

to third parties without the express written permission of Alvarion Ltd.

Alvarion Ltd. reserves the right to alter the equipment specifications and

descriptions in this publication without prior notice. No part of this publication

shall be deemed to be part of any contract or warranty unless specifically

incorporated by reference into such contract or warranty.

Trade Names

Alvarion®, BreezeCOM®, WALKair®, WALKnet®, BreezeNET®, BreezeACCESS®,

BreezeMANAGE

BreezeLITE

products and/or services referenced here in are either registered trademarks,

trademarks or service marks of Alvarion Ltd.

™

, BreezeLINK®, BreezeCONFIG™, BreezeMAX™, AlvariSTAR™,

™

, MGW™, eMGW™, BreezeCRAFT™, AlvariCRAFT™ and/or other

All other names are or may be the trademarks of their respective owners.

Statement of Conditions

The information contained in this manual is subject to change without notice.

Alvarion Ltd. shall not be liable for errors contained herein or for incidental or

consequential damages in connection with the furnishing, performance, or use of

this manual or equipment supplied with it.

Warranties and Disclaimers

All Alvarion Ltd. ("Alvarion") products purchased from Alvarion or through any of

Alvarion's authorized resellers are subject to the following warranty and product

liability terms and conditions.

Exclusive Warranty

(a) Alvarion warrants that the Product hardware it supplies and the tangible

media on which any software is installed, under normal use and conditions, will

be free from significant defects in materials and workmanship for a period of

fourteen (14) months from the date of shipment of a given Product to Purchaser

(the "Warranty Period"). Alvarion will, at its sole option and as Purchaser's sole

remedy, repair or replace any defective Product in accordance with Alvarion'

standard R&R procedure.

(b) With respect to the Firmware, Alvarion warrants the correct functionality

according to the attached documentation, for a period of fourteen (14) month from

iv Voice Gateways System Manual

Disclaimer

Legal Rights

invoice date (the "Warranty Period")". During the Warranty Period, Alvarion may

release to its Customers firmware updates, which include additional performance

improvements and/or bug fixes, upon availability (the "Warranty"). Bug fixes,

temporary patches and/or workarounds may be supplied as Firmware updates.

Additional hardware, if required, to install or use Firmware updates must be

purchased by the Customer. Alvarion will be obligated to support solely the two (2)

most recent Software major releases.

ALVARION SHALL NOT BE LIABLE UNDER THIS WARRANTY IF ITS TESTING

AND EXAMINATION DISCLOSE THAT THE ALLEGED DEFECT IN THE PRODUCT

DOES NOT EXIST OR WAS CAUSED BY PURCHASER'S OR ANY THIRD

PERSON'S MISUSE, NEGLIGENCE, IMPROPER INSTALLATION OR IMPROPER

TESTING, UNAUTHORIZED ATTEMPTS TO REPAIR, OR ANY OTHER CAUSE

BEYOND THE RANGE OF THE INTENDED USE, OR BY ACCIDENT, FIRE,

LIGHTNING OR OTHER HAZARD.

(a) The Software is sold on an "AS IS" basis. Alvarion, its affiliates or its licensors

MAKE NO WARRANTIES, WHATSOEVER, WHETHER EXPRESS OR IMPLIED,

WITH RESPECT TO THE SOFTWARE AND THE ACCOMPANYING

DOCUMENTATION. ALVARION SPECIFICALLY DISCLAIMS ALL IMPLIED

WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR

PURPOSE AND NON-INFRINGEMENT WITH RESPECT TO THE SOFTWARE.

UNITS OF PRODUCT (INCLUDING ALL THE SOFTWARE) DELIVERED TO

PURCHASER HEREUNDER ARE NOT FAULT-TOLERANT AND ARE NOT

DESIGNED, MANUFACTURED OR INTENDED FOR USE OR RESALE IN

APPLICATIONS WHERE THE FAILURE, MALFUNCTION OR INACCURACY OF

PRODUCTS CARRIES A RISK OF DEATH OR BODILY INJURY OR SEVERE

PHYSICAL OR ENVIRONMENTAL DAMAGE ("HIGH RISK ACTIVITIES"). HIGH

RISK ACTIVITIES MAY INCLUDE, BUT ARE NOT LIMITED TO, USE AS PART OF

ON-LINE CONTROL SYSTEMS IN HAZARDOUS ENVIRONMENTS REQUIRING

FAIL-SAFE PERFORMANCE, SUCH AS IN THE OPERATION OF NUCLEAR

FACILITIES, AIRCRAFT NAVIGATION OR COMMUNICATION SYSTEMS, AIR

TRAFFIC CONTROL, LIFE SUPPORT MACHINES, WEAPONS SYSTEMS OR

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(b) PURCHASER'S SOLE REMEDY FOR BREACH OF THE EXPRESS

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FULLEST EXTENT ALLOWED BY LAW, THE WARRANTIES AND REMEDIES SET

FORTH IN THIS AGREEMENT ARE EXCLUSIVE AND IN LIEU OF ALL OTHER

Voice Gateways System Manual v

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WARRANTIES OR CONDITIONS, EXPRESS OR IMPLIED, EITHER IN FACT OR BY

OPERATION OF LAW, STATUTORY OR OTHERWISE, INCLUDING BUT NOT

LIMITED TO WARRANTIES, TERMS OR CONDITIONS OF MERCHANTABILITY,

FITNESS FOR A PARTICULAR PURPOSE, SATISFACTORY QUALITY,

CORRESPONDENCE WITH DESCRIPTION, NON-INFRINGEMENT, AND

ACCURACY OF INFORMATION GENERATED. ALL OF WHICH ARE EXPRESSLY

DISCLAIMED. ALVARION' WARRANTIES HEREIN RUN ONLY TO PURCHASER,

AND ARE NOT EXTENDED TO ANY THIRD PARTIES. ALVARION NEITHER

ASSUMES NOR AUTHORIZES ANY OTHER PERSON TO ASSUME FOR IT ANY

OTHER LIABILITY IN CONNECTION WITH THE SALE, INSTALLATION,

MAINTENANCE OR USE OF ITS PRODUCTS.

Limitation of Liability

(a) ALVARION SHALL NOT BE LIABLE TO THE PURCHASER OR TO ANY THIRD

PARTY, FOR ANY LOSS OF PROFITS, LOSS OF USE, INTERRUPTION OF

BUSINESS OR FOR ANY INDIRECT, SPECIAL, INCIDENTAL, PUNITIVE OR

CONSEQUENTIAL DAMAGES OF ANY KIND, WHETHER ARISING UNDER

BREACH OF CONTRACT, TORT (INCLUDING NEGLIGENCE), STRICT LIABILITY

OR OTHERWISE AND WHETHER BASED ON THIS AGREEMENT OR

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THE LIABILITY FOR DAMAGES HEREUNDER OF ALVARION OR ITS EMPLOYEES

OR AGENTS EXCEED THE PURCHASE PRICE PAID FOR THE PRODUCT BY

PURCHASER, NOR SHALL THE AGGREGATE LIABILITY FOR DAMAGES TO ALL

PARTIES REGARDING ANY PRODUCT EXCEED THE PURCHASE PRICE PAID

FOR THAT PRODUCT BY THAT PARTY (EXCEPT IN THE CASE OF A BREACH OF

A PARTY'S CONFIDENTIALITY OBLIGATIONS).

Disposal of Electronic and Electrical Waste

Pursuant to the WEEE EU Directive electronic and electrical waste must not be disposed of with unsorted waste. Please contact your local recycling authority for disposal of this product.

vi Voice Gateways System Manual

Important Notice

This user manual is delivered subject to the following conditions and restrictions:

 This manual contains proprietary information belonging to Alvarion Ltd. Such

information is supplied solely for the purpose of assisting properly authorized

users of the respective Alvarion products.

 No part of its contents may be used for any other purpose, disclosed to any

person or firm or reproduced by any means, electronic and mechanical,

without the express prior written permission of Alvarion Ltd.

 The text and graphics are for the purpose of illustration and reference only.

The specifications on which they are based are subject to change without

notice.

 The software described in this document is furnished under a license. The

software may be used or copied only in accordance with the terms of that

license.

Legal Rights

 Information in this document is subject to change without notice. Corporate

and individual names and data used in examples herein are fictitious unless

otherwise noted.

 Alvarion Ltd. reserves the right to alter the equipment specifications and

descriptions in this publication without prior notice. No part of this

publication shall be deemed to be part of any contract or warranty unless

specifically incorporated by reference into such contract or warranty.

 The information contained herein is merely descriptive in nature, and does not

constitute an offer for the sale of the product described herein.

 Any changes or modifications of equipment, including opening of the

equipment not expressly approved by Alvarion Ltd. will void equipment

warranty and any repair thereafter shall be charged for. It could also void the

user's authority to operate the equipment.

Voice Gateways System Manual vii

About This Manual

This manual describes Alvarion's Voice Gateway units and how to install, operate

and manage them. Version R2J supports SIP only.

This manual is intended for technicians responsible for installing, setting up and

operating the Voice Gateway, and for system administrators responsible for

managing the Voice Gateways.

This manual contains the following chapters and appendices:

 Chapter 1 - System Description: Describes the Voice Gateway and its

functionality.

 Chapter 2 - Installation: Describes how to install the Voice Gateway and

connect it to the SU and to the user's equipment.

 Chapter 3 - Using the Web Configuration Server: Describes how to use the

Web Configuration Server for configuring parameters and checking system

status.

 Appendix A - Internal Class 5 Services: Describes the internal Class-5 services

that are supported by the Gateway.

 Appendix B - Default Telephony Parameters: Describe the default values for

some telephony parameters, including signals/tones parameters, CID

parameters and line impedance.

 Appendix C - New Features: Lists and explains new features and parameters

configurable in the ini file.

 Glossary: Provides definitions of various terms used in the manual.

Chapter 1 - System Description

1.1 Introducing the Voice Gateway....................................................................................2

1.2 Specifications................................................................................................................3

1.2.1 Telephony and Fax Services................................................................................3

1.2.2 Security................................................................................................................3

1.2.3 Voice Quality........................................................................................................4

1.2.4 Configuration and Management...........................................................................4

1.2.5 Bridge Functionality..............................................................................................4

1.2.6 Mechanical...........................................................................................................5

1.2.7 Electrical...............................................................................................................5

1.2.8 Connectors...........................................................................................................5

1.2.9 Regulatory Standards Compliance ......................................................................6

1.2.10 Environmental......................................................................................................6

Chapter 2 - Installation

2.1 Installation Requirements ............................................................................................8

2.1.1 Packing List..........................................................................................................8

2.1.2 Additional Installation Requirements....................................................................8

2.2 Front and Rear Panel Components.............................................................................9

2.2.1 Connectors...........................................................................................................9

2.2.2 Reset to Factory Default Configuration................................................................9

2.2.3 LEDs ..................................................................................................................10

2.3 Installation and Commissioning................................................................................11

2.4 Notes on Using the Voice Gateways in Alvarion's Systems...................................14

Contents

2.4.1 BreezeMAX System (Version 1.5 and higher) ...................................................14

2.4.2 BreezeACCESS VL System (Version 3.1).........................................................14

Chapter 3 - Using the Web Configuration Server

3.1 Introduction to the Web Configuration Server.........................................................16

3.2 Accessing the Web Configuration Server.................................................................17

3.3 Using the Web Configuration Server.........................................................................18

3.4 Home Menu - Product Info Page................................................................................20

3.5 WAN Menu ...................................................................................................................22

3.5.1 WAN Status Page..............................................................................................22

3.5.2 WAN Configuration Page...................................................................................24

3.6 VLAN Tagging Menu...................................................................................................26

3.6.1 VLAN Tagging Page ..........................................................................................26

3.6.2 Adding and Deleting VLANs...............................................................................27

3.6.3 VoIP VLAN Configuration Page.........................................................................29

3.6.4 VLAN Configuration Example 1 .........................................................................30

3.6.5 VLAN Configuration Example 2 .........................................................................32

3.6.6 VLAN Configuration Example 3 .........................................................................35

3.7 Telephone Menu..........................................................................................................38

3.7.1 SIP/H323 Configuration Page............................................................................39

3.7.2 SIP Extensions Page .........................................................................................48

3.7.3 NAT Traversal Configuration Page (SIP Only)...................................................50

3.7.4 STUN Client Configuration Page (SIP only).......................................................51

3.7.5 ToS Page...........................................................................................................52

3.7.6 Line Configuration Page.....................................................................................53

x Voice Gateways System Manual

Contents

3.7.7 Dial Plan Schemes.............................................................................................54

3.8 BW Reservation - DRAP Configuration Page...........................................................58

3.9 System Menu...............................................................................................................62

3.9.1 Set Security Password Page..............................................................................62

3.9.2 Localization Page...............................................................................................64

3.9.3 SNMP Configuration Page.................................................................................65

3.9.4 Service Access Configuration Page...................................................................66

3.9.5 RTP Statistics Page...........................................................................................67

3.10Upgrade Page..............................................................................................................68

3.10.1 Downloader Result Codes (hexadecimal)..........................................................69

3.11Restart Page................................................................................................................71

3.12Logout Page................................................................................................................72

3.13Parameters Summary.................................................................................................73

Appendix A - Internal Class 5 Services

A.1 Actions and Keypad Sequences................................................................................80

A.2 Using the Class 5 Services.........................................................................................81

A.2.1 Call Waiting........................................................................................................81

A.2.2 Call Inquiry.........................................................................................................81

A.2.3 Call Alteration.....................................................................................................81

A.2.4 Call Drop............................................................................................................81

A.2.5 3-Party Conference 1.........................................................................................82

A.2.6 3-Party Conference 2.........................................................................................82

A.2.7 Call Waiting Indication Tone ..............................................................................82

A.2.8 Call Forward.......................................................................................................83

Appendix B - Default Telephony Parameters

Voice Gateways System Manual xi

Contents

Appendix C - New Features

C.1 Metering Support.........................................................................................................90

C.2 Sending VoIP Performance Data to a Remote System............................................90

C.3 Customized Ring Signals ...........................................................................................91

C.4 Spanning Tree Working Mode Configuration...........................................................91

C.5 Ring Signal Frequency and Amplitude Configuration.............................................91

xii Voice Gateways System Manual

Chapter 1 - System Description

In This Chapter:

 “Introducing the Voice Gateway” on page 1-2

 “Specifications” on page 1-3

Chapter 1 - System Description

1.1 Introducing the V o ice Gateway

Alvarion's Voice Gateway enables operators and service providers using Alvarion's

Broadband Wireless Access system to provide subscribers with a number of

broadband services transparently. The Voice Gateway enables bundling services

such as telephony (Voice over IP) and high speed Internet to end-users.

IP-telephony services are supported for standard analog phones or G3 fax

machines. The VG-1D1V has a single POTS interface, and the VG-1D2V has two

POTS interfaces. The Voice Gateways are available with either H.323 or SIP

standard, and support both narrow (compressed) and wide band (uncompressed)

speech codecs, silence suppression with comfort noise, line echo cancellation and

regional telephone parameters. Class 5 services such call waiting and 3-party

conference call are also supported.

Up to 3 telephones can be connected in series to each telephone port. Daisy

chaining of Voice Gateways enables the service provider to offer certain end users,

for example small offices, additional telephone numbers.

The Voice Gateway also supports Internet access or any other Ethernet based

services. The unit can be installed behind a router/NAT due to NAT traversal

support allowing signaling as well as voice packets to correctly reach Softswitch or

Gatekeeper for bi-directional call initiations. The Gateway can handle up to 16

simultaneous VLANs, enabling the operator to offer different services to different

end users behind the unit.

These Gateways incorporate the proprietary DRAP (Dynamic Resources Allocation

Protocol) protocol for automatic registration and allocation of resource. DRAP is a

protocol based on IP/UDP between the Gateway and a DRAP server (e.g. the

BreezeMAX base station). The protocol provides an auto-discovery mechanism for

the Gateway, so no specific configuration is required and the Gateway can

automatically locate and register with the DRAP server. The protocol uses a few

simple messages enabling a Voice Gateway to request resources when calls are

made, and the DRAP server to dynamically allocate them.

The Voice Gateways are designed for remote management and supervision using

either the built-in internal web server or SNMP.

The Voice Gateways are easily updated and upgraded as they support remote

software and configuration file download.

For a complete list of new features, refer to Appendix C.

2 System Description

1.2 Specifications

1.2.1 Telephony and Fax Services

Table 1-1: Telephony and Fax Services

Item Description

VoIP Standard  H323 model: H323v2/4

 SIP model: SIP (RFC 3261)

Internal Class 5 Services Call Waiting, 3-party call, call hold and call alteration,

differentiated ringing tones (refer to Appendix A for more details)

External Class 5 Services Activation/deactivation of class 5 services supported by

the IP-telephony system

Specifications

Fax G3 compliant V.17 14.4 Kbps fax reception and

transmission using the T.38 standard (or in-band using G.711 codec)

Calling Number Identification (CNI)

3rd party initiated pause and rerouting

DTMF In-band and out-band using H.245 an d H.22 5 Regional Settings Telephony signals, tones and cadences (see

FSK, DTMF

External rerouting of media stream during speech, e.g. for pre-paid calling card and record announcement

Appendix B)

1.2.2 Security

Table 1-2: Security

Item Description

VLAN Support IEEE 802.1Q with up to 16 VLAN IDs Authentication Per call authentication and registration

Voice Gateways System Manual 3

Chapter 1 - System Description

1.2.3 Voice Quality

Table 1-3: Voice Quality

Item Description

Voice Codecs  G.711 Ulaw

 G.711 Alaw

 G.729ab

Prioritization  IEEE 802.1p layer-2 prioritization

 DiffServ layer-3 prioritization

General  Adaptive jitter buffer

 Echo cancellation

 Speech sampling rate: 10-60 ms

 Silence suppression with comfort noise

1.2.4 Configuration and Management

Table 1-4: Configuration and Management

Item Description

Management Options  Internal Web Server

 SNMP

SNMP Agents SNMPv1 clientMIB II (RFC 1213), Private MIB Plug & Play Functionality DHCP, including support messages option 60, 61, 43 Software Upgrade Using TFTP Configuration Download Using TFTP

1.2.5 Bridge Functionality

Table 1-5: Bridge Functionality

Item Description

Supported Ethernet Devices Up to 32 MAC addresses

4 System Description

Table 1-5: Bridge Functionality

Item Description

Specifications

Unknown address Forwarding Policy

Bridge Aging Time 180 seconds

Forward Unknown

1.2.6 Mechanical

Table 1-6: Mechanical Specificat io n s

Item Details

Dimensions (W x D x H) 17.6 x 11 x 2.8 cm Weight 230g

1.2.7 Electrical

Table 1-7: Electrical Specifications

Item Details

Power Input 12 VDC from an external power supply, 100-240 VAC,

50-60 Hz, 2A max.

Power Consumption 10.5 W max.

1.2.8 Connectors

Table 1-8: Connectors

Connection Description

LAN Type 10/100Base-TX (RJ-45)Ethernet

connection: MDI/MDIX

Cable Length max 100 m.

PHONE (1 - 2 in VG-1D2V)

Type RJ-11 Number of Phones (REN) Up to 5 Cable Length Max. 500 m

Voice Gateways System Manual 5

Chapter 1 - System Description

Table 1-8: Connectors

Connection Description

WAN Type 10/100Base-TX (RJ-45)

Ethernet Connection to SU-IDU/hub: Straight

Cable Length max 100 m.

12 VDC Standard DC power jack to external power supply

1.2.9 Regulatory Standards Compliance

Table 1-9: Standards Complian ce

Type Standard

EMC  Low Voltage Directive (LVD) 73/23/EEC

 Electromagnetic Compatibility Directive (EMC)

89/336/EEG

Safety  IEC 60950

 CSA C22.2 No. 950-95/UL 1950

 AS/NZS 3260

Emission  EN 55022:1998 Class B

 EN 61000-3-2:1995

 Harmonics; EN 61000-3-3:1995

 Flicker; FCC part 15 (1998) Class B

 AS/NZS 3548 (1995)

Immunity EN 55024:1998

1.2.10 Environmental

Table 1-10: Environmental Specifications

Item Details

Operating temperature 0 o C to 50 o C Operating humidity 10%-95% RH non condensing

6 System Description

Chapter 2 - Installation

In This Chapter:

 “Installation Requirements” on page 2-8

 “Front and Rear Panel Components” on page 2-9

 “Installation and Commissioning” on page 2-11

 “Notes on Using the Voice Gateways in Alvarion's Systems” on page 2-14

Chapter 2 - Installation

2.1 Installation Requirements

2.1.1 Packing List

 Voice Gateway with one (VG-1D1V) or two (VG-1D2V) Phone Ports

 Power supply with a DC connecting cable

 Mains power cable

2.1.2 Additional Installation Requirements

 A straight Ethernet cable for connecting the WAN port to the SU-IDU

 An Ethernet cable for connecting to the user's data equipment (straight for

connecting to a PC, crossed for connecting to a hub/switch)

 Standard phone cable(s) with RJ-11 connectors.

 Mains plug adapter (if the power plug on the supplied mains power cable does

not fit local power outlets).

 Portable PC with an Ethernet card and an Ethernet cable for configuring the

Voice Gateway parameters using a web browser.

8 Installation

Front and Rear Panel Components

2.2 Front and Rear Panel Components

2.2.1 Connectors

Figure 2-1: Voice Gateway VG-1D2V Back Panel

NOTE

The VG-1D1V has a single Phone connector.

Table 2-1: Voice Gateway Connectors

Name Connector Functionality

Phone 1 RJ-11 Connections to the user's telephones Phone 2 (VG-1D2V only) RJ-11 Connections to the user's telephones LAN 10/100Base-T (RJ-45) Connection to the user's data equipment WAN 10/100Base-T (RJ-45) Connection to the SU-IDU 12 VDC DC power jack Connection to power supply

2.2.2 Reset to Factory Default Configuration

Press down the RESET button on the back of the unit for at least 5 seconds to

reset all configurable parameters back to their original default values. After

releasing the RESET button, the PWR, WAN and LAN LEDs blink twice, indicating

proper operation. The affect on the selected IP parameters acquisition method

depends on the time the RESET button is held in the pressed position:

 If the RESET button is pressed down for 5 to 10 seconds: The unit will use

DHCP to get the WAN IP parameters.

 If the RESET button is pressed down for more than 10 seconds: The unit will

use the static (manually defined) WAN IP parameters (IP 192.168.254.254

Mask 255.255.255.0).

Voice Gateways System Manual 9

Chapter 2 - Installation

For more details on configuration of DHCP and static IP parameters, refer to

Section 3.5.2.

2.2.3 LEDs

NOTE

The VG-1D1V has a single Phone LED.

Figure 2-2: VG-1D2V Front Panel

Table 2-2: Voice Gateway LEDs

Name Symbol Description Functionality

Phone 1 Phone service

indication

 Off -Phone line does not get IP telephony

services

 On - Phone line is connected to the

IP-telephony system

Phone 2 (VG-1D2V only)

Phone service indication

 Off -Phone line does not get IP telephony

services

 On - Phone line is connected to the

IP-telephony system

LAN LAN port status

indication

 Off - Ethernet Link not detected

 On - Ethernet link connected, no activity

 Blinking - Ethernet link activity

WAN WAN port status

indication

 Off - Ethernet link not detected

 On - Ethernet link connected, no activity

POWER PWR Power Indication

10 Installation

 Blinking - Ethernet link activity  Off - unit is not powered or power failed

 Green - power OK

Installation and Commissioning

2.3 Installation and Commissioning

The unit can be placed on a desktop or a shelf. The location should be selected

taking into account the necessary connections to mains power, SU-IDU and user's

data/telephony equipment.

It is assumed that installation and commissioning of the SU has already been

completed and that the SU is connected to the Base Station.

To install the Voice Gateway:

1 Connect the DC power cable of power supply to the 12 VDC jack on the rear

panel of the unit.

2 Connect the mains power cable to the power supply. Connect the other end of

the mains power cable to the AC mains.

NOTE

The color codes of the power cable are as follows: Brown

Blue Neutral 0 Yellow/Green Ground

3 After power up, all front panel LEDs bilnk once, and then the PWR, WAN and

LAN LEDs bilnk twice, indicating that the unit operates properly. Then the

PWR LED is lit. Other LEDs may also be lit, according to the status of the

WAN, LAN and Phone ports, as described in Section 2.2.3.

4 Connect a PC to the WAN or LAN port using a crossed Ethernet cable.

Configure the PC with a static IP address 192.168.254.2 and subnet mask

255.255.255.0. (The IP address of the WAN port for management purposes

only is 192.168.254.254 and netmask 255.255.255.0)

NOTE

The VG can be accessed via the WAN or LAN port using the WAN IP address.

Phase ~

5 Open a web browser and connect to the unit by entering

http://192.168.254.254. in the address field.

Voice Gateways System Manual 11

Chapter 2 - Installation

6 If the Web Configuration Server is password protected, you will be prompted to

7 Configure the necessary parameters according to instructions supplied by the

enter your username and password in order to log in to the system. The

default username is operator and the default password is installer. See

Chapter 3 for details on using the Web Configuration Server.

system administrator. The mandatory parameters that must be configured

properly are:

 Enable DRAP (in BW Reservation page) only if DRAP is supported by the

wireless system (currently DRAP is supported by BreezeMAX equipment

with SW version 1.5 or higher and BreezeACCESS VL with SW version 4.0).

Uncheck if DRAP is not used.

 LAN/WAN VLAN Tagged Port Membership parameters (VLAN page) and

VoIP VLAN parameters (VoIP VLAN Configuration page).

 Telephony parameters (per line) in the SIP Configuration/H323 Telephone

page: Telephone Line Enable/Disable, primary SIP Server/H323 Gate

Keeper parameters, User Name and Password (SIP model), Telephone

Number, Telephone domain name (SIP model). Certain H323 Gatekeepers

require configuration of a unique H323 Alias.

 WAN IP parameters (WAN Configuration page): For operation as a DHCP

client, check the Obtain WAN Configuration dynamically. For static IP

configuration, check the Specify fixed WAN configuration option and

specify the IP Address, Subnet Mask and Default Gateway.

8 Restart the unit from the Restart page. 9 If VLANs are configured for management, you will lose management from the

PC, unless the packets are tagged from the PC towards the Voice Gateway. To

resume management capabilities, return to factory defaults (see

Section 2.2.2).

10 Disconnect the PC used for configuration. 11 Use a straight Ethernet cable to connect the WAN port on the rear panel of the

unit to the Ethernet port of the SU-IDU. The length of the indoor-to-outdoor

Ethernet cable should not exceed 90 meters. The length of the Ethernet cable

connecting the indoor unit to the user's equipment, together with the length of

the Indoor-to-Outdoor cable, should not exceed 100 meters.

12 Connect the data equipment using a 10/100 Base-T Ethernet cable to the LAN

port. The length of the Ethernet cable should not exceed 100m. Use a straight

cable for connecting to a PC, or a crossed cable for connecting to a

hub/switch).

12 Installation

Installation and Commissioning

13 Use standard telephone cord(s) with RJ-11 termination to connect the

telephony equipment to the unit.

14 Verify proper operation using the LED indicators (see Table 2-2). 15 To verify data connectivity, from the end-user's PC or from a portable PC

connected to the unit, try to connect to the Internet or to ping another unit in

the network.

16 Verify proper telephony operation by establishing a call to another telephone

(for each enabled line).

Voice Gateways System Manual 13

Chapter 2 - Installation

2.4 Notes on Using the Voice Gateways in Alvarion's Systems

2.4.1 BreezeMAX System (Version 1.5 and higher)

 Access the Monitor program of the SU from a PC connected to the LAN port of

the Gateway. The SU's Monitor program uses the fixed IP address

192.168.254.251 with the subnet mask 255.255.255.0. The PC used for

accessing the Monitor program should be configured to belong to the same

subnet. It is recommended to set the PC's IP address to 192.168.254.250,

which is the default TFTP Server IP address in the Monitor (required for

downloading SW versions and for downloading/uploading configuration files).

 Information about the DRAP-enabled Gateways that are connected to each SU

can be viewed in the Base Station's Monitor program (in the Voice/Networking

Gateways option of the Configuration menu for a selected SU). The displayed

information includes Gateway's type, IP Address, and the VLAN ID used for

management.

 In general, the same VLAN should be configured in the Voice Gateway for

Management (Default VLAN ID) and Voice (RTP and Signaling) as the Voice

Gateway uses one IP address for two VLANs and the default router in the

backbone cannot operate in this mode.

 To support the required quality of service when DRAP is used, provision the

correct VoIP Service. If DRAP is not used, provision an L2 Service with a CG

connection (refer to the BreezeMAX System Manual for details).

2.4.2 BreezeACCESS VL System (Version 3.1)

 To access the Monitor program of the SU from a PC connected to the LAN port

of the Gateway, the WAN port must be configured with static IP address that is

in the same subnet as the IP Address of the SU, and subnet mask

255.255.255.0 (the default IP address is 10.0.0.1 with a Subnet Mask

255.255.255.0). The PC used for accessing the Monitor program should be

configured to belong to the same subnet.

 Configure the Traffic Prioritization parameters in both the SU and the AU to

ensure high priority for RTP traffic. Refer to the BreezeACCESS VL System

Manual for details.

14 Installation

Chapter 3 - Using the Web Configuration Server

In This Chapter:

 “Introduction to the Web Configuration Server” on page 3-16

 “Accessing the Web Configuration Server” on page 3-17

 “Using the Web Configuration Server” on page 3-18

 “Home Menu - Product Info Page” on page 3-20

 “WAN Menu” on page 3-22

 “VLAN Tagging Menu” on page 3-26

 “Telephone Menu” on page 3-38

 “BW Reservation - DRAP Configuration Page” on page 3-58

 “System Menu” on page 3-62

 “Upgrade Page” on page 3-68

 “Restart Page” on page 3-71

 “Logout Page” on page 3-72

 “Parameters Summary” on page 3-73

Chapter 3 - Using the Web Configuration Server

3.1 Introduction to the Web Configuration Server

The Voice Gateway can be configured using the following methods:

 The Web Configuration Server

 An .ini-file loaded into the unit from a TFTP-server or automatically

downloaded using DHCP option 43.

This document describes the configuration using the Web Configuration Server.

16 Operation

Accessing the Web Configuration Server

3.2 Accessing the We b Configura t ion Server

To manage the unit you must have prior knowledge of its WAN IP Address. Follow

the steps below to access the Web Configuration Server:

1 Open a web browser. 2 Enter the WAN IP address of the unit in the Address field of the browser and

click Enter. E.g., http://192.168.254.254 (default).

3 If the Web Configuration Server is password protected, you will be prompted

to enter your user name and password in order to login to the system.

 To login with operator privileges (full access and read/write privileges), the

default user name is operator and the default password is installer.

 To login with administrator privileges (partial access and read/write

privileges), the default user name is admin. No password is required.

4 The Web Configuration Server main view appears on the screen.

Voice Gateways System Manual 17

Chapter 3 - Using the Web Configuration Server

3.3 Using the Web Configuration Server

The Web Configuration Server view consists of a number of menu links (to the

left). Clicking on each of them will display the configuration/status page for the

selected menu item, with the applicable content (configurable parameters/options

or status information) in the main area. Several pages include a page selection bar

at the top of the page, enabling selection between several pages related to the

same menu item. The displayed pages may vary depending on user privileges.

Figure 3-1: Web Configuration Page

CAUTION

Many pages include a "Save Settings" button. Click on the Save Settings button before selecting another page/menu item, or before quitting the application. The Save Settings functionality in many cases is per page - if you leave the page without clicking the Save Settings button, all the changes in the page will be lost.

Changes to most of the settings are applied only after restarting the unit (refer to

Section 3.11).

18 Operation

Using the Web Configuration Server

CAUTION

There is no control that the entered values are valid or have the correct format or range. If invalid values are entered, access to the unit may be lost and in that case a factory default procedure must be performed. Refer to Section 2.2.2 for information about how to reset the Voice Gateway to factory default parameters.

Voice Gateways System Manual 19

Chapter 3 - Using the Web Configuration Server

3.4 Home Menu - Product Info Page

The Product info page provides general information on the Voice Gateway.

Figure 3-2: Product Info Page

The Product info page includes the following components:

Table 3-1: Product Info Page Parameters

Parameter Description

Name The unit's model Mac address The MAC address of the unit Serial Number The serial number of the unit Product number Not Used Product revision The hardware revision Production week Production date in the format <yy>w<ww>. <yy> is the year (two last

digits) and ww is the week (two digits). Default configuration The unit's configuration Downloader revision The revision of the SW download SW module.

20 Operation

Home Menu - Product Info Page

Table 3-1: Product Info Page Parameters

Parameter Description

Reported download status The status of the SW download operation. For more details refer to

Section 3.10.1. Main software revision The unit's main SW version Operator defaults revision The custom .ini file (if exists)

In any case of contact with Alvarion Customer Service, include the Default

configuration, Downloader revision, Main software revision and Operator defaults

revision (.ini file) if exists.

Voice Gateways System Manual 21

Chapter 3 - Using the Web Configuration Server

3.5 WAN Menu

The WAN menu page includes settings related to the operation and functionality

on the WAN (network) side of the unit.

NOTE

Be careful when setting these parameters to avoid conflicts in the network.

The WAN page selection bar includes the following options:

 WAN Status (Section 3.5.1)

 WAN Configuration (Section 3.5.2)

3.5.1 WAN Status Page

Figure 3-3: WAN Status Page

The WAN Status page includes the following components:

22 Operation

WAN Menu

Table 3-2: WAN Status Page Parameters

Parameter Description Interface Status

Enabled The administrative status of the WAN port: Yes or No. In the current

version the administrative status cannot be disabled.

Service The configured operation mode. In current version it is always

Bridged.

Bridge Status The method of handling packets with an unknown destination address.

In the current version it is always Forwarding.

Protocol The protocol used for data transmission: In the current version it is

always Ethernet.

Interface Status The operational status of the WAN port: Up or Down.

Network Settings

Dynamic IP Assignment The method of configuring IP Address, Subnet Mask, Default Gateway

and DNS Address, as defined in the WAN Configuration page:

 Yes (via DHCP): the parameters are obtained from a DHCP server.

 No: the parameters are configured manually

IP Address The IP address of the unit MAC Address The MAC address of the unit Subnet Mask The IP Subnet Mask Default Gateway The Default Gateway address DNS Address IP DNS Server address Domain Name The Domain Name as defined in the WAN Configuration page VLAN Tag The VLAN ID tag defined for management traffic Priority Tag The Priority tag defined for management traffic

Click on the Update button to refresh the display.

Voice Gateways System Manual 23

Chapter 3 - Using the Web Configuration Server

3.5.2 WAN Configuration Page

Figure 3-4: WA N Configuration Page

The WAN Configuration page includes the following components:

Table 3-3: WAN Configuration Page Parameters

Parameter Description

Device Operating Mode The operating mode of the unit. In current version the

operation mode is always Bridge.

Obtain WAN configuration using DHCP

Select this option to obtain IP parameters from a DHCP server . See also Section 2.2.2.

24 Operation

WAN Menu

Table 3-3: WAN Configuration Page Parameters

Parameter Description

Client identity Applicable only if the "Obtain WAN configuration dynamically"

option is selected. The method used for identifying the client (Option 61). The options are:

Standard: The unit's MAC address Custom: An identification string of up to 25 characters. The

default is null (an empty string)

Vendor ID Applicable only if the "Obtain WAN configuration dynamically"

option is selected. The V endor ID (Option 60). A string of up to 25 characters. The default used by the unit is VoIP (not displayed).

Specify static WAN configuration Select this option to configure the IP parameters manually.

3.6 VLAN Tagging Menu

The VLAN Tagging page selection bar includes the following options:

 VLAN Tagging (Section 3.6.1)

 VoIP VLAN Configuration (Section 3.6.3)

3.6.1 VLAN Tagging Page

The Voice Gateway supports 802.1Q VLAN standard, allowing IEEE 802 Local

Area Networks (LANs) of all types to be connected together with Media Access

Control (MAC) Bridges, as specified in ISO/IEC 15802-3. This standard defines

the operation of Virtual LAN (VLAN) Bridges that permit the definition, operation

and administration of Virtual LAN topologies within a bridged LAN infrastructure.

Figure 3-5: VLAN Tagging Page

26 Operation

VLAN Tagging Menu

The VLAN page enables defining up to 16 VLANs, and it includes the following

components:

Table 3-4: VLAN Page Parameters

Parameter Description

Tagged Port Membership A table displaying the defined VLANs. For details on

modifying the table refer to Section 3.6.2 below.

Untagged VLAN ID The VLAN ID that is defined for untagged data on the WAN

port (text box on the left side) and the LAN port (text box on the right side). This parameter must be consistent with a properly configured VLAN in the tagged port member ship. For examples on VLAN configuration, see Section 3.6.4 and Section 3.6.5.

The range for both parameters is from 1 to 4094.

Default VLAN ID The text box on the left side is for the WAN port. This is the

VLAN defined for management frames (SNMP, HTTP, TFTP) arriving on the WAN port.

The DRAP packets are tagged with the default VLAN configuration.

The range is from 1 to 4094.

NOTE

Management of the unit can only be done from the WAN port.

3.6.2 Adding and Deleting VLANs

To add a VLAN:

1 Click on the Add VLAN button. The VLAN Editor (Add) is displayed:

Voice Gateways System Manual 27

Chapter 3 - Using the Web Configuration Server

Figure 3-6: VLAN Editor (Add VLAN)

2 Enter the VLAN ID (1 to 4094), VLAN NAME (A descriptive string of printable

characters. Do not use special characters such as space or comma), and the

VLAN priority tag (0 to 7).

3 If applicable packets need to be tagged on the WAN/LAN port, check the

relevant Yes option. Otherwise check the No option. Note that only one VLAN

can be untagged on each port (or on both).

4 Click OK. The newly added entry will be added to the Tagged Port Membership

table.

To delete a VLAN from the Tagged Port Membership table:

1 Click on the row ID number of the entry you wish to remove. The VLAN Editor

(Delete) is displayed:

28 Operation

Figure 3-7: VLAN Editor (Delete VLAN)

VLAN Tagging Menu

2 Click on the Delete button. The entry will be removed from the Tagged Port

Membership table.

3.6.3 VoIP VLAN Configuration Page

Figure 3-8: VoIP VLAN Configuration Page

Voice Gateways System Manual 29

Chapter 3 - Using the Web Configuration Server

The VoIP VLAN configuration page enables defining the following parameters:

Table 3-5: VoIP VLAN Configuration Page Parameters

Parameter Description Call Signaling

VLAN Tag The VLAN ID tag for VoIP call signaling packets. If not set, the

Default VLAN ID set for WAN (in the VLAN Tagging page) will also apply for VOIP.

Priority Tag The Priority tag for VoIP call signaling packets. If not set, the

priority tag defined for the Management VLAN in the Tagged Port Membership (in the VLAN Tagging page), will also apply for VOIP.

RTP

VLAN Tag The VLAN ID tag for RTP and RTCP packets. If not set, the

Default VLAN ID set for WAN (in the VLAN Tagging page) will also apply for VOIP.

Priority Tag The Priority tag for RTP and RTCP packets. If not set, the priority

tag defined for the Management VLAN in the Tagged Port Membership (in the VLAN Tagging page), will also apply for VOIP.

Typically, the same VLAN is used for management, call signaling and RTP. In this

case, the same VLAN and Priority Tags should be configured for management

(Default VLAN on WAN port in the VLAN Tagging page), Call Signaling and RTP.

However, the Voice Gateway supports separation of VLANs and allows defining 3

different VLANs for management, call signaling and RTP traffic (this may require a

proper router). Different Priority tags for management, call signaling and RTP can

be configured. The Priority tag for management is defined in the Priority field of

the management VLAN ID (configured in the Tagged Port Membership table).

3.6.4 VLAN Configuration Example 1

This example describes how to define the following configuration:

 VLAN ID 100, VLAN Priority 7 for Voice (call signaling, RTP and RTCP) and 5

for Management packets on the WAN port.

 VLAN ID 200, VLAN Priority 0 for data on the WAN port and untagged to/from

the LAN port.

30 Operation

VLAN Tagging Menu

VLAN200 (Data)

VLAN100 (Voice & Management)

Untagged

POTS

Figure 3-9: VLAN Configuration Example 1

1 In the VLAN page, click Add VLAN to open the VLAN Editor. 2 In the VLAN Editor, enter the follwing for Voice and Management VLAN:

 VLAN ID: 100

 VLAN NAME: Voice&Mng

 VLAN Priority: 5

 WAN: Yes

 LAN: No

3 Click OK to add the VLAN to the Tagged Port Membership table. 4 Enter the VLAN ID for Voice and Management (100) in the field Default VLAN

ID on WAN port, and click Save.

5 In the Page Selection bar, click on VoIP VLAN Configuration to open the VoIP

VLAN Configuration page. Enter 100 in the VLAN Tag fields for both Call

Signaling and RTP. Enter 7 in the Priority Tag field for both Call Signaling and

RTP. Click Save VoIP VLAN Settings. Go back to the VLAN Tagging page.

6 In the VLAN page, click Add VLAN to open the VLAN Editor to configure the

data VLAN.

Voice Gateways System Manual 31

Chapter 3 - Using the Web Configuration Server

7 In the VLAN Editor, enter the follwing for data:

 VLAN ID: 200 (an arbitrary selection-a VLAN ID is required for defining the

untagged data. This VLAN tag is only used internally in the unit)

 VLAN NAME: Data

 VLAN Priority: 0

 WAN: Yes

 LAN: No

8 Click OK to add the VLAN to the Tagged Port Membership table. 9 Enter the VLAN ID for untagged data (200) in the fields Untagged VLAN ID on

LAN port and click Save.

10 Restart the unit to apply the changes.

3.6.5 VLAN Configuration Example 2

This example describes how to define the following configuration:

 Two daisy-chained Voice Gateways: VG-1 and VG-2.

 VLAN ID 100, VLAN Priority 7 for Voice (call signaling, RTP and RTCP) and

Management packets on the WAN port.

 VLAN ID 200, VLAN Priority 4 for data on WAN port (VG-1)

 No VLAN for data on the LAN port (VG-2).

32 Operation

VLAN Tagging Menu

VG-1

VG-2

Untagged

Figure 3-10: VLAN Configuration Example 2

3.6.5.1 VG-1 Configuration

VLAN 200

VLAN100

POTS

1 In the VLAN page, click Add VLAN to open the VLAN Editor. 2 In the VLAN Editor, enter the follwing for Voice and Management VLAN:

 VLAN ID: 100

 VLAN NAME: Voice&Mng

 VLAN Priority: 7

 WAN: Yes

 LAN: Yes

3 Click OK to add the VLAN to the Tagged Port Membership table. 4 Enter the VLAN ID for Voice and Management (100) in the fields Default VLAN

ID on WAN port, and click Save.

5 In the Page Selection bar, click on VoIP VLAN Configuration to open the VoIP

VLAN Configuration page. Enter 100 in the VLAN Tag fields for both Call

Voice Gateways System Manual 33

Chapter 3 - Using the Web Configuration Server

Signaling and RTP. Enter 7 in the Priority Tag field for both Call Signaling and

RTP. Click Save VoIP VLAN Settings. Go back to the VLAN Tagging page.

6 In the VLAN page, click Add VLAN to open the VLAN Editor. 7 In the VLAN Editor, enter the follwing for Data VLAN:

 VLAN ID: 200

 VLAN NAME: Data

 VLAN Priority: 4

 WAN: Yes

 LAN: Yes

8 Click OK to add the VLAN to the Tagged Port Membership table. 9 Enter the VLAN ID for untagged data (200) in the field Untagged VLAN ID on

LAN port and click Save.

10 Restart the unit to apply the changes.

3.6.5.2 VG-2 Configuration

1 In the VLAN page, click Add VLAN to open the VLAN Editor. 2 In the VLAN Editor, enter the follwing for Voice and Management VLAN:

 VLAN ID: 100

 VLAN NAME: Voice&Mng

 VLAN Priority: 7

 WAN: Yes

 LAN: No

3 Click OK to add the VLAN to the Tagged Port Membership table. 4 Enter the VLAN ID for Voice and Management (100) in the field Default VLAN

ID on WAN port, and click Save.

5 In the Page Selection bar, click on VoIP VLAN Configuration to open the VoIP

VLAN Configuration page. Enter 100 in the VLAN Tag fields for both Call

Signaling and RTP. Enter 7 in the Priority Tag field for both Call Signaling and

RTP. Click Save VoIP VLAN Settings. Go back to the VLAN Tagging page.

6 In the VLAN page, click Add VLAN to open the VLAN Editor. 7 In the VLAN Editor, enter the follwing for untagged data:

34 Operation

 VLAN ID: 300 (an arbitrary selection-a VLAN ID is required for defining the

untagged data. This VLAN tag is only used internally in the unit)

 VLAN NAME: Untagged

 VLAN Priority: 0

 WAN: Yes

 LAN: Yes

8 Click OK to add the VLAN to the Tagged Port Membership table. 9 Enter the VLAN ID for untagged data (300) in the fields Untagged VLAN ID on

LAN port and Untagged VLAN ID on WAN port, and click Save.

10 Restart the unit to apply the changes.

3.6.6 VLAN Configuration Example 3

This example describes how to define the following configuration:

VLAN Tagging Menu

 One Voice Gateway.

 VLAN ID 60, VLAN Priority 6 for Voice (call signaling, RTP and RTCP) and

Management packets on the WAN port.

 No VLAN for data packets on WAN and LAN ports

o VLAN

VLAN 60

Untagged

POTS

Figure 3-11: VLAN Configuration Example 3

Voice Gateways System Manual 35

Chapter 3 - Using the Web Configuration Server

3.6.6.1 Method 1

1 In the VLAN page, click Add VLAN to open the VLAN Editor. 2 In the VLAN Editor, enter the follwing for Voice and Management VLAN:

 VLAN ID: 60

 VLAN NAME: Voice&Mng

 VLAN Priority: 6

 WAN: Yes

 LAN: No

3 Click OK to add the VLAN to the Tagged Port Membership table. 4 Enter the VLAN ID for Voice and Management (60) in the field Default VLAN

ID on WAN port, and click Save.

5 In the Page Selection bar, click on VoIP VLAN Configuration to open the VoIP

VLAN Configuration page. Enter 60 in the VLAN Tag fields for both Call

Signaling and RTP. Enter 6 in the Priority Tag field for both Call Signaling and

RTP. Click Save VoIP VLAN Settings. Go back to the VLAN Tagging page.

6 In the VLAN page, click Add VLAN to open the VLAN Editor. 7 In the VLAN Editor, enter the follwing for untagged data:

 VLAN ID: 90 (an arbitrary selection-a VLAN ID is required for defining the

untagged data. This VLAN tag is only used internally in the unit)

 VLAN NAME: Untagged

 VLAN Priority: 0

 WAN: Yes

 LAN: Yes

8 Click OK to add the VLAN to the Tagged Port Membership table. 9 Enter the VLAN ID for untagged data (90) in the fields Untagged VLAN ID on

LAN port and Untagged VLAN ID on WAN port, and click Save.

10 Restart the unit to apply the changes.

3.6.6.2 Method 2

1 In the VLAN page, click Add VLAN to open the VLAN Editor. 2 In the VLAN Editor, enter the follwing for Voice and Management VLAN:

36 Operation

VLAN Tagging Menu

 VLAN ID: 60

 VLAN NAME: Voice&Mng

 VLAN Priority: 6

 WAN: Yes

 LAN: No

3 Click OK to add the VLAN to the Tagged Port Membership table. 4 Enter the VLAN ID for Voice and Management (60) in the field Default VLAN

ID on WAN port, and click Save.

5 In the Page Selection bar, click on VoIP VLAN Configuration to open the VoIP

VLAN Configuration page. Enter 60 in the VLAN Tag fields for both Call

Signaling and RTP. Enter 6 in the Priority Tag field for both Call Signaling and

RTP. Click Save VoIP VLAN Settings. Go back to the VLAN Tagging page.

6 There is no need to define VLAN in the Port Tag Membership table or in the

Untagged WAN and LAN fields. Untagged packets will pass through LAN to

WAN and WAN to LAN.

7 Restart the unit to apply the changes.

Voice Gateways System Manual 37

Chapter 3 - Using the Web Configuration Server

3.7 Telephone Menu

In the SIP model, the Telephone page selection bar includes the following options:

 SIP (Section 3.7.1)

 SIP Extensions (Section 3.7.2)

 NAT (Section 3.7.3)

 STUN Client (Section 3.7.4)

 ToS (Section 3.7.5)

In the H323 model, the Telephone page selection bar includes the following

options:

 H323 (Section 3.7.1)

 ToS (Section 3.7.5)

38 Operation

3.7.1 SIP/H323 Configuration Page

SIP Configuration page:

Telephone Menu

Figure 3-12: SIP Configuration Page (VG-1D2V)

Voice Gateways System Manual 39

Chapter 3 - Using the Web Configuration Server

H323 Telephone page:

Figure 3-13: H323 Telephone Page (VG-1D2V)

The SIP Configuration page/H323 Telephone pages include the following

components:

40 Operation

Telephone Menu

Table 3-6: SIP Configuration/H323 Telephone Page Parameters

Parameter Description

Dialplan The Dialplan parameter defines how the Voice Gateway decides

that a complete number has been dialed. See more details in Section

3.7.7.

The default value is xx.T|xx.#, which means that each of the following schemes can be used:

 xx.T: Dial timeout. Any number of digits may be dialed.

Following T seconds in which no new digit is dialed, a decision is reached that dialing was completed and the unit will send the dialing sequence received up to this time as a complete telephone number . This is necessary since the whole telepho ne number is sent at once and not digit by digit.

 xx.#: Any number of digits may be dialed. A decision that dialing

was completed will be reached once # is pressed.

The combination of both schemes means that dialing is completed either after a timeout of T seconds or after pressing #.

Dial timeout The timeout in seconds for the dial timeout dialplan. The number of

seconds that the unit waits before it sends a complete telephone number . This is necessary since the whole telephone number is sent at once and not digit by digit.

The range is 1 to 60 seconds Default value is 4 seconds.

Use # Use # as a quick dial function. T o se nd the # along with the number

to the server, un check the box. The default is enabled.

RTP Port Range (SIP model only)

The start and end UDP port-range for RTP protocol. Recommended values for Start and End ports are in the range 1030-65535.

The default Start port is 8000. The default End port is 8015.

Telephone line Switch the telephone line On or Off. The default is Off.

Voice Gateways System Manual 41

Chapter 3 - Using the Web Configuration Server

Table 3-6: SIP Configuration/H323 Telephone Page Parameters

Parameter Description

HA mode The High Availability mode defines the support of a secondary Gate

Keeper/SIP Server for high system availability, redundancy, and scalability. When a secondary server is available, the unit will try to register to the secondary server after 10 failed attempts to register to the primary server.

The available options are:

 Fixed: The secondary Gate Keeper/SIP Server IP address is

defined manually by the Gate Keeper/SIP Server IP (secondary) parameter.

 Auto: The secondary Gate Keeper /SIP Server IP address is

supplied by the primary Gate Keeper/SIP Server.

 Off: Secondary Gate Keeper/SIP Server is not supported.

SIP Server IP (primary) (SIP model only)

SIP Server Port (primary) (SIP model only)

SIP Server IP (secondary) (SIP model only)

SIP Server Port (secondary) (SIP model only)

Gate Keeper IP (primary) (H323 model only)

The default is Off. The IP address for the primary SIP server/proxy who is responsible

for managing the Voice Gateway in the specific network. If HA-mode is set to Auto, the primary SIP server/proxy provides to the Voice Gateway during registration an IP address for the secondary system.

The port used for the primary system. The recommended values are in the range 1030-65536. The defa ult is 506 0.

The IP address of the secondary SIP server/proxy.

The port used for the secondary system. The recommended values are in the range 1030-65536. The defa ult is 506 0.

The IP address for the primary Gate Keeper who is responsible for managing the V oice Gateway in the specific network. If HA-mode is set to Auto, the primary Gate Keeper provides to the Voice Gateway during registration an IP address for the secondary system.

Gate Keeper IP (secondary) (H323 model only)

42 Operation

The IP address of the secondary Gate Keeper.

Table 3-6: SIP Configuration/H323 Telephone Page Parameters

Parameter Description

Telephone Menu

User Name (SIP model only)

Password

The SIP user Name. Format (name or number) depend s on the SIP server. A string of up to 25 characters.

The SIP user Password. Format (name or number) depend s on the SIP server. A string of up to 25 characters.

(SIP model only) H323 Alias The unit's name used when registering the unit at the Gate Keeper.

If used, the H323 alias must be set to a unique value for each telephone line in the network in order for the system to accept it. Up to 25 characters. The default is null (not used during registration).

Outgoing Display name The name to be displayed on the caller ID display of a receiving

party (if supported by the network). Up to 25 characters with no spaces.

Telephone number The telephone number of the specific telephone line to be used

when registering the unit at the Gate keeper/SIP Server. The telephone number is limited to 25 characters. It may also be an

e-mail address (limited to 25 characters before the @ sign). The Telephone number must be set to a unique value for each

telephone line in the network in order for the system to accept it.

Telephone domain name (SIP model only)

Port (SIP model only)

Message Waiting Account (SIP model only)

The domain-name. The Telephone domain name is limited to 25 characters, i.e. 25 characters after the @-sign. If not specified by the user, the same information as defined in the SIP Server IP field will be used.

The number of the outgoing signaling port on the telephone line. Line1 and Line 2 cannot have the same port number. The range is from 1030 to 65535. The default is 5060 for Line 1 and 5061 for Line 2.

When a message is waiting in the network-based voice mail system, a discontinuous dial tone will be played when the handset goes off hook. To enable, a SIP server supporting Interactive Voice Response (IVR) is required.

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Table 3-6: SIP Configuration/H323 Telephone Page Parameters

Parameter Description

Incoming CLIP The Calling Line Identity Presentation (Caller ID) option for the

telephone line. If On is selected, the Caller ID information of a calling party in incoming calls will be displayed on a caller ID display attached to the telephone line.

Caller ID can be restricted permanently using a customized .ini file. The default is Off.

Keepalive Timeout (seconds) The interval of waiting for acknowledgement message from the

server. If Keep-alive timeout is sent from the network, it will override the setting in the Voice Gateway. The interval for s ending Keep-alive registration messages from the Gateway is half the configured value (600 seconds with the default timeout of 1200 seconds).

In case of registration problem, try changing the value to 1800 seconds.

The range is from 10 to 65535 seconds. The default is 1200 seconds.

Ring signal [0 - 9] The Ring signal parameter provides a selection of 10 different ring

patterns (0-9) that the unit can use. The default is 0.

Transport (SIP model only)

Preferred codecs Displays the currently supported codecs, according to the defined

Configure whether signaling shall use UDP or TCP. The default is UDP.

priorities. Click the Set Codecs/Fax button to change codecs

settings/priorities. NOTE: Click Save before clicking the Set Codecs/Fax button.

Otherwise, all configuration changes in the Telephone page will be lost.

Click on the Save button before leaving the page to save the new settings. The new

settings will be applied after restarting the unit.

Click the

Set Codecs/Fax button to change codecs settings/priorities as

described in the following section.

44 Operation

3.7.1.1 Codecs and Fax Configuration

After clicking the Set Codecs/Fax button, the Codecs and Fax Configuration page

is displayed.

Telephone Menu

Figure 3-14: Codecs and Fax Configuration Window - VG-1D2V

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The jitter buffer options are common to both lines (if applicable):

Table 3-7: Jitter Buffer Options

Parameter Description

Adaptive Jitter Buffer Maximum Delay

Fixed Jitter Buffer When using fax only, it is recommended to use a fixed jitter

Automatically switch to Fixed Jitter Buffer

The Voice Gateway uses a Jitter Buffer to eliminate jitter effects. The size of the buffer chan ges dynamically to reflect actual jitter conditions. The Adaptive Jitter Buffer Maximum Delay defines the maximum size that is available for the jitter buffer (the larger the size, the greater the potential delay).

The range is from 100 to 300 milliseconds. The default duration is 100 milliseconds.

buffer. The fixed jitter buffer may affect voice conversation performance.

The range is from 100 to 300 milliseconds. The default duration is 40 milliseconds.

Select this option in order to use both fax and voice. The Voice Gateway automatically switches to the configured Fixed Jitter Buffer upon detecting a fax/modem tone.

Faxes can be transmitted when Codec G.711 or T38 are selected.

The following settings are available for each line:

Table 3-8: Codecs and Fax Configuration Parameters

Parameter Description

Codec The Codec check boxes identify which codecs are used.

By default all three codecs are selected (checked). NOTE: G 729 with Annex A is implemented in the Voice

Gateway. It enables communication with devices using either G729 with Annex A or G729 with Annex A and Annex B. It is not possible to communicate with devices using G729 with Annex B only.

For each Codec in use, the following can be configured:

46 Operation

Telephone Menu

Table 3-8: Codecs and Fax Configuration Parameters

Parameter Description

SS The SS (Silent Suppression) option for outgoing calls. When the

SS option is enabled, silence intervals are identified and only relevant information is transmitted, using less bandwidth than during voice activity intervals. This allows for a better overall utilization of the available bandwidth. It is possible to enable Silent Suppression with G729 codec. Silent Suppression is not applicable when using the G711 codecs.

The default (G729) is SS disabled.

EC The EC (Echo Cancellation) op tion, defines whether to activate

the echo cancellation mechanism for improved voice quality. EC is not used during Fax (T.38) transmissions.

The default is enabled.

Packet The packet size in milliseconds.

The range is from 10 to 150 milliseconds. The default is 30 ms for G729 and 20 ms for G711A and G711U.

Keypad The "Keypad" field indicated which transmission method to be

used for user input DTMF signaling (i.e. phone banking ). "None" means in-band, which should be used with G.711 only.

For SIP model the options are None, RFC2833 and SIP INFO. RFC2833 and SIP INFO should be used primarily with G.729 but could also be used with G.711. The default is None for G711 codecs and RFC2833 for G729.

For H323 model the options are H225, H245, RFC2833 and None. The default is None for G711 codecs and H225 for G729.

Priority The Priority parameter defines the relative priorities to be offered

during capabilities' exchange. If only G711A and G711U are used, the permitted priorities are 1 and 2.

If all 3 codecs are used, the permitted priorities are 1, 2 and 3. Voice codec negotiation/priority is always performed between 2

end-points and depending on which side initiated the negotiation.

The default is Priority 1 to G711A, Priority 2 to G711U.

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Table 3-8: Codecs and Fax Configuration Parameters

Parameter Description

T38 Fax The T38 check box indicates for each line whether to support

the T38 Fax protocol. The default is checked (T38 Fax supported).

Click on the Save button before leaving the page to save the new settings. The new

settings will be applied after restarting the unit.

3.7.2 SIP Extensions Page

Figure 3-15: SIP Extensions Page

The SIP Extensions page includes the following components:

Table 3-9: SIP Extensions Page Parameters

Parameter Description

Support PRACK method with provisional response reliability

The PRACK request plays the same role as ACK, but for provisional responses. PRACK is a normal SIP message, like BYE. As such, its own reliability is ensured hop-by-hop through each stateful proxy . Also like BYE, but unlike ACK, PRACK has its own response. If this were not the case, the PRACK message could not traverse proxy servers compliant to RFC 2543. For more details refer to RFC 3262: Reliability of Provisional Responses in the Session Initiation Protocol (SIP).

48 Operation

Table 3-9: SIP Extensions Page Parameters

Parameter Description

Telephone Menu

Encode SIP URI with user parameters

Encode default port in SIP URI

Include default port in INVITE

Send INVITE with timer header value

User=Phone will be inserted in the Contact field of SIP uniform resource identifier (URI).

Include default port in SIP uniform resource identifier (URI) even though it is not mandatory according to standard.

Include default port in the INVITE even though it is not mandatory according to standard

If the called user agents (UA) or the SIP Proxy Server (SPS) requires a session timer for a requested session and the calling UA does not include the Session-Expires header in the INVITE message, then the called UA or the SPS may reject the request with a 487-request failure message. If the use of a session timer is desirable but optional for the session and the calling UA does not include the Session-Expires header in the INVITE then the called UA or SPS may add a Session-Expires header to the next session setup message. In this case, the SPS shall add the Session-Expires header to the INVITE message and the called UA shall add the Session-Expires header to the 200 OK response message. The range for the timer header value is from 1 to 999.

SIP Session timer value The SIP Session Timer Support feature adds the capability to periodically

refresh Session Initiation Protocol (SIP) sessions by sending repeated INVITE requests. The repeated INVITE requests, or re-INVITEs, are sent during an active call leg to allow user agents (UA) or proxies to determine the status of a SIP session. Without this keep alive mechanism, proxies that remember incoming and outgoing requests (stateful proxies) may continue to retain call state needlessly. If a UA fails to send a BYE message at the end of a session or if the BYE message is lost because of network problems, a stateful proxy does not know that the session has ended. The re-INVITES ensure that active sessions stay active and completed sessions are terminated. The range for the timer value is from 1 to 999 seconds.

Use NOTIFY message to keep alive the session with SIP proxy every X seconds

The gateway will send a SIP NOTIFY message to the SIP proxy at the configured interval. These messages can keep the connection with SIP proxy alive, as well as the NAT port mapping when the Voice Gateway is behind NAT.

The range is: 0-99999 Default interval: 15 seconds

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Click on the Save SIP Extensions Settings button before leaving the page to save

the new settings. The new settings will be applied after restarting the unit.

3.7.3 NAT Traversal Configuration Page (SIP Only)

NAT Traversal function can be used to allow the Voice Gateway to register to a SIP

proxy server even though the Voice Gateway is connected behind a NAT device.

Port forwarding may need to be activated for all telephone ports used by the Voice

Gateway: For example, RTP port range and SIP signaling ports.

The Keep alive timeout parameter in the Telephony page may also need to be set

to a value lower than 1200 seconds to ensure that the Voice Gateway will not lose

registration to the SIP server.

Figure 3-16: NAT Traversal Configuration Extensions Page

The NAT Traversal Configuration page includes the following components:

Table 3-10: NAT Traversal Configuration Page Parameters

Parameter Description

External NAT-mapped IP Address

The IP address that the NAT device uses on the WAN side. If the Voice Gateway is set to Auto NA T mode (see belo w), the IP address of the outside IP will be automatically inserted. If the NAT Mode is set to On, a NAT IP Address must be set.

50 Operation

Table 3-10: NAT Traversal Configuration Page Parameters

Parameter Description

Static NAT Mode: The NAT mode:

 On = Enable NAT Traversal function using manual setting.

 Auto = Enter NAT mode if any of the following conditions is met:a.

IP-address = Private IP address b. "received" parameter in INVITE or REGISTER IP-address is not equal to internal IP address.

 Off = NAT Traversal function is disabled.The default is Off.

When using a NAT device, it is recommended to set this parameter to ON and to enter the External NAT-mapped IP Address.

Click on the Save button before leaving the page to save the new settings. The new

settings will be applied after restarting the unit.

Telephone Menu

3.7.4 STUN Client Configuration Page (SIP only)

Simple Traversal of UDP (STUN) is a method of NAT traversal through UDP, based

on RFC 3489. For proper voice conversations in networks based on more than one

NAT, the NAT devices must be able to transfer RTP packets in both directions.

The STUN protocol uses the default server port 3478. The Voice Gateway

communicates with a server over the internet. Based on the RTP packets, the

Server knows the number of NATs behind which the Voice Gateway is located, the

IP address of the Voice Gateway, and which of the Voice Gateway's ports are

actually used.

The STUN application supports the following NAT Types: FullCone, FullRestrict,

and PortRestrict. The Voice Gateway does not support symmetrical NAT as it is

not meant to traverse symmetrical NAT devices.

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Figure 3-17: STUN Client Configuration Page

The STUN Client page includes the following components:

Table 3-11: STUN Client Configuration Page Parameters

Parameter Description

STUN Client Mode Switch the STUN Client mode on or off. When on, turn off the

Static NAT Traversal mode. STUN Server Address (IP or Domain) The IP address or Domain of the STUN server. STUN Server Port The port used by the STUN Server.

The default is port 3478.

Click on the

the new settings. The new settings will be applied after restarting the unit.

STUN enabled cannot operate with NAT traversal enabled. In any case, the Voice

Gateway receives the external IP and the port information using the STUN.

Save STUN Client Settings button before leaving the page to save

3.7.5 ToS Page

Outgoing packets from the Voice Gateway can be marked with DSCP (DiffServ

Code Point) values. The ToS page enables defining the 8-bits ToS field in the IP

header for different packet types. Diffserv use the first 6 out of these 8 bits.

For more information about DiffServ Code Points please refer to RFC2474.

52 Operation

Figure 3-18: ToS Page

The ToS page includes the following components:

Telephone Menu

Table 3-12: ToS Page Parameters

Parameter Description

Call signaling Packets ToS marking for call signaling packets. Enter a number in the range 0 to

255 (The first 6 bits is the value of the DSCP field) or null. The default is

RTP Packets ToS marking for RTP and RTCP packets. Enter a number in the range 0

to 255 (The first 6 bits is the value of the DSCP field) or null. The default is 0.

SNMP Packets ToS marking for SNMP packets. Enter a number in the range 0 to 255

(The first 6 bits is the value of the DSCP field) or null. The default is 0.

Default setting ToS marking for other types of packets (e.g. HTTP, TFTP). Enter a

number in the range 0 to 255 (The first 6 bits is the value of the DSCP field) or null. The default is 0.

Click on the

settings. The new settings will be applied after restarting the unit.

Save ToS Settings button before leaving the page to save the new

3.7.6 Line Configuration Page

The Line Configuration page enables to select the country standard for Caller ID.

When using a caller ID device, select your country/standard from the list.

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Figure 3-19: Line Configuration Page

Click on the Save button before leaving the page to save the new settings. The new

settings will be applied after restarting the unit.

The default is Sweden.

3.7.7 Dial Plan Schemes

A dialplan gives the unit a map to determine when a complete number has been

entered and should be passed to the gatekeeper/SIP server for resolution into an

IP address. Dialplans are expressed using the same syntax as used by MGCP NCS

specification. The following notation describes the formal syntax of the dialplan:

Digit ::= "0" | "1" | "2" | "3" | "4" | "5" | "6" | "7" | "8" | "9"

Timer ::= "T" | "t"

Letter ::= Digit | Timer | "#" | "*" | "A" | "a" | "B" | "b" | "C" | "c" | "D" | "d"

Range ::= "X" | "x" -- matches any digit

| "[" Letters "]" -- matches any of the specified letters

Letters::= Subrange | Subrange Letters

Subrange::= Letter -- matches the specified letter

| Digit "-" Digit -- matches any digit between first and last

Position::= Letter | Range

StringElement::= Position -- matches any occurrence of the position

| Position "." -- matches an arbitrary number of occurrences including 0

String ::= StringElement | StringElement String

54 Operation

Telephone Menu

StringList::= String | String "|" StringList

DialPlan::= String | "(" StringList ")"

[0-9] denotes a single digit between 0 and 9. To configure a range of more than 10

numbers (e.g., 800xxx-819xxx) use the scheme: 80xxxx|81[0-9]xxx.

A dialplan, according to this syntax, is defined either by a (case insensitive) string

or by a list of strings. Regardless of the above syntax a timer is only allowed if it

appears in the last position in a string (12T3 is not valid). Each string is an

alternate numbering scheme. The unit will process the dialplan by comparing the

current dial string against the dialplan. If the result is under-qualified (partial

matches at least one entry) then it will do nothing further but wait until a full

match is reached. If the result is over-qualified (no further digits could possibly

produce a match) then it aborts the dial attempt and notifies end-user with an

audio signal. Only a full match will trigger to initiate a call, by sending the dialed

information to a Gatekeeper/SIP server.

The Timer T is activated when it is all that is required to produce a match. The

period of timer T is 4 seconds as default (configurable). For example a dialplan of

(xxxT|xxxxx) will match immediately if any 5 digits are entered. It will also match

following a 4 second pause after entering 3 digits.

IMPORTANT

The dialplan is according to section 2.1.5 of RFC 3435. The “.” notation, denotes zero or more keys. That is, x.# means none or at least one digit followed

by # and x.T means none or at least one digit followed by T. However, having only T in the dialplan (where x is null) activates the Hotline function (see

Section xx.#|xx.T.

3.7.7.1). To avoid unwanted activation of the hotline function, use the default dialplan,

Simple dialplan (Example 1):

Following example allows dialing any 7-digit number (e.g. 5551234) or an operator

on 0.

Dialplan is: (0T|xxxxxxx)

Complex dialplan (Example 2):

Local operator on 0, long distance operator on 00, four digit local extension

number starting with 3,4 or 5, seven digit local numbers are prefixed by an 8, two

digit star services (e.g. 69), ten digit long distance prefixed by 91, and

international numbers starting with 9011+one or more digits.

Dialplan for this is: (0T|00T|[3-5]xxx|8xxxxxxx|*xx|91xxxxxxxxxx|9011xx.T)

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Using supplemental external services (Example 3):

When a soft switch or a SIP server/gatekeeper exists in the network and the user

would like to use class 5 services which are not internal to the Voice Gateway e.g.,

*xy#, *xy*abcd#, #xy#, etc., the VG dialplan should be configured as follows:

[*#][0-9*][0-9*].#

Note that when VG internal class 5 services are enabled (default) in addition to the

above dialplan, the internal class 5 activation codes remain valid. See Appendix A.

Call completion

Call completion means allowing user to skip the timer period T after finished

dialing, by ending number sequence with '#' (no other character is valid for this

feature). A valid dialplan to accomplish this would be: (xx.#|xx.T)

3.7.7.1 Hotline

The hotline function allows a predetermined number to be called automatically by

waiting T seconds (which can also be configured) without pressing any keys.

The hotline function can also be used to receive tones from the Local Exchange.

This is achieved by leaving the number in the hotline dialplan empty. Additional

modifications may be required, in which case, contact Customer Support for

assistance.

The hotline feature is activated by specifying "T" (time-out) in the dialplan (by

default, T is set to 4 seconds).

For example: (xx.#|xx.T|<:1234>T)

The number 1234 will be dialed after T seconds.

3.7.7.2 Adding/Removing Prefixes

For outgoing calls

VG can add a pre-defined prefix to a dialed number via the dialplan and send the

number with the added prefix to the server. The prefix can automatically replace a

dialed digit using the following notation:

"<'dialed substring':'transmitted-string'>"

For example:

Set the dialplan to "<8:1860>xxx"

When dialing 8123, the digit 8 is replaced with 1860, and the actual number sent

is 1860123.

56 Operation

Telephone Menu

In ethereal trace, the "To" field in SIP INVITE is "1860123"

For incoming calls

VG can remove a prefix in the dialplan and show the number without the prefix on

the phone display. Use the following notation:

"<'replacement string':'received-string'>"

For example:

Set the dialplan to "<8:1860>xxx"

When a number with the 1860 prefix is received (e.g., "1860123"), the prefix 1860

is replaced with the digit 8 and the number displayed is 8123.

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3.8 BW Reservation - DRAP Configuration Page

The Voice Gateway uses DRAP (Dynamic Resource Allocation Protocol) for efficient

management of bandwidth resources for telephone calls.

Figure 3-20: DRAP Configuration Page

The DRAP Configuration page includes the following components:

Table 3-13: DRAP Configuration Page Parameters

Parameter Description DRAP Server Settings

58 Operation

BW Reservation - DRAP Configuration Page

Table 3-13: DRAP Configuration Page Parameters

Parameter Description

Enable DRAP The Enable DRAP option defines whether DRAP is used for

establishing telephone (voice and fax) calls. If enabled, a DRAP Server must be available to provision telephone calls.

The default is disabled (unchecked).

Enable Pre-allocation The Enable Pre-allocation option defines whether resource

allocation is requested immediately upon off-hook condition or only after dialing the requested number. When disabled (unchecked), a request for resource allocation will be sent only after dialing the number. When enabled, the resource allocation request will be sent immediately, and a dial tone will be provided only if the requested resources are available.

The default is enabled (checked).

DRAP Server IP Address The IP address of the DRAP server that should serve the

resource allocation requests of the unit. Leave empty for Auto Discovery.

The default is an empty field (Auto Discovery).

Server Port The UDP port used for the DRAP server. The port number

indicated will be used for originating ALLOC messages and the port number indicated +1 will be used for receiving CONFRM messages.

The available range is from 8000 to 8200. The default is 8171.

DRAP Protocol Options

Discovery Time The Discovery Time is the timeout to be used when the Auto

Discovery process is used for finding a DRAP server. The Auto Discovery process is based on sending empty broadcast allocation requests, and the Discovery Time is the time that the unit will wait for a response before sending a new request.

The range is 1 to 255 seconds. The default is 10 seconds.

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Table 3-13: DRAP Configuration Page Parameters

Parameter Description

Acknowledge Time The Acknowledge Time is the timeout out to be used between

allocation requests. If no confirmation is received within this time, a new allocation request should be sent.

The range is 1 to 10 (x 100 milliseconds). The default is 3 (300 milliseconds).

Clear Count The Clear Count parameter indicates the number of allocation

requests (ALLOC) that can be sent without being acknowledged before clearing all pending reservation attempts.

Note: Established reservations (existing calls) are not cleared. The range is 1 to 10. The default is 2.

Retry Count The Retry Count parameter indicates the number of allocation

requests (ALLOC) that can be sent without being acknowledged before reaching a decision that the unit should search for another server. When this number is reached est ablished reservations are to be cleared (existing calls are disconnected) and auto discovery procedure is initiated.

The range is 1 to 10. The default is 5.

RTP Packing Ratio The RTP Packing Ratio parameter defines the packet size to be

used until an actual call is established. It is recommended to set a value that supports the worst-case scenario, e.g. the smallest expected size (20 milliseconds) that results in the highest expected number of packets per second.NOT E: The configured RTP Packing Ratio is used by the unit until an actual call is established. Once a call is established, the unit will use a packet size according to the actual value being used for the call.

The available range is 10 to 100 milliseconds in multiples of 10 (10, 20, …100).

The default value is 30 milliseconds.

60 Operation

BW Reservation - DRAP Configuration Page

Table 3-13: DRAP Configuration Page Parameters

Parameter Description

Vocoder Type The Vocoder Type parameter defines the codec to be used until

an actual call is established. It is recommende d to set a value that supports the worst-case scenario, e.g. the codec with the highest bandwidth requirement. Typically G711 should be configured, except in networks where only G729 is used. NOTE: The configured Vocoder Type is used by the unit until an actual call is established. Once a call is established, the unit will use the actual codec type being used for the call.

The available options are G711 and G729. The default is G729.

Click on the Save DRAP Settings button before leaving the page to save the new

settings. The new settings will be applied after restarting the unit.

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3.9 System Menu

The System page selection bar includes the following options:

 Security (Section 3.9.1)

 Localization (Section 3.9.2)

 SNMP (Section 3.9.3)

 Service Access (Section 3.9.4)

 RTP Stats (Section 3.9.5)

3.9.1 Set Security Password Page

Figure 3-21: Set Security Password Page

The Set Security Password page includes the following components:

62 Operation

System Menu

Table 3-14: Set Security Password Page Parameters

Parameter Description

User name Enter the user name. The user name for users with operator

privileges (full access and read/write privileges) is operator and for user with administrator privileges (partial acces s and re ad /wr ite privileges) is admin. These user names cannot be changed.

Old password A password used previously. The default password for users with

operator privileges is installer. No password is required for users with administrator privileges.

New password Enter the new password. A password includes up to 20 printable

characters and is case sensitive.

A null (empty) string means no password. Confirm new password Enter the new password again (must be the same as above). Access Select the mode in which the PC can manage the IDU-DV unit. The

PC can manage the unit through the LAN (User Ethernet) port, the

WAN (Radio) or BOTH. It is recommended that you select BOTH.

Click on the Save Password button before leaving the page to save the new

password. Click on the

Save Access Mode button before leaving the page to save

the access mode. The new settings will be applied after restarting the unit.

When upgrading the unit, the new password is retained and does not revert to the

default password.

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3.9.2 Localization Page

Figure 3-22: Localization Page

The Localization page includes the following components:

Table 3-15: Localization Page Parame te rs

Parameter Description

NTP Server The IP address of the NTP-server (optional). If an IP address is

configured the NTP server usage is activated. The feature must be activated to support FSK-based caller ID.

The default is disabled (no IP address).

Time Zone The appropriate time zone. Use the drop-down list to change the time

zone.

Adjust clock for daylight savings

Click on the

the new settings. The new settings will be applied after restarting the unit.

By checking the "Adjust clock to daylight savings" the Voice Gateway will automatically adjust to daylight saving time (set the time one hour ahead).

The default is enabled (checked).

Save Localization Settings button before leaving the page to save

64 Operation

3.9.3 SNMP Configuration Page

System Menu

Figure 3-23: SNMP Configuration Page

The SNMP Configuration page includes the following components:

Table 3-16: SNMP Configuration Page Parameters

Parameter Description SNMP Trap Configuration

Trap Destination 1 to Trap destination 6

SNMP MIB Parameter Configuration

Specify up to 6 IP addre sses to which SNMP trap s should be sen t. Only these stations will be able to manage the Voice Gateway. If all Trap Destinations are null, SNMP traps will be sent as broadcasts, and any station will be able to manage the V oice Gateway.

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Table 3-16: SNMP Configuration Page Parameters

Parameter Description

Read Community The read community string, up to 20 printable characters, case

sensitive. Default string is public.

Write Community The write community string, up to 20 printable characters, case

sensitive. Default string is private

Click on the Save SNMP Settings button before leaving the page to save the new

settings. The new settings will be applied after restarting the unit.

3.9.4 Service Access Configuration Page

Figure 3-24: Service Access Configuration Page

The Service Access Configuration page enables to enable/disable access to

various services. Access from each of the ports (LAN or WAN) using HTTP and/or

SNMP can be either enabled or disabled. The default for all options is enabled

(checked).

Click on the

the new settings. The new settings will be applied after restarting the unit.

Save Service Access Settings button before leaving the page to save

66 Operation

3.9.5 RTP Statistics Page

The RTP Stats page enables to monitor from remote the performance of the last

call. The displayed information includes: Bandwidth (kb/s), jitter, packet loss,

and latency.

NOTE

The bandwidth relates to the payload only.

System Menu

Figure 3-25: RTP Statistics Page

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3.10 Upgrade Page

The Upgrade page enables to control the process of downloading either a software

file (with the extension .ro) or a configuration file (with the extension .ini) from a

TFTP-server or from an HTTP server.

Figure 3-26: Upgrade Page

The Upgrade page includes the following components:

Table 3-17: Upgrade Page Parameters

Parameter Description

Upgrade Type Auto The Voice Gateway will automatically select the server type for

download.

TFTP Downloads the file from the TFTP server according to the

specified host address.

HTTP Downloads the file from the HTTP server according to the

specified URL. Not implemented in the current release. Host (TFTP) The IP address of the TFTP server URL (HTTP/Auto) Use the following syntax: IP/filename. E.g.,

192.168.254.1/DMA0027R2F201.ro

File name The file name in the HTTP/TFTP server of the software or the

configuration .ini file. Up to 25 characters.

Click on the

process. The downloading and installation of the new SW version or configuration

Start Auto/HTTP/TFTP Upgrade button to start the download

68 Operation

Upgrade Page

file is done automatically, including a restart of the unit. When the installation is

complete and the unit has restarted, the Home Product Info page will be displayed

(if not, click on the Refresh button).

The version R2H implementation of the dialplan module is fully compliant with

RFC3435. However, some previously acceptable dialplans are inconsistent with

RFC3435 operation and must therefore be changed to ensure continued operation

consistent with previous releases. You should review your dialplan to ensure that

the dial behavior will continue to be that required by your customers.

For example, the default dialplan should be changed from (x.#|x.T) to (xx.#|xx.T).

It is recommended to review your dialplan to ensure that its proper operation.

You can change the dialplan as follows:

 Default dialplan

 If an INI file is not used: upgrade to version R2H and then install the INI file

provided with the upgrade package.

 If an INI file is used: you will be provided with an updated INI file.

 Non-default dialplan

 If an INI file is not used: Refer to Section 3.7.7 for information on updating

your dialplan. The dialplan can be changed via the Telephone menu, or via

option 43.

 If an INI file is used: you will be provided with an updated INI file.

3.10.1 Downloader Result Codes (hexadecimal)

If something goes wrong during download or installation, you will be notified

according to the following:

 0 (0x00): normal boot (no upgrade requested or needed)

 bit-0 (0x01): upgrade requested or main application not valid

 bit-1 (0x02): failed to download new image

 bit-2 (0x04): TFTP server not defined

 bit-3 (0x08): TFTP file not defined

Voice Gateways System Manual 69

Chapter 3 - Using the Web Configuration Server

 bit-4 (0x10): TFTP session failed

 bit-5 (0x20): CRC error in downloaded image

 bit-6 (0x40): incompatible image

3.10.1.1 Examples

 An attempt to download from a non-existing TFTP-server results in code 0x7 (=

0x07):

 bit-2 0x04 TFTP server not defined plus…

 bit-1 0x02 failed to download new image plus…

 bit-0 0x01 upgrade requested or main application not valid

 An attempt to download a non-existing file results in code 0xb (= 0x0b):

 bit-3 0x08 TFTP file not defined plus…

 bit-1 0x02 failed to download new image plus…

 bit-0 0x01 upgrade requested or main application not valid

 A successful download results in code 0x01

 A restart without download of main application results in 0x00.

70 Operation

3.11 Restart Page

When settings have been inserted or altered, the Voice Gateway must be restarted

in order to apply the new settings.

Restart Page

Click the

Figure 3-27: Restart Page

Restart button to restart the Voice Gateway.

Voice Gateways System Manual 71

Chapter 3 - Using the Web Configuration Server

3.12 Logout Page

Use this page to log out the system.

Figure 3-28: Logout Page

72 Operation

3.13 Parameters Summary

Table 3-18: Parameters Summary

Parameter Range/Options Default WAN Configuration Page

Device Operating Mode Only Bridge option is available Bridge

Parameters Summary

Obtain WAN configuration dynamically

Client identity Standard/Custom

Vendor ID A string of up to 25 characters VoIP (used by default but is

Specify fixed WAN configuration Yes (checked)/No

IP Address IP address Default: Dynamic

Subnet Mask IP address 255.255.255.0 Default Gateway IP address Null DNS Address IP address Null

Yes (checked)/No (unchecked)

The Custom string can include up to 25 characters

(unchecked)

Default: Yes (selected) Factory default (above 10 sec

HW reset): No (unchecked) Standard

The default Custom string is null

not displayed) No (unchecked)

Factory default (above 10 sec HW reset): 192.168.254.254

Host Name A string of up to 25 characters Null Domain Name A string of up to 25 characters Null

VoIP VLAN Configuration Page

Call Signaling VLAN Tag 1-4094 or null Null Call Signaling Priority Tag 0-7 or null Null RTP VLAN Tag 1-4094 or null Null RTP Priority Tag 0-7 or null Null

SIP Configuration / H323 Telephone Page

Voice Gateways System Manual 73

Chapter 3 - Using the Web Configuration Server

Table 3-18: Parameters Summary

Parameter Range/Options Default

Dialplan A string of up to 100

characters. For details on

format see Section Dial Timeout 1-60 seconds 4 seconds Use # Yes (checked)/No(unchecked) Checked RTP Port Range

(SIP model only) Telephone Line On/Off Off HA Mode Fixed/Auto/Off Off SIP Server IP (primary)

(SIP model only) SIP Server Port (primary)

(SIP model only) SIP Server IP (secondary)

(SIP model only)

Start/Stop: 1030-65535 Start: 8000

IP address Null

1030-65535 5060

IP address Null

3.7.7

xx.T|xx.#

End: 8015

SIP Server Port (secondary) (SIP model only)

Gate Keeper IP (primary) (H323 model only)

Gate Keeper IP (secondary) (H323 model only)

User Name (SIP model only)

Password (SIP model only)

Outgoing Display Name A string of up to 25 characters Null Telephone number A string of up to 25 characters Null

1030-65535 5060

IP address Null

A string of up to 25 characters Null

74 Operation

Table 3-18: Parameters Summary

Parameter Range/Options Default

Parameters Summary

H323 Alias

A string of up to 25 characters Null (H323 model only)

Telephone domain name

A string of up to 25 characters Null (SIP model only)

Port (SIP model only)

Message Waiting Account

1030-65535 Line 1: 5060

Line 2: 5061 Null

(SIP model only) Incoming CLIP On/Off Off Keepalive timeout 10-65535 seconds 1200 seconds Ring signal 0-9 0 Transport

UDP/TCP UDP (SIP model only)

Codecs and Fax Configuration Jitter Buffer:

Adaptive Jitter Buffer 100-300 milliseconds 100 milliseconds Fixed Jitter Buffer 100-300 milliseconds 40 milliseconds Automatically switch to Fixed Jitter

Buffer

Yes (checked)/No

(unchecked)

Unchecked

Codec:

G711A Select/Deselect Yes (checked)/No(unchecked) Yes (checked)

SS Not applicable EC Enable/Disable Enabled Packet 10-150 milliseconds 20 ms Keypad SIP: None, RFC2833,

None

SIP INFO

H323: H225, H245, RFC2833,

None

Priority 1-3 1

Voice Gateways System Manual 75

Chapter 3 - Using the Web Configuration Server

Table 3-18: Parameters Summary

Parameter Range/Options Default

G711U Select/Deselect Yes (checked)/No(unchecked) Yes (checked)

SS Not applicable EC Enable/Disable Enabled Packet 10-150 milliseconds 20 ms Keypad SIP: None, RFC2833,

SIP INFO

H323: H225, H245, RFC2833,

None

Priority 1-3 2

G729 Select/Deselect Yes (checked)/No(unchecked) Yes (checked)

SS Enable/Disable Disabled EC Enable/Disable Enabled Packet 10-150 milliseconds 30 ms Keypad SIP: None, RFC2833,

SIP INFO

H323: H225, H245, RFC2833,

None

Priority 1-3 Not Applicable

T38 Fax Enable/Disable Enable

SIP Extensions Page (SIP model only)

None

SIP: RFC2833 H323: H225

Support PRACK method with provisional response reliability

Encode SIP URI with user parameters

Encode default port in SIP URI Yes/No No (unchecked) Include default port in INVITE Yes/No Yes (checked) Send INVITE with timer header

value

SIP Session timer value Yes/No, plus a value in the

Yes/No No (unchecked)

Yes/No, plus a value in the

range 1-999 seconds if Yes

(checked) is selected.

range 1-999 seconds if Yes

(checked) is selected.

No (unchecked) The default value is null

76 Operation

Table 3-18: Parameters Summary

Parameter Range/Options Default

Parameters Summary

Use NOTIFY message to keep alive the session with SIP proxy every X seconds

Yes (checked)/No

(unchecked)

0-99999 seconds

15 seconds

NAT Traversal Configuration Page (SIP model only)

External NAT-mapped IP Address IP address Null Static NAT Mode On/Auto/Off Off

STUN Client Page

STUN Client Mode On/Off Off STUN Server Address IP address/Domain Null STUN Server Port 1-65534 3478

ToS Page

Call signaling Packets 0-255 or null 0 RTP Packets 0-255 or null 0 SNMP Packets 0-255 or null 0 Default setting 0-255 or null 0

Line Configuration Page

CLIP Standard A list Sweden

DRAP Configuration Page

Enable DRAP Enable/Disable Disable (unchecked) Enable Pre-allocation Enable/Disable Enable (checked) DRAP Server IP Address IP address or null for auto

Null (auto discovery)

discovery Server Port 8000-8200 8171 Discovery Time 1-255 seconds 10 seconds Acknowledge Time 1 to 10 (x 100 milliseconds) 3 (300 milliseconds) Clear Count 1-10 2 Retry Count 1-10 5 RTP Packing Ratio 10 to 100 milliseconds in

30 milliseconds

multiples of 10 (10, 20, …100) Vocoder Type G711/G729 G729

Voice Gateways System Manual 77

Chapter 3 - Using the Web Configuration Server

Table 3-18: Parameters Summary

Parameter Range/Options Default Set security Password Page

New password/ Confirm new password

Localization Page

NTP Server IP address or null for disable

Time Zone Drop Down Menu GMT+01:00 Adjust clock for daylight savings Yes/No Yes (checked)

SNMP Configuration Page

Trap Destination 1 to Trap destination 6

Read Community Up to 20 printable characters,

Write Community Up to 20 printable characters,

Up to 20 printable characters,

case sensitive. A null (empty)

string means no password.

NTP server

IP addresses.If all Trap

Destinations are null, SNMP

traps will be sent as

broadcasts.

case sensitive.

Null (NTP server disabled)

Null for all addresses

public

private

Service Access Configuration Page

HTTP LAN Yes/No Yes (checked) HTTP WAN Yes/No Yes (checked) SNMP LAN Yes/No Yes (checked) SNMP WAN Yes/No Yes (checked)

Upgrade Page

Upgrade Type Auto/TFTP/HTTP TFTP Host/URL IP address/Domain Name Null File name A string of up to 25 characters

78 Operation

Appendix A - Internal Class 5 Services

In This Appendix:

This appendix provides a description of the internal Class 5 services that are

supported by the Voice Gateway.

Appendix A - Internal Class 5 Services

A.1 Actions and Keypad Sequences

Table A-1: Keypad Sequences

Action Description Keypad Sequence

(R=hook-flash)

HOLD Holds an on-going call R0 DROP Drops an on-going call R1 FLASH Switche s be tween on-going call

sessions or starts new call

inquiry CONFERENCE Activates 3-party conference R3 CONFERENCE DROP Deactivates 3-party conference R5 CW ACTIVATION Enables Call Waiting indication

tone CW DEACTIVATION Disable Call Waiting indication

tone CW STATUS CHECK Informs about the current

configuration of Call Waiting

indication tone CALL FORWARD ACTIVATION Call Forward activation *21* <telephone number># CALL FORWARD

DEACTIVATION

Call Forward de-activation #21#

*43#

#43#

*#43#

80 Voice Gateways System Manual

A.2 Using the Class 5 Services

A.2.1 Call Waiting

Description: One on-going call active, audible CW tone indicating new incoming

call in progress.

Table A-2: Call Waiting Service

Action Event

Using the Class 5 Services

R0 Reject incoming call

call.0 R1 Disconnect on-going call and answer incoming call. R2 Place on-going call on hold, answer incoming call.

→ Calling party hears busy tone. Continue with active

A.2.2 Call Inquiry

Description: One on-going call active, place a new call to a third party.

Table A-3: Call Inquiry Service

Action Event

R2+telephone number Place on-going call on hold (dial tone), Inquire new call to a third party. R1 Return to call placed on hold if third party is not answering.

A.2.3 Call Alteration

Description: Two on-going calls active, switch between calls.

Table A-4: Call Alteration Service

Action Event

R2 Switch between two on-going calls. Places non-active call on hold.

A.2.4 Call Drop

Description: Two on-going calls active, disconnect one of the calls.

Voice Gateways System Manual 81

Appendix A - Internal Class 5 Services

Table A-5: Call Drop Service

Action Event

R0 Disconnect call that is put on hold. Continue with on-going call. R1 Disconnect on-going call and return to call that is put on hold.

A.2.5 3-Party Conference 1

Description: One on-going call active, place a new call to a third party and start

conference.

Ta ble A-6: 3-Party Conference Service 1

Action Event

R3+telephone number Place on-going call on hold (dial tone), inquire new call to a third party

and mix all session into a conference when third party has answered.

R5 End conference with third party and return to first initiated call session.

A.2.6 3-Party Conference 2

Description: Two on-going calls active, mix them into a conference session.

Ta ble A-7: 3-Party Conference Service 2

Action Event

R3 Start conference with all active parties (mix audio streams). R5 End conference with third party and return to first initiated call session.

A.2.7 Call Waiting Indication Tone

Description: Available only when there are no calls active/in progress.

Table A-8: Call Waiting Indication Tone Service

Action Event

*43# Enable Call Waiting indication tone #43# Disable Call Waiting indication tone (calling party will hear a busy tone

when calling)

82 Voice Gateways System Manual

Using the Class 5 Services

Table A-8: Call Waiting Indication Tone Service

Action Event

*#43# Informs about present Call Waiting indication tone configuratio n:

Three short beeps = off Two long beeps = on

A.2.8 Call Forward

Description: Available only when there are no calls active/in progress.

Table A-9: Call Forward Service

Action Event

*21*<telephone number># Enable Call Forward and do forward to <telephone number>.

Indication tone is heard.

#21# Deactivate Call Forward

Voice Gateways System Manual 83

Appendix B - Default Telephony Parameters

In This Appendix:

This appendix provides the default settings for various telephony parameters.

Appendix B - Default Telephony Parameters

Table B-1: Default Telephony Parameters

Parameter Definition Default

Normal Ringing Signal The signal that end user will

hear from the telephone set when a call is received

Ringing Tone The tone that sounds on the

telephone set when ringing on the other side.

Dial Tone The tone that the call originator

hears in the handset before dialing the destination telephone number.

Busy Tone The tone that the end user that

originates a call hears when the destination telephone line is busy.

Cadence: 1 second on, 4 second off

Duration: 180 seconds Frequency: 25 Hz

Cadence: 1 second on, 5 second off

Duration: Not limited Frequency: 425 Hz Level: -10 dBmO

Cadence: Continuous Duration: Not limited Frequency: 425 Hz Level: -5 dBmO

Cadence: 0.25 second on, 0.25 second off

Duration: Not limited Frequency: 425 Hz Level: -10 dBmO

Network Busy Tone The tone that the end user that

originates a call will hear when

Cadence: 0.25 second on, 0.75 second off

the network is congested.

Duration: Not limited Frequency: 425 Hz Level: -10 dBmO

Call Waiting Tone The tone that the end user that

originates the call hears when there is a second incoming call during the original call session.

Cadence: 0.2 second on, 0.5 second off, 0.2 seconds on

Duration: One On-Off-On cycle Frequency: 425 Hz Level: -10 dBmO

86 Voice Gateways System Manual

Table B-1: Default Telephony Parameters

Parameter Definition Default

Caller ID Standard The country standard and data

transmission method

CID Alerting Method The method of alerting on the

existence of CID data

On-hook data transmission timing method

The timing for transmission of CID data

2-Wire Impedance The impedance presented

between the A and B wires of the telephone line in active state (nominal impedance).

Sweden, DTMF

Polarity Reversal

Post first ring signal

CTR21 (ETSI complex =

270Ω+750Ω||150nF))

Voice Gateways System Manual 87

Alvarion SIP R2J User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual

About This Manual

Contents

Chapter 1 - System Description

1.1 Introducing the V o ice Gateway

1.2 Specifications

1.2.1 Telephony and Fax Services

1.2.2 Security

1.2.3 Voice Quality

1.2.4 Configuration and Management

1.2.5 Bridge Functionality

1.2.6 Mechanical

1.2.7 Electrical

1.2.8 Connectors

1.2.9 Regulatory Standards Compliance

1.2.10 Environmental

Chapter 2 - Installation

2.1 Installation Requirements

2.1.1 Packing List

2.1.2 Additional Installation Requirements

2.2 Front and Rear Panel Components

2.2.1 Connectors

2.2.2 Reset to Factory Default Configuration

2.2.3 LEDs

2.3 Installation and Commissioning

2.4 Notes on Using the Voice Gateways in Alvarion's Systems

2.4.1 BreezeMAX System (Version 1.5 and higher)

2.4.2 BreezeACCESS VL System (Version 3.1)

Chapter 3 - Using the Web Configuration Server

3.1 Introduction to the Web Configuration Server

3.2 Accessing the We b Configura t ion Server

3.3 Using the Web Configuration Server

3.4 Home Menu - Product Info Page

3.5 WAN Menu

3.5.1 WAN Status Page

3.5.2 WAN Configuration Page

3.6 VLAN Tagging Menu

3.6.1 VLAN Tagging Page

3.6.2 Adding and Deleting VLANs

3.6.3 VoIP VLAN Configuration Page

3.6.4 VLAN Configuration Example 1

3.6.5 VLAN Configuration Example 2

3.6.5.1 VG-1 Configuration

3.6.5.2 VG-2 Configuration

3.6.6 VLAN Configuration Example 3

3.6.6.1 Method 1

3.6.6.2 Method 2

3.7 Telephone Menu

3.7.1 SIP/H323 Configuration Page

3.7.1.1 Codecs and Fax Configuration

3.7.2 SIP Extensions Page

3.7.3 NAT Traversal Configuration Page (SIP Only)

3.7.4 STUN Client Configuration Page (SIP only)

3.7.5 ToS Page

3.7.6 Line Configuration Page

3.7.7 Dial Plan Schemes

3.7.7.1 Hotline

3.7.7.2 Adding/Removing Prefixes

3.8 BW Reservation - DRAP Configuration Page

3.9 System Menu

3.9.1 Set Security Password Page

3.9.2 Localization Page

3.9.3 SNMP Configuration Page

3.9.4 Service Access Configuration Page

3.9.5 RTP Statistics Page

3.10 Upgrade Page

3.10.1 Downloader Result Codes (hexadecimal)

3.10.1.1 Examples

3.11 Restart Page

3.12 Logout Page

3.13 Parameters Summary

Appendix A - Internal Class 5 Services

A.1 Actions and Keypad Sequences

A.2 Using the Class 5 Services

A.2.1 Call Waiting

A.2.2 Call Inquiry

A.2.3 Call Alteration